POP-1
Table of Contents
Volume 1
Course Introduction
1
Overview
1
Learner Skills and Knowledge
Course Goal and Objectives
3
4
Course Flow
5
Additional References
5
Cisco Glossary of Terms
Your Training Curriculum
6
1-1
Multisite Deployment Implementation
L
1-1
Overview
1-1
Module Objectives
1-3
Identifying Issues in a Multisite Deployment
1-3
Objectives
1-4
Multisite DeploymentIssues Overview
1-6
Quality Issues
1-7
Quality Issues Example: Jitterand Packet Drops
1-8
1-8
Bandwidth Issues
Example: Bandwidth Issues
Bandwidth Issues Example: Voice and DataTraffic Competing for Bandwidth
Bandwidth Issues Example: Load Caused by Centralized Media Services
1-11
Availability Issues
1-12
Availability Issues Example: IP WAN Failure
1-13
Dial Plan Issues
Example: Variable-Length Numbering, E.164 Addressing, and DID
1-16
Fixed vs. Variable-Length Numbering Plans
Detection of End of Dialing in Variable-Length Numbering Plans
1-17
1-18
1-20
Optimized Call Routing and PSTN Backup
1-21
1-22
1-23
Example: TEHO
jm
Example: Overlapping and Nonconsecutive Numbers
Various PSTN Requirements
1-25
Issues Caused by Different PSTN Dialing
1-26
1-28
Dial Plan Scalability Issues
NAT and Security Issues
Example: NAT Security Issues
Summary
1-29
1-30
1-30
References
L
identifying Multisite Deployment Solutions
„
Objectives
Multisite Deployment Solution Overview
QoS
mm
1-9
1-10
QoS Advantages
Solutions to Bandwidth Limitations
1-31
1-31
1-32
1-33
1-34
1-35
Low-Bandwidth Codecs and RTP-Header Compression
1-37
Codec Configuration in Cisco Unified Communications Manager
1-38
1-39
1-40
Disabled Annunciator
Local vs. Remote Conference Bridges
Transcoders
1-41
Guidelines for Transcoder Configuration
1-42
Mixed Conference Bridge
Multicast MOH from Branch Router Flash
Multicast MOH from Branch Router Flash Example
1-44
Multicast MOHfrom Branch Router Flash: Cisco IOS Configuration Example
Alternatives to Multicast MOH from Branch Router Flash
1-49
1-50
Preventing Too Many Calls by CAC
1-51
1-45
1-47
Availability
1-52
PSTN Backup
MGCP Fallback; Normal Operation
1-53
1-54
MGCP Fallback: Fallback Mode
1-55
Fallback for IP Phones; Normal Operation
1-56
Fallback for IP Phones: Fallback Mode
Using CFUR toReach Remote-Site IP Phones Over the PSTN During WAN Failure
Using CFUR to Reach Usersof Unregistered Software IP Phones on Their Cell Phones
Automated Alternate Routing
Mobility Solutions
Dial Plan Solutions
1-64
Globalized Call-Routing Overview
Globalized Call Routing—Three Phases
Globalized Call Routing Advantages
1-65
1-68
1-70
NAT and Secunty Solutions
1-72
Cisco Unified Border Element in Flow-Through Mode
1-73
Summary
1-74
References
1-75
Implementing Multisite Connections
1-77
Objectives
Multisite Connection Options
1-77
1-78
1-79
Cisco Unified Communications ManagerConnection Options Overview
Cisco IOS Gateway Protocol Functions Review
Cisco IOS Gateway Protocol Comparison Review
1-80
1-81
SIP Trunk Characteristics
H.323 Trunk Overview
1-82
1-83
H 323 Trunk Comparison
1-84
MGCP Gateway Implementation Review
Cisco IOS Gateway MGCP Configuration Methods Review
1-86
1-87
Configuring Cisco IOS Gatewayfor MGCP—Example
1-89
H.323 Gateway Implementation Review
1-91
Cisco Unified Communications Manager H.323 Gateway Configuration
1-93
1-94
1-95
Cisco IOS H.323 Gateway Configuration
Trunk Implementation Overview
Gatekeeper-Controlled ICT and H.225 Trunk Configuration Overview
Trunk Types Used by Special Applications
SIP Trunk Implementation
Intercluster and H.225 Trunk Implementation
Cisco Unified Communications Manager Gatekeeper-Controlled ICT and H.225 Trunk
Configuration
Summary
References
1-96
1-97
1-98
1-100
1-102
1-105
1-105
Implementing a Dial Plan for International Multisite Deployments
1-107
Objectives
1-107
Multisite Dial Plan Overview
1-108
Dial Plan Requirements for Multisite Deployments with Distributed Call Processing
Digit Manipulation Requirements for UsingAccess and Site Codes
Centralized Call-Processing Deployments: Access and Site Codes
Implementing PSTN Access in Cisco IOS Gateways
PSTN Access Example
1-58
1-59
1-60
1-61
1-62
Dial Plan Components in Multisite Deployments
Dial Plan Scalability Solutions
Implementing Site Codes for On-Net Calls
1-57
1-109
1-110
1-112
1-113
1-114
1-115
1-116
ISDN TON
1-117
Example: ISDN TON—Calling NumberTransformation of Incoming Call
1-118
Implementing Selective PSTN Breakout
Configuring IP Phones to Use Local PSTN Gateway
Implementing PSTN Backup for On-Net Intersite Calls
DigitManipulation Requirements for PSTN Backup of On-Net Intersite Calls
ImplementingCisco Unified Communications Manager. Part 2 (CIPT2)v8.0
1-119
1-120
1-121
1-122
)2010 Cisco Systems, Inc.
Implementing TEHO
Considerations for Using Remote PSTN Gateways
mm
mm
&.
TEHO Example Without Local Route Groups
TEHO Example with Local Route Groups
Implementing Globalized Call Routing
Globalized Call Routing: Number Formats
Normalization ofLocalized Call Ingress onGateways
Normalization of Localized Call Ingress from Phones
I Icl*
Module Summary
,
References
Module Self-Check
Module Self-Check Answer Key
Overview
f
•*
f
**
*
mm
f'
*•'
**
*•*
"*
<•*»
*
1-151
1'153
1'158
2^
2-1
Module Objectives
2-1
Examining Remote Site Redundancy Options
^m
1-146
I'ltn
References
Centralized Call-Processina Redundancy Implementation
^.
^^
1-137
1-147
1-148
Summary
Me
1-133
1-136
Globalized Call Routing—TEHO Advantages
Globalized Call Routing—TEHO Example
*
m-
1"127
1"129
]-] 31
1-138
1-140
1-142
Considering Globalized Call-Routing Interdependences
fk.
'§m
]_J24
l-i«
Localized Call Egress atGateways
Localized Call Egress at Phones
Globalized Call-Routing Example: Emergency Dialing
mm
mm
-
Objectives
Remote Site Redundancy Overview
2^
2"3
2-4
Remote Site Redundancy Technologies
2-5
When to Use MGCP Fallback
When to Use Cisco Unified SRST
2-7
When to Use Cisco Unified Communications Manager Express inSRST Mode
Cisco Unified SRST Operation
Cisco Unified SRST Function: Switchover Signaling
2-6
2-9
2-10
2-11
Cisco Unified SRST Function: Call Flow After Switchover
2-12
Cisco Unified SRST Function: Switchback
Cisco Unified SRST Timing
2-13
2-14
MGCP Fallback Operation
MGCP Gateway Fallback: Switchover
MGCP Gateway Fallback: Switchback
MGCP Gateway Fallback Process
Cisco Unified SRST Versions and Feature Support
Cisco Unified SRST 8.0 Platform Density
Plus (+) Prefix and E.164 Supportin Cisco Unified SRST
Support for Multiple MOH Sources
Dial Plan Requirements for MGCP Fallbackand Cisco Unified SRST Scenarios
Ensure Connectivity for Remote Sites
Ensure Connectivity from Main Site UsingCFUR
CFUR Considerations
CFUR Interaction with Globalized Call Routing
CFUR Example WithoutGlobalized Call Routing
CFUR Example with Globalized Call Routing
2-15
2-16
2-17
2-18
2-19
2-20
2-21
2-22
2-23
2-24
2-25
2-26
2-28
2-30
2-31
Keeping Calling Privileges Active in SRST Mode
2-32
Cisco Unified SRST Dial Plan Requirements Example
Summary
2-33
2-34
References
© 2010 Cisco Systems, Inc.
2-34
Implementing Cisco Unified Communications Manager, Part2(CIPT2) v8.0
Implementing SRST and MGCP Fallback
2-35
Objectives
MGCP Fallbackand Cisco Unified SRST Configuration Overview
MGCP Fallback and Cisco Unified SRST Configuration Requirements
Cisco Unified Communications ManagerSRST Configuration
Step 1: SRST Reference
Step 2: Device Pool
Cisco IOS Gateway SRST Configuration
Steps 1 and 2: Enabling Cisco Unified SRST and Setting Cisco Unified SRST IP Address
Steps 3 and 4: Setting Maximum Directory Numbers and Telephones
Steps 5 and 6: Setting Maximum Directory Numbers Per Phone and Keepalive Timer
Cisco Unified SRST Configuration Example
Cisco IOS GatewayMGCP GatewayFallback Configuration
Steps 1 and 2: Enabling MGCP Fallback and Setting Fallback Service
MGCP Fallback Configuration Example
2-35
2-36
2-37
2-38
2-39
2-40
2-41
2^2
2-43
2^J4
2-45
2-46
2-47
2-48
Cisco Unified Communications Manager Dial Plan Configuration for SRSTSupport
2-49
Step 1: Define a CSS for CFUR
2-49
2-50
Step 2: Setting Max Forward Unregistered Hops to DN
Step 3: Configuring CFUR
2-51
Cisco IOS GatewayMGCP Fallback and Cisco Unified SRST Dial Plan Configuration
2-53
Additional SRST Dial Plan Requirements
Cisco Unified SRST Dial Plan Commands: Dial Peer
2-54
2-56
Cisco Unified SRST Dial Plan Commands: Open Numbering Plans
2-60
Cisco Unified SRST Dial Plan Commands: Number Modification (Voice Translation Profiles) 2-62
Cisco Unified SRST Dial Plan Commands: Number Modification (Voice Translation Rules) 2-63
Cisco Unified SRST Dial Plan Commands: Number Modification (Profile Activation)
2-64
Cisco Unified SRST Dial Plan Commands: COR
2-65
Cisco Unified SRST Dial Plan Example
Summary
2-67
2-70
References
2-70
Implementing Cisco Unified Communications Manager Express in SRST Mode
Objectives
2-71
Cisco Unified Communications Manager Express Overview
Cisco Unified Communications Manager Express in SRST Mode
When to Use Cisco Unified Communications Manager Express in SRST Mode
Cisco Unified Communications Manager Express Features
Important Cisco Unified Communications Manager Express Features
General Configuration of Cisco Unified CommunicationsManager Express
Cisco Unified Communications Manager Express: Basic Configuration Example
Providing Phone Loads
Cisco Unified Communications Manager Express: MOH
2-72
2-73
2-74
2-78
2-79
2-80
2-82
2-83
2-84
Additional MOH Sources
2-85
Additional Music on Hold Sources—Configuration Example
Configuration of Cisco Unified Communications Manager Express in SRST Mode
Phone Provisioning Options
Advantages of Cisco Unified Communications Manager Express in SRST Mode
Phone Registration Process
Configuring Cisco Unified Communications Manager Express in SRST Mode
Cisco Unified Communications Manager Express in SRST Mode Configuration Example
Summary
References
2-71
2-86
2-87
2-89
2-90
2-91
2-92
2-94
2-95
2-95
Module Summary
2-97
References
Module Self-Check
2-97
2-99
Module Self-Check Answer Key
Implementing Cisco Unified Communications Manager. Part 2 (CIPT2] v8.0
2-102
©2010 Cisco Systems, Inc
Bandwidth Management and CAC Implementation
3-1
3-1
Overview
3-1
Module Objectives
Managing Bandwidth
Objectives
Bandwidth Management Overview
Cisco Unified Communications Manager Codec Configuration
Review of Cisco Unified Communications l\ anager Codecs
Example: Codec Configuration
Local Conference Bridge implementation
Example: Implementing Local Conference I ridges at Two Sites
Transcoder Implementation
Example: Implementing a Transcoder at the Main Site
Configuration Procedure for Implementing ranscoders
Step 1: Add Transcoder Resource in Cisco Jnified Communications Manager
Step 2: Configure Transcoder Resource in isco IOS Software
Multicast MOH from Branch Router Flash Implementation
Multicast MOH from Branch Router Flash: I legion Considerations
Multicast MOH from Branch Router Flash: , .ddress and Port Considerations
Multicast MOH: Address and Port Incremer t Example
Example: Implementing Multicast MOH froi i Branch Router Flash
Configuration Procedure for Implementing lulticast MOH from Branch Router Flash
Step 1; Enable Multicast Routing on Cisco OS Routers
Step 2a: Configure MOH Audio Sources foi Multicast MOH
Step 2b: Configure Multicast MOH in Cisco|Unified Communications Manager
ia Resource Groups
Step 2c: Enabling Multicast MOH at the
Step 3: Enable Multicast MOH from Branch!Router Flash at the Branch Router
Used for MOH RTP Packets
Step 4a: Configure the Maximum Hops to
Step 4b: Use IP ACL at IP WAN Router Ints rface
Step 4c: Disable Multicast Routing on IP W VN Router Interface
Summary
3-3
3-3
3-4
3-6
3-7
3-8
3-10
3-11
3-13
3-15
3-17
3-18
3-19
3-21
3-23
3-24
3-25
3-27
3-30
3-31
3-32
3-33
3-34
3-35
3-36
3-37
3-38
3-39
3-39
References
Implementing CAC
Objectives
3-41
CAC Overview
3-42
3-41
CAC in Cisco Unified Communications Manjager
3-43
3-44
Standard Locations
Locations: Hub-and-Spoke Topology
Locations: Full-Mesh Topology
3-45
Configuration Procedure for Implementing Ijocations-Based CAC
3^17
Locations Configuration Example: Hub-and Spoke Topology
3-48
Step 1: Configure Locations
Step 2: Assign Locations to Devices
3-49
3-46
3-50
3-51
RSVP-Enabled Locations
Three Call Legs with RSVP-Enabled Locations
3-52
Characteristics of Phone-to-RSVP Agent Ci II Legs
Characteristics of RSVP Agent-to-RSVP Agent Call Leg
3-53
How RSVP Works
3-55
Configuration Procedure for Implementing RSVP-Enabled Locations-Based CAC
Example: RSVP-Enabled Locations Configuration
Step 1: Configure RSVP Service Parameters
Step 2: Configure RSVP Agents in Cisco IOS Software
Step 3: Add RSVP Agents to Cisco Unified Communications Manager
Step 4: Enable RSVP Between Location Pairs
Automated Alternate Routing
3-54
3-57
3-58
3-59
3-63
3-65
3-66
3-68
AAR Characteristics
3-69
AAR Example Without Local Route Groups and Globalized Numbers
AAR Example with Local Route Groups and Globalized Numbers
3-70
© 2010 Cisco Systems. Inc.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
3-72
AAR Considerations
3.74
3-75
3-76
3.77
3-78
3-80
3-81
AAR Configuration Procedure
Step 1: Configure AAR Service Parameters
Step 3: Configure AAR Groups
Step 4: Configure Phones for AAR
SIP Preconditions
CAC Without SIP Preconditions
CAC with SIP Preconditions
SIP Preconditions Operation
3-82
3-83
SIP Preconditions Call Flow Summary
3-85
Fallback from End-to-End RSVP to Local RSVP
SIP Preconditions Configuration Procedure
Step 2a: Configure SIP Profile
Step 2b: Apply SIP Profile to Trunk
H.323 Gatekeeper CAC
3-87
3-89
3-90
3-91
3-92
Example: H.323 Gatekeeper Used for Call Routing (Address Resolution) Only
3-94
Using an H.323 Gatekeeper for CAC
3-96
Example: H.323 Gatekeeper Also Used for CAC
3-98
Providing PSTN Backup for Calls Rejected by CAC
3-100
Configuration Procedure for Implementing H.323 Gatekeeper-Controlled Trunks with CAC 3-102
Summary
3-104
References
Module Summary
3-104
3-105
References
3-105
Module Self-Check
3-107
Module Self-Check Answer Key
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
3-109
©2010 Cisco Systems, Inc.
CIPT2
Course Introduction
Overview
Implemenling Cisco Unified Communicatioi s Manager, Part 2 (C1PT2) v8.0prepares you for
implementing a Cisco Unified Communications solution in a multisite environment. It covers
globalized call routing.Cisco Service Advertisement Framework (SAF) and Call Control
Discovery (CCD). tail-end hop-off (TEHO), Cisco Unified Survivable Remote Site Telephony
(SRST). and mobility features such as Devi e Mobility and Cisco Extension Mobility.
You will apply a dial plan for a multisite en ironment including TEHO, configure survivability
solutions toreduce bandwidth
for remote sites during WAN failure and implement
i
requirements in the IP WAN. You will also mable Call Admission Control (CAC) including
Session Initiation Protocol (SIP) Preconditk ns and automated alternate routing (AAR).
Learner Skills and Knowledge
This subtopic lists the skills and knowledge that learners must possess to benefit fully from the
course. The subtopic also includes recommended Cisco learning offerings that learners should
first complete to benefit fully from this course.
Learner Skills and Knowledge
• Working knowledge of converged voice and data networks
• Working knowledge of the MGCP, SIP, and H.323 protocols
and their implementation on Cisco IOSgateways
• Ability to configure and operate Cisco routers and switches
* Ability to configure and operate Cisco Unified
Communications Manager in a single-site environment
Learner Skills and Knowledge (Cont,)
• Cisco learning offerings:
Implementing Cisco Voice Communications and QoS
(CVOICE) v8.0
Implementing Cisco Unified Communications Manager,
PartT(CIPT1)v8.0
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8 0
>2010 Cisco Systems. Inc
Course Goal and Objectives
This topic describes the course goal and objectives.
"To provide learners with the necessary knowledge and skills
to implement a multisite Cisco Unified Communications
deployment, including new features of Cisco Unified
Communications Manager version 8 0 such as Call Control
Discovery, which is based on the Cisco Service
Advertisement Framework"
!iW!sme'-'i'ng Osco UnfrsdComfrvmcabons Manager, ran 2('CIP[2) v8 0
Uponcompleting this course, you will be able to meet these objectives:
•
Describe multisitedeployment issuesand solutions, and describe and configure required
dial plan elements
•
Implement call-processing resiliency in remote sitesby using Cisco Unified SRST, MGCP
fallback, and Cisco Unified Communications Manager Express in Cisco Unified SRST
mode
•
Implement bandwidth management and CAC to prevent oversubscription of the IP WAN
•
Implement Device Mobility and Cisco Extension Mobility
•
Describe and implement CCD deployments
© 2010 Cisco Systems. Inc.
Course Introduction
Course Flow
This topic presents the suggested flow of the course materials.
Course Flow
s implementation
[Course Introduction [
. J
Aj
MJtsite
|
Multisite
Deployment
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•
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(Cor*)
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(Cont)
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Bandwidth
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.
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;
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(Cont)
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CAC Implementation;
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and CAC
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'
Cal-Processing
Implementation
Applications for
M • fmplenientaliort
Redun&ncy
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CCD
(Cont)
Multisite
Deplovmerts
The schedule reflectsthe recommended structure for this course. This structure allowsenough
time for the instructor to present the course information and for you to work through the lab
activities. The exact timing of the subjectmaterials and liibs depends on the pace of your
specific class.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
©2010 Cisco Systems. Inc
Additional References
This topic presents the Cisco icons and symbols thatareused in this course, as well as
information on where to find additional technical references.
Cisco Icons and Symbols
Cisco Urified
Presence
Cisco Unified
Network
Communications
Cloud
Manager
Gatekeeper
Cisco Unrty
Connection
_^^ Cisco Unified
t^H Border Element
Gateway
Cisco Unified
Messaging
Gas* a/
Voice Router
Personal
Communicator
Cisco Un ified
Coco Adapt ve
Secirty Appliance
Cisco Unified
Communications
SRST Router
Manager Express
Cisco Unified
SAF Enabled
Communications
Router
Manager Express with
Cisco Unity Express
Cisco Glossary of Terms
For additional information on Cisco terminology, refer to the Cisco Internetworking Terms and
Acronyms glossary of terms at
http;/'docwiki.ciseo.com/wiki.'CatcgoiT:lntcrnctw'orking_"fernis_and_Acronytris_(I'fA).
© 2010 Cisco Systems. Inc.
Course Introduction
Your Training Curriculum
This topic presents the training curriculum for this course.
You are encouraged to join the Cisco Certification Community, a discussion forum open to
anyone holdinga valid CiscoCareerCertification (such as CiscoCCIII",CCNA'. CCDA",
CCNP'. CCDP". CCIP". CCVP".or CCSP*). It provides a gathering place for Ciscocertified
professionals to share questions, suggestions, and information about Cisco Career Certification
programs and other certification-related topics. For more information, visit
http://\v\\\\.cisa>.eom;go certifications.
Implementing Cisco Unified Communications Manager,Part 2 (CIPT2) v8 0
© 2010 Cisco Systems, Inc.
Cisco Career Certifications: CCNP Voice
Expand your professional options and advance your career.
Achieve professional-level recognition in voice networking.
Recommended TrawwgThiouBtt
Cisco Leamw*g Partners
implementing Qsco voiceCammunieaBans
and QoS
tmpmmnms Cisco UnlfletS Cormmiricetans
"
Manage*. Pa* i
imptormntmg Oseo Unified Communfeafions
Manager, Pari 2
TroiJWMBODflng Cttco Urftea
CamnwrtcaSons
___
'imgn^aseoWllkKlCommunications
Appticatiom
Voice Metw or king
i 2010 Cisco Systems, Inc.
www Cisco com/go/certificatons
Course Introduction
8
Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0
©2010Cisco Systems, Inc.
Module 1
Multisite Deployment
Implementation
Overview
In a multisite Cisco Unified Communications Manager deployment, special requirements exist
that are not necessary insingle-site deployments. To successfully deploy a multisite Cisco
Unified Communications Manager solution, you need to understand the issues and be aware of
their possible solutions.
This module discusses the issues in a multisite Cisco Unified Communications Manager
deployment, including selective public switched telephone network (PSTN) access and tail-end
hop-off (TEHO).
Module Objectives
Upon completing this module, you will be able to describe multisite deployment issues and
solutions, and describe and configure required dial plan elements.
This ability includes being able to meet these objectives:
•
Explain issues pertaining to multisite deployment and relate the issues to multisite
connection options
•
Describe solutions for multisite deployment issues
•
Configure gateways and trunks in multisite environments
• Implement adial plan to support inbound and outbound PSTN dialing, site-code dialing,
and TL1IO in an international environment
1-2
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems. Inc.
Lesson 1
Identifying Issues in a Multisite
Deployment
Overview
When deploying Cisco Unified Communications Manager in a multisite environment, there are
some unique aspects that pertain only to multisite deployments that you need to consider.
Deploying Cisco Unified Communications solutions among multiple sites requires an
appropriate dial plan, adequate bandwidth between sites, implementation of quality of service
(QoS). and a design that can survive IP WAN failures. This lesson identifies the issues that can
arise in a multisite Cisco Unified Communications Manager deployment.
Objectives
Upon completing this lesson, you will be able to explain issues pertaining to multisite
deployment and relate the issues to multisite connection options. This ability includes bein^
able to meet these objectives:
•
Describe issues pertaining to multisite deployments
•
Describe quality issues in multisite deployments
•
Describe issues with bandwidth in multisite deployments
•
Describe availability issues in multisite deployments
•
Describe dial plan issues in multisite]dep!oyments
•
Describe NAT and security issues in multisite deployments
i
Multisite Deployment Issues Overview
This topic provides an overview about issues pertaining to Cisco Unified Communications
Manager multisite deployments.
Issues in Multisite Deployments
Mam Site
Cisco Unified
Communications
Manager
In a multisite deployment, several issues can arise. Of those issues, here are the most important:
•
•
Quality issues: When real-time traffic like voice or video travels over a packet-switching
network such as an IP network, delay-sensitive packets have to be given priority to avoid
jitter resulting in decreased voice quality
Bandwidth issues: Cisco Unified Communications solutions can include voice and video
streams, signaling traffic, management traffic, and application traffic, such as rich-media
conferences. The additional bandwidth that is required when deploying a Cisco Unified
Communications solution has to be calculated and provisioned. These tasks ensure thai
classical data applications and Cisco Unified Communications applications do not overload
the available bandwidth. You should optimize bandwidth consumption by eliminating
unnecessary IP WAN traffic.
•
Availabilih issues: When you are deploying Cisco Unified Communications Manager
with centralized call processing. IP phones register with the Cisco Unified Communications
Manager over the IP WAN. If gateways in remote sites arc using the Media Gateway
Control Protocol (MGCP) as a signaling protocol, they also depend on the availability of
the Cisco Unified Communications Manager as a call agent. It is important to implement
fallback solutions for IP phones and gateways in scenarios in which the connection to Cisco
Unified Communications Manager is broken because of IP WAN failure.
•
Dial plan issues: Directory' numbers are usually unique per site, but they can overlap
across multiple sites. Overlapping directory numbers and other issues,such as numbers that
are not consecutive, have to be solved by the design of a multisite dial plan. Various public
switched telephone network (PSTN)access codes in variouscountries arc anotherexample
of dial plan issues.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) u8 0
© 2010 Cisco Systems. Inc.
•
Network Address Translation (NAT) and security issues: Cisco Unified
Communications Manager and IP phones use IP primarily to communicate within the
enterprise. The use of private IP addresses is very common within the enterprise. When the
system should interact with a public IP network—for instance, when placing calls via an
Internet telephony service provider (ITSP)—then Cisco Unilied Communications Manager
and IP phone IP addresses have to be translated to public IP addresses. This translation
makes them visible on the Internet and, therefore, subject to attacks.
Note
The issue of vulnerability when IP addresses are translated to public IP addresses is not
limited to multisite deployments.
)2010 Cisco Systems, Inc.
Multisite Deployment Implementation
1-5
Quality Issues
This topic describes quality issues in a Cisco Unified Communications Manager multisite
deployment.
Quality Issues
IP networks are not designed to carry real-time traffic
• Packet-by-packet delivery
• Packets can take different paths.
No guarantee for correct order.
Problem is solved by RTP sequence numbers.
• Bandwidth shared by multiple users and applications
Unpredictable available bandwidth
Dunng peaks, packets need to be buffered in queues.
• Causes variable delays (jitter)
• Packets get dropped in case of buffer congestion.
• Likely on highly loaded links like IP WAN used between sites
in a multisite environment.
Jitter and packet drops impact voice quality
IP networks arc not designed to carry real-time traffic. Because of thenature of the network and
paeket-bv-packet delivery in which each packet could takea different path, there is no
guarantee that packets will arrive in thecorrect order at the destination. You can resolve this
issuebv using Real-Time Transport Protocol (RTP)sequence numbers.
Another issueis the fact that multiple usersand applications share the bandwidth, and the
actual required bandwidth varies significantly evenover short lapses of time. Therefore, the
bandwidth that is available for Cisco Unified Communications Manager traffic is
unpredictable. During peaks, packets need to bebuflered inqueues. Ifthecongestion occurs for
too long, buffers get tilled up and packets aredropped. Higher queuing delays and packet drops
aremore likely on highly loaded, slow links, such as WAN links that areused between sites in
a multisite environment. As a result, quality issues are common and need to be resolved by
implementing QoS. Otherwise, voice packets are subject tovariable delays (jitter) and packet
drops, bothof which impactvoice quality.
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Quality Issues Example: Jitter and Packet Drops
The figure illustrates how packets are queued during congestion.
Quality Issues Example: Jitter and
Packet Drops
During congestion, packets are buffered in queues. If
a queue is full, packetsget dropped.
Tail drops cause packet loss
Queuing delay causes jitter.
• Voice quality is reduced.
Voice quality is reduced.
During peaks, packets cannot be sent immediately because ofinterface congestion, so they have
tobe stored in a buffer ("queued"). The time that the packet waits in such aqueue isreferred to
as the "queuing delay." The length ofthis delay can vary widely. Ifthe queue is full, newly
received packets cannot be buffered, so they get dropped (this action is called "tail drop").
Without any special treatment ofvoice packets, such as a FIFO processing model, the resulting
jitter and packet lossdecrease voice quality.
)2010 Cisco Systems. Inc.
Multisite Deployment Implementation
Bandwidth Issues
This topicdescribes bandwidth issuesin a multisite environment.
Bandwidth Issues
Individual sites of a multisite deployment are
separated by an IP WAN:
* All intersite traffic (voice, data, video, etc.) competes for
available bandwidth.
« Bandwidth on IP WAN links is usually limited and costly.
Link bandwidth should be used as efficiently as possible.
No unnecessary trafficshould be sent over the IP WAN.
Default codec (G.711) is not efficient in noncircuit-based
environments.
Voice traffic causes lots of overhead.
• Large headers, small payload
• High packet rate
The individual sites of a multisite deploy ment areusually interconnected bv an IP WAN.
Bandwidth on WAN links is limited and relatively costly. Therefore, the goal is touse (he
available bandwidth asefficiently aspossible. Discourage unnecessary traffic, and consider
implemenling methods forbandwidth optimization.
Voice streams can he verv wasteful, considering that voice is sent in small packets but atavery
high packet rate. And they are particularly wasteful when using default codecs (G.711 requires'
64 kb/s for digitized voice). The overhead of packeli/ation—encapsulating digitized voice into
RTP. User Datagram Protocol (UDP). IP. and a Layer 2 protocol—is extremely high compared
to the size of the pavload. The more (tf these packets thai are sent, the more often the headers
are addedto the actual voice information (the RTPpayload).
Example: Bandwidth Issues
A voice call with default setting's uses the G.71 I codec, anda paeketization period of 20 ms.
The 160 bvtes (equivalent to 20msof voice) areencapsulated into a 12-byte RTP header, an 8
byte UDP header, anda 20-byte IPheader. The header-to-payload ratio is 1:4, and the
bandwidth requirement is 80 kb/s (instead of the 64-kb/s nominal bandwidth of G.711).
Note
This calculation considersonlyencapsulation to IP packets. Layer 2 encapsulation (which
vanes depending on the data linkprotocol being used) is not yet considered.
Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Bandwidth Issues Example: VDice and Data Traffic Competing
for Bandwidth
The example illustrates the higher ovdVh ead ofvoice packets when comparing them with file
transfer packets.
Bandwidth Issues Sample: Voice and
Data Traffic Compdt:ing for Bandwidth
In multisite environments, media services such as
music on hold (MOH), conferences, annunciators,
and so on, can cause considerable bandwidth
consumption over the IP WAN.
Voice packets:
• Small size
• High packet rate
• Large overhead
-wwr
Vo-ceand data packets are
competing for IP WAN bandwdlh
Data packets:
• Large size
• Lower packet rate
• Small overhead
As shown mthe figure and as already mentioned, voice packets consume lots of bandwidth that
is caused by the overhead ofIP, UDP. and RTP headers that are added to small packets and
sent at ahigh packet rate. Data packets such as afile transfer also add 40 bytes ofoverhead (20
bytes IP and 20 bytes TCP), bu, the payload is as large as possible (filling up the maximum
transrmsston umt. or MTU^ypically about 1500 bytes. Because of the large payload the
packet rate ,s also lower, and overhead is not added, as is often the case with voice packets.
Because ofthe inefficiency of voice packets, all unnecessary voice streams should be kept
away from .he IP WAN. Media resources, in particular, can be optimized in such away that
they do not have to cross the IP WAN all the time, thus conserving valuable bandwidth You
can achieve this optimization by utilizing local media resources.
>2010 Cisco Systems, Inc.
Multisite Deployment Implementation
1-9
Bandwidth Issues Example: Load Caused by Centralized Media
Services
The figure illustrates the bandwidth issue that is caused by acentralized conference bridge.
Bandwidth Issues Example: Load
Caused by Centralized Media Services
The use ofa centralized conference bridge requires RTP streams of
phones that are located at the remote site to cross the IP WAN, even
when only remote phones areparticipating in the conference.
The same problem applies to media termination points (MTPs),
annunciators, and MOH
Cisco Unified
Communications
Manager
in the example, aconference bridge has been deployed at the mam site. 1here >s no conference
bridge at the remote site. If remote IP phones join aconference, their Rl Pstreams are sent
across the WAN to the conference bridge. The conference bridge mixes the received audio
streams and then sends them back again to the IP phones over the IP WAN.
There arc three members in the conference in this example, and all of .hem are physically
locked at the remote site. In total, three RTP streams are flowing toward the coherence bndge.
and three RTP streams are flowing back to the remote site. Assuming ^f^ett.ngs each
RTP stream requires 80 kb/s (ignoring the Layer 2overhead), resulting in 240 kb/s of IP WAN
bandwidth that is required bv this voice conference. If the conference bndge was not located n
life olr side ofthe IP WAN. this traffic would have avoided the WAN link entirely, since all
participants ofthe conference are local to the remote site.
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>2010 Cisco Systems, Inc.
Availability Issues
This topic describes availability issues in Cisco Unified Communications Manager multisite
deploy ments.
Availability Issues
Availability of the IP network, especially of the IP WAN that
interconnects sites in a multisite deployment, is critical for
several services;
• Signaling
- Remote IPphones registering with centralized Cisco Unified
Communications Manager
- Remote MGCP gateways controlled bycentralized Cisco Unified
Communications Manager
- Signaling to othervblP domains
- CiscoUnified Communications Manager tnterclustertrunks
• SIP trunks
• H.225 trunks
• Media transmission
- RTP streams sent across the IP WAN
• Other services, such as IP phone XMLservices
When deploying Cisco Unified Communications Manager in multisite environments, you
access services over the IP WAN. Affected services include the following:
• Signaling in Cisco Unified Communications Manager multisite deployments with
centralized call processing: Remote IP phones register with acentralized Cisco Unified
Communications Manager server. Remote MGCP gateways are controlled by acentralized
Cisco Unified Communications Manager server thatacts asan MGCP call agent.
• Signaling in Cisco Unified Communications Manager multisite deployments with
distributed call processing: In such environments, sites are connected via H.323
(nongatekeeper-controlled. gatekeeper-controlled, or H.225) or Session Initiation Protocol
(SIP) trunks.
• Media exchange: RTP streams between endpoints that are located in different sites.
•
Otherservices: These services include IPphone XML services, access to applications
such asattendant console. Cisco Unified Communications Manager Assistant, and others.
Ifthe IP WAN connection is broken, these services are not available. This unavailability might
be acceptable for some services, but strategic applications such as telephone call services
should bemade available during WAN failure viabackup methods.
) 2010 Cisco Systems, Inc.
Multisite Deployment Implementation
1-11
Availability Issues Example: IP WAN Failure
The figure illustrates an example oflost services during WAN failure.
Availability Issues Example: IP
Failure
IP WAN failure impacts connection to the remote
cluster, phones at remote sites, and access to the
ITSP.
Remote Cluster
Cisco Unified
Communications
Manager
In the example, there is a main site with an inlerclusler trunk to a remote Cisco Unified
Communications Manager cluster. There is also aremote site with IP phones that register at the
Cisco Unified Communications Manager cluster that is located at the main site. ASIP trunk is
used to connect to an ITS!'.
Ifa WAN failure occurs, no calls to the other cluster orto the ITSP are possible. In addition, all
phones that arelocated at the remote site lose registration with Cisco Unified Communications
Manager, sothev do not operate atall. They cannot even place calls within the remote site.
Note
Adeployment as shown inthe exampleis badly designedbecause of the lack of IP WAN
backup.
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© 2010 Cisco Systems, Inc
Dial Plan Issues
This topic describes dial plan issues in Cisco Unified Communications Manager multisite
environments.
Dial Plan Issues
Dial plans for multisite deployments have to address
several issues:
* Direct inward dial versus attendants
• Overlapping directory numbers
• Nonconsecutive directory numbers
• Variable-length numbering and interdigit timeout handling
- Optimized call routing
- Toll bypass
- Tail-end hop-off
-
PSTN backup
• Various PSTN requirements in various countries
- Access codes for PSTN, national, and international dialing
- Number presentation (ISDN TON)
• Scalability
In a multisite deployment, dial plan design requires the consideration of several issues that do
not exist in single-site deployments:
•
Direct inward dialing (DID) ranges and E.164 addressing: When you are considering
integration with the PSTN, internally used directory numbers have to be related to external
PSTN numbers (E.164 addressing). Depending on the numbering plan (fixed or variable)
and services that are provided bv the PSTN, these solutions are common:
Each internal directory number relates to a fixed-length PSTN number: In this
case, each internal director; number has its own, dedicated PSTN number. The
directory number can, but does not have to, match the least significant digits of the
PSTN number. In countries with a fixed-numbering plan, such as the North
American Numbering Plan (NANP). the four-digit station codes, for instance, are
used as internal director;' numbers. If these numbers are not unique, digits of office
codes or administratively assigned site codes might be added, resulting in five or
more digits being used for internal directory numbers.
Another solution is not to reuse any digits of the PSTN number, but to simply map
each internally used director.' number lo any PSTN number that is assigned to the
company. In this case, the internal and external numbers do not have anything in
common.
©2010 Cisco Systems. Inc.
Multisite Deployment Implementation
If the internally useddirector;' numbermatches the least significant digits of its
corresponding PSTN number, you can set significant digits at the gateway or trunk.
Then you can also configure general external phone number masks, transformation
masks, or prefixes, because all internal director; numbers are changed to fully
qualified PSTN numbers in the same way. The internal directory number can be
composed ot parts of the PSTN number and administratively assigned digits. In that
case, one or more translation rules have to be used for incoming calls, and one or
more calling-party transformation rules (transformation masks, external phone
number masks, prefixes, and so on) have to be configured.
An internal director; number can be composed of site codes with PSTN station
codes, site codes with various ranges (such as PSTN station codes 4100 to 4180 that
map to director; numbers 1100 to 1180), or site codes with completely independent
mappings of internal directory numbers to PSTN numbers.
—
No direct inward dialing support in fixed-length numbering plans: To avoid the
requirement of one PSTN number per internal directory number when using a fixedlength numbering plan, it is common to not allow direct inward dial to an extension.
Instead, the PSTN trunk has a single number, and all PSTN calls that are routed to
that number arc sent to an attendant (or an autoattendant) from where the calls are
transferred to the appropriate internal extension.
Internal director; numbers arc part of a variable-length number: In countries
with variable-length numbering plans, a (usually shorter) '"subscriber" number is
assigned to the PSTN trunk. However, the PSTN routes all calls starting with this
number to the trunk: the caller can add more digits to identify the extension. There is
no fixed number of additional digits or total digits (there is a maximum, however),
which provides the freedom to select the length of directory numbers. A caller
simplv adds the appropriate extension when placing a call to a specific user to the
company's (short) PSTN number. If only the short PSTN number without an
extension is dialed, the call is routed to an attendant within the company. Residential
PSTN numbers are usuall; longer and do not allow additional digits to be added.
The feature that is described here is available only on trunks.
Overlapping numbers: Users that are located at different sites can have the same director;'
numbers assigned. Because director;' numbers are usually unique only within a site, a
multisite deployment requires a solution for overlapping numbers.
Nonconsccutive numbers: Continuous ranges of numbers are important for summarization
of call-routing information. Such blocks can be represented by one or a few entries in a
call-routing table (route patterns, dial-peer destination patterns, voice translation rules, and
so on) and can keep the routing table short and simple. If each endpoint requires its own
entry in the call-routing table, the table gets too large, lots of memory is required, and
lookups take more time. Therefore, nonconsecutive numbers (somenumbers at one site,
and other numbers of the same block at a different site) are not optimal for efficient call
routing.
Variable-length numbering: Some countries have fixed-length numbering plans for
PSTN numbers, while others have variable-length numbering plans. A problem of variable-
length numbers is thatthe length can be determined only by waiting fora timeout. If no
moredigits have been dialed for the specifiedtime, the numberis considered to be
complete. Waiting for the timeout adds to the postdial delay.
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©2010 Cisco Systems. Inc
•
•
Optimized call routing: An IP WAN between sites with PSTN access allows PSTN toll
bypass. Cisco Unified Communications Manager servers route calls between sites over the
IP WAN instead of using the PSTN (toll bypass). In such scenarios, the PSTN should be
used as a backup path only in the case of WAN failure. Another solution, which extends the
idea of toll bypass,is to use the IP WAN also for PSTN calls: With tail-endhop-off
(THHO), the IP WAN is used as much as possible, and the gateway that is closest to the
dialed PSTN destination is used for the PSTN breakout. Finally, a backup path over the
PSTN should be enabled for when a call cannot be sent over the IP WAN (for example, if
the IP WAN is down or the maximum number of allowed calls is reached).
Various PSTN requirements: Various countries—and sometimes even various PSTN
providers withinthe same country—can have variousrequirements regarding the PSTN
dial rules. This situation can cause issueswhen calls can be routedvia multiplegateways,
For example, if the requirements of a primary gateway are different from the requirements
of a backup gateway, numbers have to be transformed accordingly.
p,„
The calling number (the Automatic Number Identification, or ANI) ofcalls that are being
^
received from the PSTN can be represented in various ways: as a7-digit subscriber
number, as a 10-digitnumber including the area code, or in international format with the
country code in front of the area code. To standardize the calling number for all calls, the
format that is used mustbe known, and the number has to be transformed accordingly. In
countries where PSTN numbers do not have fixed lengths, it is impossible to detect the type
(local, national, or international) of the number by the number only by looking at its length.
Insuch cases, thetype of number has to be specified in signaling messages (forexample,
by the ISDN type of number, or TON).
•
Scalability: In large or very large deployments, dialplanscalability issues arise. When
interconnecting multiple Cisco Unified Communications Manager clusters or CiscoUnified
Communications Manager Express routers viatrunks, it is difficult to implement a dial plan
on an any-to-any basis where each device or cluster needs to know which numbers or
prefixes are found at which other system. In addition to the need to enter almost the same
dial plan ateach system, a static configuration does not reflect true reachability. Ifthere are
any changes, the dial plans ateach system have to be updated. Although there aresolutions
thatallow centralized dial plan configuration (forexample, H.323 gatekeepers), in very
large deployments a dynamic discovery ofdirectory number ranges and prefixes would
simplify the implementation and provide a more scalable solution.
)2010 Cisco Systems. Inc.
Multisj,e Dep|0ymen| implementation
M5
Example: Variable-Length Numbering, E.164 Addressing, and
DID
The figure illustrates an international deployment with variousnumbering schemes.
Example: Variable-Length Numbering,
E.164 Addressing, and
NANP.
No DID
Variable-Length E.164
Aulo-Attendanl Used
Addressing with DID
Cisco Unified
Communications
Manager
"fhe example features a main site in the United Slates. The NANP PSTN number is408 5551234. DID isnotused. All calls placed to themain site are managed by anattendant. There is a
remote site in Gennanv with a PSf N numberof 149 404 13267. The Cierman location uses
four-digit extensions, and DID isallowed, since digits can be added to the PSTN number.
When calling the German office attendant (not knowing aspecific extension), users in the
United States would dial +9 011 49 404 13267. If they know that they want to contact
extension 1001 dircclK. thev would dial+9 011 49 404 13267 1001.
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Implementing Cisco Unmeet Communications Manager. Part 2 (CIPT2) v8.0
©2010 Cisco Systems.
Fixed vs. Variable-Length Numbering Plans
The table compares characteristics of a fixed numbering plan with characteristics of a variablelength numbering plan.
Fixed vs. Variable-Length Numbering Plans
Examples:
• Witriir It S : "9 1 408 555-1234" or '1 555-1234" twilrifi Bia same area code)
•
U.S to Austria1 "9 011 43 1 12345"
• WitHii Austria: "0 0 1 12345" or "0 1234" (withn the same area code)
• Austria to U.S.. "0 00 1 406 555-1234 " (1 is country code, not national access code)
Components
,
Example
_,
VsriSUle-Length Numbing
"
Country code
1
43
3 digit
1-4 (Halt*
3-tfio.texchsng» cads
♦
3«mor»-<Igflt
^cfigistaawtcoas
PSTN access code
9
National access code
1
International access code
*
Avetiia
MAW
Area code
Subscriber numoer
_ „
£|*ea Numbering Plan
011
"~
0
0
00 or*
(•+' uwd by e#Jpfion«)
A fixed numbering plan, such as the plan used in North America, features fixed-length area
codes and local numbers. An open numbering plan, such as the plan used in various countries
that have not yet standardized their numbering plans, features variance in the length of the area
code or the local number, or both.
A country code is used to reach the particular telephone system for each country or special
service.
An area code is used within many countries to route calls to a particular city, region, or special
service. Depending on the country or region, it may also be referred to as a Numbering Plan
Area (NPA), subscriber trunk dialing code, national destination code, or routing code.
The subscriber number represents the specific telephone number to be dialed, but does not
include the country code, area code (if applicable), international prefix, or trunk prefix.
A trunk prefix refers to the initial digits to be dialed in a call within the United States,
preceding the area code and the subscriber number.
An international prefix is the code to be dialed before an international number (the country
code, the area code if any, and then the subscriber number).
The table contraststhe NANP and a variable-length numbering plan (the Austrian numbering
plan, in this example).
>2010 Cisco Systems, Irrc
Multisite Deployment Implementation
1-17
Detection of End of Dialing in Variable-Length Numbering
Plans
There are three ways of detecting end of dialing in variable-length numbering plans.
Detection of End of Dialing in VariableLength Numbering Plans
There are three ways to detect end of dialing in variablelength numbering plans:
• IntErdigit timeout
- Simple to configure
Least convenient
• Use of # key
Different implementation in Cisco IOS Software (simple) versus
Cisco Unified Communications Manager (complex)
-
Convenient
Requires users to be informed aboutthis option
• Use of overlap sending and overlap receiving
-- Convenient
-- Must be supported by PSTN
- Complex implementation, especially when differentPSTN calling
privileges are used
From an implementation perspective, the simplest way to detect end of dialing is to wait for an
interdigit timeout to expire.This approach, however, provides the least comfort to the end user
because it adds postdial delay. In an environment with only few numbers of variable length (for
example, the NANP. where only international calls are of variable length), waiting for the
interdigit timeout ma\ be acceptable. However, even in such an environment, it may make
sense to at least reduce the value of the timer, because the default value in Cisco Unified
Communications Manager is rather high (15 s).
Note
In Cisco Unified Communications Manager, the interdigit timer is set by the dusterwide
Cisco CallManager service parameter T302 timer that is found under Device > General.
In Cisco IOS Software, the default for the inlerdigit timeout is 10 seconds. You can modifj this
value using the voice port timeouts interdigit command.
Another solution for detecting end of dialing on variable-length numbers is the use of the # ke\.
An end usercan press the # kej to indicate thatdialing hasfinished. The implementation of the
# kev is different in Cisco Unified Communications Manager versus Cisco IOS Software. In
Cisco IOS gatewa\s. the # is seen as an instruction to stop digit collection. It is not seen as part
of the dialed string. Therefore, the # is not part of the configured destination pattern. In Cisco
Unified Communications Manager, the # is considered to be part of the dialed number, and
therefore its usage hasto be explicitly permitted by the administrator by creating patterns that
include the it. If a pattern includes theit. the # hasto be used; ifa pattern doesnot include the#.
the pattern is not matched if the user pressed the # key. Therefore, it is common in Cisco
Unified Communications Manager to createa variable-length patterntwice: once with the it at
the end and once without the #.
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An alternative wayof configuring such patterns is to end the pattern with ![0-9#]. In thiscase,
a single pattern supports bothwaysof dialing: withthe # and without the #. However, be aware
ff^
that the use ofsuch patterns can introduce other issues. For example, when using discard digits
instructions that includeTrailing-^(for example, PreDot-Trailing-#). This discard digit
instruction will have an effectonly when there is a trailing # in the dialed number. If the # was
p.,*
not used, the discard digit instruction is ignored and hence the PreDot component ofthe discard
IM
digit instruction is also not performed.
Allowing the use of the # to indicate end of dialing providesmore comfortto end users than
having them wait forthe interdigit timeout. However, thispossibility has to be communicated
to the end users, and it shouldbe implemented consistently. As mentioned earlier, it is
automatically permitted in Cisco IOS Softwarebut not in Cisco UnifiedCommunications
Manager.
The third way to indicate end of dialing is the use of overlapsend and overlapreceive.If
overlap is supported end-to-end, the digits that are dialed by the end user are sent one by one
over the signaling path. Then the receiving end system can inform the calling device once it has
received enough digits to route the call (number complete). Overlap send and receive is
common in some European countries such as Germany and Austria. From a dial plan
implementation perspective, overlap send and receive is difficult to implement when different
jte
PSTN callingprivileges are desired. In this case, you have to collectenoughdigits locally(for
example, in Cisco Unified Communications Manager or Cisco IOS Software) to be able to
decide to permit or deny the call. Only then can you start passing digits on to the PSTN one by
one using overlap. For the end user, however, overlap send and receive is very comfortable,
pw
§^
because each call is processed as soon asenough digits have been dialed. Thenumber of digits
that are sufficient varies perdialed PSTN number. Forexample, one local PSTN destination
may be reachable by a seven-digit number, whereas another local number may be uniquely
identified only after receiving nine digits.
>2010 Cisco Systems, Inc.
L
Multisite Deployment Implementation
1-19
Optimized Call Routing and PSTN Backup
Using an 1P WAN enables savings on the cost of PSfN calls in a multisite en\ ironment.
Optimized Call Routing and PSTN
Backup
In multisite environments, using the IP WAN for routing
calls to remote destinations allows PSTN cost savings:
• Toll bypass
Calls between sites use the IP WAN instead of PSTN.
PSTN is used as backup in case of IP WAN or CAC failure.
* Tail-end hop-off (TEHO)
- TEHO extends the conceptrjf toll bypass.
• Calls to remote PSTN locations use the IP WAN as much as
possible.
• PSTN breakout occurs at gateway located closestto the
PSTN destination.
Local PSTN breakout is used as backup in case of IP WAN or
CAC failure
"there are two wa\s to save costs for PSTN calls in a multisite deployment:
•
•
Toll bypass: Calls between sites that use the IP WAN instead of the PSTN are toll-bypass
calls. The PSTN is used only when calls over the IP WAN are not possible—either because
of a WAN failure or because the call is not admitted by Call Admission Control (CAC).
TEHO: TEHO extends the concept of toll bypass by also using the IP WAN for calls to the
remote destinations in the PS'fN. With TIT 10. the IP WAN is used as much as possible and
PSTN breakout occurs at the gateway that is located closest to the dialed PSTN destination.
Local PSTN breakout is used as a backup in case of an IP WAN or CAC failure.
Caution
Some countries do not allow the use of TEHO. When implementing TEHO, ensure that the
deployment complies with legal requirements.
When using the IP WAN lo reach remote PS'fN destinations or internal director) numbers at a
different site, it is important to consider backup paths. When the IP WAN is down or when not
enough bandwidth is available for an additional voice call, calls should be routed via the local
PSTN gateways as a backup path.
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Implementing Cisco Unified Communications Manager, Part 2 (CIPT2)v8 0
©2010 Cisco Systems. Inc
Example: TEHO
The figure illustrates the use of TEHO in a multisite deployment.
Example: TEHO
Chicago
San Jose
l^.
403-555-6666
In the example, a call from Chicago to San Jose, California, would berouted inthe following
way:
1. A Chicago userdials9 1 408 555-6666, the number fora PSTN phone thatis located in
San Jose.
2. Thecall is routed from theCisco Unified Communications Manager Express in Chicago to
the Cisco Unified Communications Managercluster in San Jose over the IP WAN.
3. The Cisco Unified Communications Managerin San Jose routes the call to the San Jose
gateway, which breaks out to the PSTN with a (now) local call to the San Jose PSTN.
4. The San Jose PSTN phone rings.
IM
'2010 Cisco Systems, Inc.
Multisite Deployment Implementation
1-21
Example: Overlapping and Nonconsecutive Numbers
The figure illustrates a deployment with overlapping and nonconsecutive numbers.
Example: Overlapping and
Nonconsecutive Numbers
Remote Site
Wain Site
Cisco Unified
Communications
Manager
••1001-1099
1001-1099 •*
Nonconsecutive
2000-2157 +•
2365-2999
Numbers
-+•
2158-2364
Inthe example. IP phones ai the main site use directorv' numbers 1001 to 1099. 2000 to 2157.
and 2365 to 2999. At the remote site. 1001 to 1099 and 2158 to 2364 are used. There are two
issueswith thesedirector, numbers: 1001 to 1099overlap; and thesedirectory numbers exist at
both sites, so they are notunique throughout the complete deployment. Inaddition, the
nonconsecutive use ofthe range 2000 to 2999 (some numbers outof this range areused at the
remote site, and the others are used at the main site) wouldrequire lots of entries in call-routing
tables, since the ranges can hardly be summarized by one (ora few) entries.
Note
The solutions to the problems that are listed inthis lesson are discussed inmoredetail inthe
next lesson of this module.
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Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v80
© 2010 Cisco Systems, Inc.
ttM.
Various PSTN Requirements
Various countries can have various PSTN requirements, which make it difficult to implement
dial plans in international deployments.
Various PSTN Requirements
Various countries have various requirements for
PSTN calls:
• Dial rules for the called-party numberon outbound PSTN
calls
-
PSTN access code
-
National access code
-
International access code
• Presentation of called- and calling-party numbers on inbound
and outbound calls
- Length of number and its components
- ISDN number types
- Overlap send and overlap receive
- + prefix on E.164 numbers
• Emergency dialing
One of the issues in international deployments is various PSTN dial rules. For example, in the
United States, the PSTN access code is 9, while in most countries in Europe, 0 is used as the
PSTN code. The national access code in the United States is I, while 0 is commonly used in
Europe. The international access code is 011 in the United States,while 00 is used in many
European countries. Some PSTN provider networks require the use of the ISDN TON, while
others do not support it. Some networks allow national or international access codes to be
combined with ISDN TON. Others require you to send the actual number only (that is, without
any access codes) when setting the ISDN TON.
**»
Thesame principle applies to the calling-party number. As mentioned earlier, in variable-length
numbering plans, the TON cannot be detectedby its length.Therefore, the only way to
determine whetherthe receivedcall is a local,national, or international call is by relying on the
availability of the TON information in the receivedsignalingmessage.
Some countries thathavevariable-length numbering plansuseoverlap sendandoverlap
receive. With overlap send, a numberthat is dialed by an end user is passedon to the PSTN
digit by digit. Then the PSTN willindicate when it has received enough numbers to routethe
call. Overlap receivedescribes the same conceptin the oppositedirection:when a call is
received from the PSTN in overlap mode, the dialed number is delivered digit by digit, andnot
en bloc. Some providers thatuse overlap sendtoward theircustomers do notsend the prefix
that is configured forthe customer trunk, but only the additional digitsthat are dialed by the
user who initiates the call.
When dialing PSTN numbers in E.164 format (that is,numbers thatstart with the country
code), the + sign is commonly prefixed to indicate that the number is in E.164 format. The
advantage of using the+ sign as a prefix forinternational numbers is thatit is commonly
known as a + sign around the world. Incontrast, PSTN access codes suchas 011 (used in the
NANP) or 00 (often used in Europe) areknown onlyin the respective countries.
>2010 Cisco Systems. Inc.
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Multisite Deployment Implementation
1-23
Finally, emergency dialing can be an issue in international deployments. As various countries
have various emergenc\ numbers and various ways to place emergency calls, users are not sure
how to dial the emergency number when roaming to other countries. An international
deployment should allow roaming users to use their home dialing rules when placing
emergency calls. The system should then modify the called number as required at the respective
site.
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Implementing Cisco Unified Communications Manager. Part 2(CIPT2) v8.0
©2010 Cisco Systems, Inc.
Issues Caused by Different PSTN Dialing
Different local PSTN dial rules can cause several issues, especially in international
deployments.
Issues Caused by Different PSTN
Dialing
How to store PSTN contacts so that they can be used
from any site?
• Different ways to storeor configure PSTN destinations;
- Speed dials
-
Fast dials
-
Address book entries
-
Call lists
- Automated alternate routing (AAR) targets
- Call Forward destinations
• Stored numbercan be used at multiple sites (countries) because of
roaming users using local PSTN gateways.
- Cisco Extension Mobility
- Cisco Device Mobility
- PSTN backup
- Tail-end hop-off and Least Cost Routing (LCR)
The main problem that needs to be solved in international environments is how telephone
numbers ofcontacts should bestored. Address book entries, speed and fast dials, call list
entries, and other numbers should be in a format that allows them tobe used atany site,
regardless of the local dial rules that apply to the site where the user is currently located.
The same principle applies to numbers that are configured by the administrator—for example,
the target PSTN number for automated alternate routing (AAR) targets. Call-forwarding
destinations should also be in a universal format—a format that allows the configured number
to be used atany site. The main reason for auniversal format is that amultisite deployment has
several features'that make itdifficult to predict which gateway will be used for the call. For
example, aroaming user may use Cisco Extension Mobility or Device Mobility. Both features
allow an end user toutilize local PSTN gateways while roaming. Ifno universal format isused
to store speed dials or address book entries, itwill be difficult for the end user to place aPSTN
call to a number that was stored according to the NANP dial rules while incountries that
require different dial rules. Even when not roaming, the end user can use TEHO orLeast Cost
Routing (LCR). so that calls break out to the PSTN at aremote gateway, not at the local
gateway. Ifthe IP WAN link to the remote gateway is down, the local gateway is usually used
as abackup. How should the number that isused for call routing look in such an environment?
It isclearly entered according to local dial rules by the end user, but ideally itischanged to a
universal format before call routing is performed. Once thecall has been routed and the egress
gateway has been selected, the number could then be changed as required by the egress
gateway.
© 2010 Cisco Systems. Inc.
Multisite Deployment Implementation
1-25
Dial Plan Scalability Issues
In large Cisco Unified Communications Manager deployments, it can be difficult to implement
dial plans, especially when using features such as TEHO with local PS'fN backup.
Dial Plan Scalability Issues
Dial plans are difficult to implement in large Cisco Unified
Communications deployments
• Static configuration for multiple sites ordomains isvery complex
because of any-to-any call-routing requirements.
• Centralized H.323 gatekeepersor SIP network services offer dial
plan simplification.
- Less configuration because ofany-to-one call-routing topology.
• Static configuration nevertheless.
• No dynamic recognition of routes
• No automatic PSTN rerouting
No built-in redundancy
• Still no optimal solution for large deployments
- Call Control Discovery allows dynamic learning of dial plans.
The main scalability issue oflarge deployments is that each call routing domain (for example, a
Cisco Unified Communications Manager cluster or aCisco Unified Communications Manager
Express router) needs to be aware of how to getto all other domains.
Such adial plan can become very large and complex, especially when multiple paths (for
example, a backup path for I'EHO) have to be made available. As each call routing domain has
to be aware ofthe complete dial plan, astatic configuration does not scale. For example, anv
changes in the dial plan have to be applied individually at each call routing domain.
Centralized H.323 gatekeepers orSIP network services can be used to simplify ihe
implementation ofsuch dial plans, because there is no need toimplement the complete dial
plan at each call routing domain. Instead ofan any-to-any dial-plan configuration, only ihe
centralized component has tobe aware ofwhere lo find which number. This approach,
however, means thai vou rely on a centralized service. Ifthe individual call-routing entities
have noconnectivity to the centralized call-routing intelligence, all calls would fail. Further, the
configuration isstill static. Any changes atone call-routing domain (for example, new PSTN
prefixes because of changing the PS'fN provider) have to be implemented also at the central
call-routing component.
In addition, these centralized call-routing services do not have built-in redundancy.
Redundancv can be provided, but requires additional hardware, additional configuration, and so
on. Redundancv is not an integrated part of the solution.
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ImplementingCisco Unified Communications Manager. Part 2 (CIPT2]vB.O
©2010 Cisco Systems, Inc.
Answer Key
The correct answers and expected solutions for the activities that are described in this guide
appear here.
Lab 1-1 Answer Key: Implementing Basic Multisite
Connections
fhe solution is part of the activity procedure and verification.
Lab 1-2 Answer Key: Implementing a Dial Plan for International
Multisite Deployments
fhe solution ispart of the activity procedure and verification.
Lab 2-1 Answer Key: Implementing SRST and MGCP Fallback
fhe solution ispari of the activity procedure and verification.
Lab 2-2 Answer Key: Implementing Cisco Unified
Communications Manager Express in SRST Mode
The solution ispart of the activity procedure and verification.
Lab 3-1 Answer Key: Implementing Bandwidth Management
The solution is part ofthe activity procedure and verification.
Lab 3-2 Answer Key: Implementing CAC
The solution ispart of the activity procedure and verification.
Lab 4-1 Answer Key: Implementing Device Mobility
The solution ispart of the activity procedure and verification.
Lab 4-2 Answer Key: Implementing Cisco Extension Mobility
fhesolution is part of the activity procedure and verification.
Lab 5-1 Answer Key: Implementing SAF and CCD
The solution ispart of the activity procedure and verification.
Implementing Cisco Unrfied Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems, Inc
router eigrp SAF
service-family ipv4 autonomous-system 2
shutdown
Step 13
Verify that the IP path to all learned patterns is marked unreachable by using the
show voice saf dndb all command.
Activity Verification
You have completed this task when you attain these results:
• The BR-.t router learns patterns by CCD as described in the activity procedure.
. Verifv that calls can be placed to patterns learned by CCD while being in SRST and MGCP
fallback mode.
Step 14
Place calls from your BR-.v site to all other three sites.
• Verifv the path that each call takes by using the debug isdn q931 command at
all four gateways.
• Use site code dialing for calls to the other pod: 8-5 lv-200! or 8-5I r-2002 to
IIQ-v phones and 8-52.V-3001 to the BR-v phone.
• Use four-digit extensions for calls to the local HQ-.t phones: 2001 or 2002.
Note
When acall is placed to aphone located at one of the BR sites, the SIP INVITE is sent to
the advertising device (CUCM1-X or CUCM1-y). As the BR sites are in SRST mode, Cisco
Unified Communications Managers will route the calls to their BR gateway via the PSTN
basedon the existing CFUR configuration.
m
Lab Guide
© 2010 Cisco Systems. Inc.
95
Note
This dial peer is required for the CCD PSTN backup calls. The learned toDID rule changes
the internally used site-code numbers from 8 followed by seven digits to E.164 numbers with
a plus prefix The CCD PSTN backup calls then match this dial peer where the + is stripped
and then 011 is prefixed.
The learned pattern 851x-2XXX refers to the HQ-x site. Until now calls between the two
sites of the same pod used four-digit dialing. CCD cannot advertise the HQ-x site with 4
digits to the BR-x SAF client and with site code prefixes to other SAF clients (the SAF clients
of the other pod). Usually, when using CCD, the internally used pattern for a given site has
to be the same at all sites.
However, the problem can also be solved in a way that allows users to continue using 4
digits for intersite calls within the same pod. You need to modify the dialed four- digit number
2XXX at the BR-x router to 851x2XXX before the outbound dial peer is selected.
Step 10
On the BR-.rrouter configure the following number expansion in order to allow
BR-.v users to continue using four-digit intersite dialing towards the HQ site of the
local pod (HQ-.t):
num-exp 2...
Note
851x2...
Byconfiguring the number expansion command calls to four-digit numbers starting with 2
(for example 2001) are expanded to site code dialing format (851x2001) before the selection
of the outgoing dial peer. The expanded number is used to select the outgoing dial peer
(dial-peer 8 in this case) which refers to SAF-learned patterns. The BR-xgateway finds a
match in a learned pattern (851x2XXX) that is currently marked unavailable. Therefore, a
PSTN backup call is placed using the learned toDID rule (4:+5551x555). The resulting call to
•••5551x5552001
matches dial peer 999, which sends the call to the PSTN with a called
number of 0115551x5552001.
Simulate IP WAN Failure Between the HQ and the BR Sites
In this section \ou will break MGCP, SCCP. and SAF to simulate an IP WAN failure between
the HQ and BR sites of your pod.
Step11
You hav e to force the BR-.v router into SRST and MGCP fallback mode by breaking
connectivity between the BRsite andCiscoUnified Communications Manager. You
can do that by reapplying the access-list that was already used in an earlier lab task
at the HQ-.r router:
i
interface serial
...
ip access-group 100 in
Note
Step12
Use the interface that connects the HQ-x route with the BR-x router.
Break the SAT connection between your HQ router (HQ-.v) and your BR router
(BR-.r) in order to have the IP palhs marked unreachable by enleritig the following
commands in global configuration mode of the \K)-x router:
94
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®2010 Cisco Systems. Inc.
-Jfe
Note
The learned pattern 852x-3XXX refers to the BR-x site itself.This pattern will not be used at
a phone that is located at the BR-x site because four-digit dialing is used for internal calls. If
it was dialed, the call cannot use the advertised IP path (to Cisco Unified Communications
Manager CUCM1-x) because the IP WAN link is down. Remember that the BR-x gateway
normallydoes not use a local dial plan as it is configured as an MGCP gateway. It will only
look to its local call routing table when the connection to the HQ site is broken, if a BR-x
user dials a number out of the 8-52x-3XXX range during SRST mode, the BR-x gateway
finds a learned pattern that is currently marked unavailable Therefore, the PSTN backup
path (toDID 4:+6652x555) is used to place a PSTN backup call. When this call is set up by
the BR-xgateway, the PSTN routes the call back to the BR-xgateway. This means that
calling to the own site by site-code dialing works, but it would use two ISDN circuits as the
call is hairpinned at the PSTN
Enable Call Routing for CCD-learned Patterns and for CCD PSTN Backup Calls
Step9
Onthe BR-\ router configure the following dial peersin orderto enable the gatewav
to use CCD-learned patterns and to enable CCD PSTN backupcalls:
dial-peer voice 8 voip
destination-pattern 8.
session target saf
Note
Thisdial peer instructs Cisco Unified Communications Manager Expressto look to the CCDlearned patterns when a user diats 8 followed by seven digits.
The learned patterns 851y-2XXX and 852y-3XXX refer to the other pod. The IP destination
for both patterns is the Cisco Unified Communications Manager of the other pod
(CUCM1-y) However, as the IP WAN is down, the PSTN backup path has to be used
Based on the learned toDID rules (4:+5551y555 for pattern 851y-2XXX and 4:+6652y555 for
pattern 852y-3XXX) the BR-x gateway will placea directcall to the respective gatewayof
the other pod (HQ-yor BR-y).
dial-peer voice 999
potE
destination-pattern +T
voice-port 0/0/0:23
prefix Oil
) 2010 Cisco Systems. Inc
Lab Guide
exit-service-family
i
Step S
Configure the SAF Client function on your BR-jr router:
voice
service
saf
profile trunk-route 1
session protocol sip interface FastEthernetO/0 transport tcp port
5C6C
channel 1 vrouter SAF asystem 1
subscribe callcontrol
Step 6
wildcarded
Configure outbound dial peers for intersite calls to use SAP:
dial-peer voice 8 voip
destination-pattern 8
session target saf
dial-peer voice 2 voip
destination-pattern 2...
session target saf
Verify CCD-learned Patterns
Step 7
On the BR-.xrouter enter the show voice saf dndb all command lo view learned
routes. You should have learned a pattern for each of the four sites.
Step 8
On the BR-.v router enter the show voice saf dndb detail 851a2XXX command lo
view details of this specificlearned route. Repeat the command for the other three
teamed patterns.
Caution
Note
The X in the pattern of the show voice saf dndb detailcommand is case sensitive.
In each pod, Cisco Unified Communications Manager is the call agentthatadvertises the
HQ-x and BR-x patterns. Therefore, at the BR-x router 851x-2XXX and 852X-3XXX should
be listed as reachable bySIPat 10.x 1.1 while 851y-2XXX and 852y-3XXX should be listed
as reachable by SIP at 10.y.1.1.
The BR-x router will never use thelearned SIP path. As long as there isnoIPconnectivity
problem, the ISDN PRl is MGCP-controlled and the BR-x phones are registered to Cisco
Unified Communications Manager. Therefore no call routing occurs at the BR-x gateway
under normal situation.
The learned patterns are only used incase ofIPWAN failure. In this case, however, the IP
path ofthe learned patterns ismarked unavailable andthe BR-x gateway which then
operates in SRSTand MGCP fallback mode will use the learned patterns to placeCCD
PSTN backup calls based on the learned toDID rules.
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems, Inc.
Step 10
mm
Note
Repeat the test calls. Verify thatthe callsarererouted via the local PS'fN gatewa;
b\ usingdebug isdn q931 command at bothgateways.
Be aware of the call flow in this scenario. After calling a phone of the other pod using site-
code dialing a CCD-learnedpattern is matched. The SAF-enabledSIP trunk is not useO
because the IP path is marked unreachable (dueto shutting down SAF). The ToDID rule is
applied to the matched pattern and a call to the {now globalized) number is placed using the
*••
AAR CSS.
The call matches the intersite pattern, which refers to the non-SAF-enabled SIP trunk as first
option and to the local routegroupas second option. As ihe first option does notwork (due
tothe access list), the call is finally sent via the PSTN using the local gateway.
"•*
imm
Step 11
Re-enable the SAF connection between the two HQ routers by entering the
following commands in global configuration mode at your HQ~.v router:
<**
router eigrp SAF
mm
service-family ipv4 autonomous-system 2
no
Step 12
shutdown
Remove the access listfrom the serial interface at yourIIQ-.v router.
*"
Task 3: Configure Cisco Unified Communications Manager
^
Express SRST on Branch Router to Learn Routes Using CCD
m
In this task, you will configure Cisco Unified Communications Manager Express SRST on the
BR-.v router to learn routes from SAF using CCD. The learned patterns arc used for intersite
calls when the gateway is in SRST and MGCP fallback mode.
*
Activity Procedure
Complete these steps:
wm
Step 1
Open a Telnet session to your BR-x router (10.v.5.102).
Step 2
loiter the configure terminal command lo access the global configuration mode.
Make the Statically Configured Call Routes at the Cisco Unified Communications Manager
**»
Express SRST Router Inaccessible
Step 3
<mm
Shutdown ihe following dial peers:
•
Dial peer8y2 to destination pattern 85ly2...
•
Dial peer8v3 to destination pattern 852y3...
•
Dial peer2000to destination pattern 2..,
mm
—
Configure the SRST Router toSubscribe toSAF in Order to Learn Routes toOther Sites
Step 4
mm
Configure the SAF Forwarder function onyour BR-x router:
router eigrp SAF
-mm
service-family ipv4 autonomous-system 1
-imt
topology base
exit -sf -topology-
& 2010 Cisco Systems, Inc
Lab Guide
(CUCMl-v) by reapplying the access-list that was already used in earlier lab
exercises in global configuration mode at your HQ-x router:
interface serial
.. .
ip access-group 100 in
Note
Use the interface that connects the HQ-x route with the HQ-y router.
Caution
IP connectivity between the two Cisco Unified Communications Managers needs to be
broken in order to avoid that the CCD PSTN backup call is sent over the (non-SAF-enabled)
SIP trunk that has been created in an earlier lab exercise. Remember that intersite calls
placed to the other pod are not sent via the PSTN but via the IP WAN. The same applies to
TEHO calls placed to PSTN destinations attached to the HQ and BR gateways that are
located in the other pod. By applying the access list the SIP trunk is not operational anymore
and intersite calls placed to the other pod will use the PSTN as a backup path.
Make sure that the access list is applied at the HQ router of both pods. If the access list is
only applied at your local HQ router, you will experience very high post-dial delays when you
try to reach the other pod.
The reason for the post-dial delay is that the originating Cisco Unified Communications
Manager has to wait for a timeout when not being notified that the packet has been dropped.
Ifthe HQ router of the other pod does not drop the SIP INVITE sent by the local Cisco
Unified Communications Manager, then only the response packet that is originated by the
Cisco Unified Communications Manager of the other pod is dropped inbound at your HQ
router In this case only the Cisco Unified Communications Manager of the other pod is
notified that its response packet was dropped. The local, originating Cisco Unified
Communications Manager is not aware of any packet drops and hence has to wait for the
timeout to expire. After timeout expiration it willretry the call setup and the timeout will have
to be waited for again.
When both HQ routers are configured with the inbound access list at their interconnecting
serial interface, then Cisco Unified Communications Manager will always be immediately
aware of the network issue (packet drop) and switch over to the PSTN backup path without
additional delay.
•
Break the SAF connection between your HQ router (I IQ-.v) and the IIQ router of
the other pod (HQ-v) by enteringthe following commands in global
configuration mode at your HQ-x router:
router eigrp SAF
service-family ipv4 autonomous-system 2
shutdown
Step 9
Use Cisco Unitied RTMT to verity that the IP paths of the learned patternsare
marked unreachable.
implementing Cisco Unified Communications Manager, Part2 (CIPT2) u8.0
©2010 Cisco Systems, Inc.
Step 32
Move the SAFSlPTrunk to Selected SAF Trunks.
Step 33
Check the Activated Feature check box and click Save.
Activity Verification
You have completed this task when you attain these results:
•
Verify registration of the external SAF client.
Step 1
•
from the I IQ-.v router, enter the show eigrp service-family ipv4 clients command
in privilege mode to verify that Cisco Unified Communications Manager has
registered with the SAF forwarder.
Verify the learned patterns by using RTMT.
Step 2
Install and launch Cisco Unified Real Time Monitoring Tool (RTMT) on your
student PC.
•
Navigate to Applications > Plugins and click Find.
•
Click the Download Link next to the Cisco Unified CM Real-Time
•
Install and launch Cisco Unified RI M f on your student PC.
•
Enter the IP address of jour Cisco Unilied Communications Manager (10_v. 1.1)
and specify the Administrator ID and password (cucmadmin /cucmpassl).
Monitoring Tool - Windows link.
Step 3
In Cisco Unified RTMT navigate to CallManager > Report > [.earned Pattern.
Step 4
From the Select a Node drop down menu, choose CUOMI-a-.
Step 5
Once the configuration of the other pod has finished, you should see a list of patterns
that were learned by CCD.
4M
Note
You should see a pattern of 851/2XXX with a toDIDof 4:+5551y555 and a pattern of
852/3XXX with a toDIDof4:+6652y555 Bothpatterns should be reachable by SIP at IP
address I0.y.1.1.
•
Verifv that calls can be placed to patterns learned by CCD.
Step 6
From jour IIQ and BR phones, placecalls to both sites of the other pod by dialing
8-5l.v2001 or8-5ly2002and8-52v3001.
Note
Cisco Unified Communications Manager will set up a call to the learned IP address using the
learned protocol (SIP inthiscase). The receiving Cisco Unified Communications Manager
cluster will strip the called numberto the internally used four-digit directory numbers
because the SAFSlPTrunk was configured with significant digits 4.
•
Verify that the backup path works.
Step 7
Make sure that all your phones have the AAR CSS set. The AAR CSS isused for
Call Control Discoverj (CCD)backup calls and it should be set to Global.CSS.
Step 8
Simulate an IP WAN failure between the HQ routers ofthe two pods (HQ-.-c and
HQ-y).
•
©2010 Cisco Systems, Inc
Break IP connectivity between your Cisco Unilied Communications Manager
(CUCMI-.x)and the Cisco Unified Communications Manager of the other pod
Lab Guide
Configure the SAF SIP Trunk in Cisco Unified Communications Manager
Step 10
Navigate to Device > Trunk and click Add New.
Step 11
Choose SIP Trunk, set the Trunk Service Type to Call Control Discovery, and
then click Next.
Step 12
Configure the trunk with the following settings, and then click Save.
•
Device Name: SAFSlPTrunk
•
Device Pool: Trunks
•
Significant Digits: 4
•
CallingSearchSpace: Trunk_css.
•
SIPTrunk Security Profile: Non Secure SIP Trunk Profile
•
SIP Profile: Standard SIP Profile
Configure the Call Routing Information to Be Advertised
Step 13
Navigate to Call Routing >Call Control Discovery >Hosted DN Croup and click
Add New.
Step 14
Step 15
In the Name field, enterPod-*_DN. Click Save.
Na% igate to Call Routing >Call Control Discovery >Hosted DN Pattern and
click Add New.
Step 16
In the Hosted Pattern field, enter 851jc2XXX.
Step 17
From the Hosted DN Group drop-down menu, choose Pod-Jt DN.
Step 18
For PSTN Failover Strip Digits, enter 4, and for PS'fN Failover Prepend Digits
enter +5551jc555. Click Save.
'
Step 19
Click Add New.
Step 20
In the Hosted Pattern field, enter 85Zx3XXX.
Step 21
Step 22
From the Hosted DN Group drop-down menu, choose Pod-:r_DN.
For PSTN Failover Strip Digits, enter 4, and for PSTN Failover Prepend Digits
enter +6651r555. Click Save.
Configure the Advertising Service
Step 23
Navigate to Call Routing >Call Control Discovery >Advertising Service and
click Add New.
Step 24
In the Name field, enter CCD Advertising Service I.
Step 25
From the SAF SIP Trunk drop-down menu, select SAFSlPTrunk, and from the
Hosted DN Group drop-down menu, choose Pod-.v DN.
Step 26 Check the Activated Feature cheek box and click Save.
Configure the Requesting Service
Step 27
Na\ igate to Call Routing >Call Control Discovery >Partition.
Step 28
Step 29
In the Name field, enter CCD_pt. Click Save.
Add partition CCD lo the CSS GIobal_css.
Step 30
Navigate to Call Routing >Call Control Discovery >Requesting Service.
Step 31
In me Name field, enter CCD Requester.
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
©20)0 Cisco Systems, Inc.
• Once the configuration ofthe other pod has finished, enter the show eigrp service-family
ipv4 2 neighbors command at your HQ-.v and BR-.r router:
At the HQ-x router you should see the BR-.r router and the HQ-y router as
neighbors.
Atthe BR-.\ router you should seethe IIQ-.v router as neighbor.
Task 2: Configure Cisco Unified Communications Manager as
SAF Client
In this task you will enable CCD by configuring Cisco Unified Communications Manager to
register uith jour pre\ iousK configured SAF forwarder.
Activity Procedure
Complete these steps:
Make the SUtically Configured Call Routes tothe Other Cluster Inaccessible
Step 1
Create a partition called Inaccessible.
Step 2
Change the partition of translation patterns 851y.2XXX and 852y.3XXX from
Global to Inaccessible.
Step 3
VcrifS that intersite calls using site-code dialing (851y2XXX and 852y3XXX) do
not work anymore.
Configure the SAF Forwarder in Cisco Unified Communications Manager
Step 4 Na^ igate to Advanced Features >SAF >SAF Security Profile and click Add
New.
Step 5
Configure the SAF security profile with the following settings and then click Save.
•
Name: HQ_SAF_Profile
•
Username: SAFl'SER
•
Password: SAFPASSWORD
Note
The username and password need to match the credentials configured in Task 1
Step 6
Navigate to Advanced Features >SAF >SAF Forwarder and click Add New.
Step 7
Configure the SAF forwarder with the following settings:
•
Name: HQ.v_SAF
•
Client Label: HQr_SAF
Note
The client label needs to match the external client configured at Task 1
•
SAF Security Profile: IIQjSAF_Profile
•
SAF Forwarder Address: 10jf.250.l01
Step 8
Click Show Advanced Link and move the CM_CUCM-1 Server to the Selected
StepS
Click Save.
Cisco Unified Communications Manager field.
Lab Guide
© 2010 Cisco Systems, Inc
87
Task 1: Configure SAF Forwarder Functionality on the HQ-x
and BR-x Router
In this task, you will configure the HQ-.r router to act as an SAF forwarder.
Activity Procedure
Complete these steps:
Step 1
Open a Telnet session to your HQ-.r router(IOjc. 1.10I).
Step 2
Enter theconfigure terminal command to access theglobal configuration mode.
Step 3
Enter the following commands:
i
router eigrp SAF
service-family ipv4 autonomous-system 2
i
topology base
external-client HQx_SAF
exit-sf-topology
exit service-family
service-family external-client listen ipv4 5050
external-client HQx_SAF
username
SAFUSER
password SAFPASSWORD
Note
Do not use special characters like spacesor dashesfor theexternal client definition.
Step 4
Open a Telnet session to your BR-x router (IO.v. 1. 102).
Step 5
Enter the configure terminal command toaccess the global configuration mode.
Step 6
Enter the following commands:
router eigrp SAF
service-family ipv4 autonomous-system 2
topology base
exit-sf-topology
neighbor 10.x.250.101 LoopbackO remote 16
exit-service-family
Activity Verification
You have completed this task when you attain these results:
• The SAF forwarders have been configured as described in the activity procedure.
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Implementing Cisco Unified Communications Manager. Part 2(CIPT2) v8.0
©2010 Cisco Systems. Inc.
Lab 5-1: Implementing Cisco SAF and CCD
Complete this lab activity lopractice what you learned inthe related module.
Activity Objective
In this activity. \ou will implement CCD using Cisco SAF clients and forwarders. After
completing this activity, you will be able to meet these objectives:
• Configure SAF forwarder functionality on the HQ-.t and BR-.r routers
• Contigurc Cisco Unified Communications Manager asan advertising and requesting SAF
client
•
Configure Cisco Unified Communications Manager Express onthe BR-.r router asa
requesting SAF client
Visual Objective
The figure illustrates what you will accomplish inthis activity.
Lab 5-1: Implementing Cisco SAF
CCD
Phnnel-*
Phane2-'
4
DHCP 1
ICx 30/24 ioy_;y
101
Enable CCD to
leam and advertise
call routing
information
"' Frame S^SS PSTN
s: Enable SAF. configure
SRST gateway to leam
call routing information
j&
J2£
Required Resources
These are the resources and equipment that arc required tocomplete this activitv:
Cisco Unified Communications Manager
Student PC
Cisco IP Phones
Cisco IOS MGCP gateway
H.323 galeway
PSTN with PSTN phone
© 2010 Cisco Systems. Inc.
Lab Guide
85
Step 15
Step 16
From the Selecta Servicedrop-down menu, choose the EM service. Click Next.
Click Subscribe, "fhe Cisco F.xiension Mobility service isdisplayed under
Subscribed Services.
Step 17
Click Save, then close the window.
Step 18
Step 19
Click Reset in the Phone Configuration window to reset the phone.
Repeat the previous steps (enabling Cisco Unified Communications Manager
Fxtension Mobility and subscribing tothe Cisco Extension Mobility IP phone
service) for Phone2-.r and Phone.Vr.
Note
As an alternative loperforming steps 3to19 you could have activated the Enterprise
Subscription check box when configuring theCisco Extension Mobility IP phone service.
Enterprise subscriptions apply to all phones and to all device profiles.
Activity Verification
You have completed this task whenyou attain these results:
• You can log in and log out at Phone I-x, Phone2-jr, and Phone3-.r by performing Ihe
following steps:
Step 1
Press the Services button.
Step2
Choose the Cisco Extension Mobility service.
Step 3
Log in with username and PIN.
Step 4
The phone will reset and should (hen be loaded with your device profile, fhe
director, numbershouldchangeto 2405.
Step 5
Note
Place calls to internal and external (PSTN) destination.
Cisco Extension Mobility does not modify device level settings such as region and location
ordevice CSS and AAR CSS. These parameters arenot configurable in the device profile.
The line CSS of the phone where a Cisco Extension Mobility user logs misupdated with the
line CSS of the device profile. In this lab. the line CSS isHQ_css. This CSS provides access
to theHQ translation patterns. Therefore, the PSTN dial rules ofthe HQ sitehave to be
used.
• The configured service parameters are working. This can be verified by performing the
following steps:
Step 1
Step 2
Log in at Phone3-.v and place acall. Then wait for the maximum login timer (3
minutes) to expire. You should be automatically logged out when the timer expires.
Log in again at Phone3,v. Verify that the call list was cleared alter logout: use the
Redial softkey and verify that the phone does not remember Ihe last destination.
Step 3
Do not log out. Log in at Phone 1-.t before the 3-minute timer expires. Once you
have logged in at Phone I-.t. you should be automatically logged out at Phone3-.v.
because the multiple login behavior has been set lo auto-logout.
Note
After logging out or being logged out of a phone, the phone reconfigures itself to its standard
settings
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Implementing Cisco Unified Communications Manager. Part 2(CIPT2) v8.0
©2010 Cisco Systems, Inc.
Task 4: Add the Cisco Extension Mobility IP Phone Service and
Subscribe to IP Phones and Device Profiles
Inthistask, vou will addthe Cisco Fxtension Mobility IP phone service andsubscribe it to IP
phones and the device profile.
Activity Procedure
Complete these sleps:
Add the Cisco Extension Mobility IP Phone Service
Step 1
Go to Device >Device Settings >Phone Services and click Add New.
Step 2
Define the following fields and click Save:
Note
•
Sen. ice Name: EM
•
ASCII Service: EM
•
Service Description: Extension Mobility Login/Logoff
•
Service URL:
hHp://10jc.l.l:8080/emapp/EMAppServlet?tlevice=#DKVICENAME#
The service URL is case-sensitive.
Service Category: XML Service
Service Type: Standard IP Phone Service
finable: Check
Note
After you click Save, the Parameters pane will appear—Cisco Extension Mobility does not
need any additional parameters to be specified.
Subscribe Device Profiles to the Cisco Extension Mobility IP PhoneService
Step 3
Nav igate to Device >Device Settings >Device Profile and click the Find button.
Step4
Click the device profile andy_dp.
Step 5
Step 6
Step 7
At the Related Links, choose Subscribc/linsubscribe Services: then click Go.
From the Select aService drop-down menu, choose the EM service. Click Next.
Click Subscribe. The Cisco Extension Mobility service is displayed under
Subscribed Services.
Step 8
Click Save, then close the window.
Enable Cisco Unified Communications Manager Extension Mobility atthe Phones and
Subscribe the Cisco Extension Mobility IP Phone Service to IP Phones
Step 9 Nav igate to Device >Phone and click Eind.
Step 10
Open Phonel-a.
Step 11 In the Extension Information pane check the Enable Extension Mobility check box.
Step 12 At the Log Out Profile drop-down menu, choose Use Current Device Settings.
Step 13
Click Save and click OK in the pop-up window.
Step 14
At the Related Links choose Subscribc/linsubscribe Services: then click Co.
Lab Guide
© ;;010 Cisco Systems, Inc
Task 3: Add and Associate an End User with the User Device
Profile
In this task, you will add an end user to Cisco Unified Communications Manager and associate
this user with the newly created device profile.
Activity Procedure
Complete these steps:
Add New User Through Cisco Unified Communications Manager Administration
In Cisco Unified Communications Manager, configure an end user:
Step 1 from PC-.r. access Cisco Unified Communications Manager Administration.
Step 2
Go to Eser Management > End User and click Add New.
Step 3
Configure a user with the attributes that follow, and save the newly created account
by clicking Save at the bottom of" the page or the Save symbol at the top ofthe End
User Configuration window.
•
UserID:andy
•
Password: password
•
PIN:12345
•
Last Name: Szoldatics
•
First Name: Andreas
Assign the Device Profile to the End User
Step 4
In the t.xtension Mobility pane of the End User Configuration window, from the
Available Profiles list, choose the profile andy_dp and add it to the Controlled
Steps
Click Save.
Profiles using the down arrow.
Step 6
In User Management > End User, verify that the end user "andy" has been added to
Cisco Unified Communications Manager.
Activity Verification
You have completed thistaskwhen you attain these results:
• The end user "andv" is configured in User Management > End User as described in the
activity procedure.
• The device profile andy_dp is assigned to the end user as described in the activity
procedure.
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems, Inc
Task 2: Create a Device Profile for a User
In this task, vou will create a device profile for the user of Phone l-.v.
Activity Procedure
Complete these steps:
Step 1
Nav igate to Device >Device Settings >Device Profile.
Step 2
Click Add New.
Step 3
Tor ihe phone model, choose Cisco 7965. Click Next.
Step 4
Keep the SCCP signaling option and click Next.
Note
If your Phonel -x is a different model, usethe model of your phone
Step 5
Filterthe following parameters:
•
Device Profile Name: andy_dp
•
Description: Device Profile-Andy
•
User Hold MOH Audio Source: I-Samplc Audio Source
•
User Locale: Knglish, I'nited States
•
Phone Button Template: Standard 7965 SCCP
Note
The phone button template depends on the phone model that you chose earlier.
•
Soflkey Template: Standard User
Step 6
Click Save.
Step 7
In the Association Info pane, click the Line 111 Add anew DN link.
Step 8
Step 9
Step 10
lor the director) number, enter 2405.
Choose the route partition Internal.
Choose the CSS HQ_Phones_CSS.
Step 11
For the Exiemal Phone Number Mask value, enter +5551.v5552X\X.
Step 12
Click Save.
Activity Verification
You nav e completed this task when you attain these results:
• The new profile is configured in Dev ice >Device Settings >Device Profile as described in
the activity procedure.
• Director) number 2405 is associated with the new device profile andy_dp as described in
the activity procedure.
Lab Guide
© 2010 Cisco Systems, Inc
•
H.323 gateway
•
PSTN with PS'I N phone
Task 1: Activate the Cisco Extension Mobility Service and
Configure the Corresponding Service Parameters
In this task, you will activate the Cisco Extension Mobility service and configure Ihe login
behavior by setting ihe appropriate service parameters for the Cisco Extension Mobility service.
Activity Procedure
Complete these steps:
Step 1
Open Cisco Unified Serviceability.
Step 2
Go to Tools > Service Activation.
Step 3
Check the check box for the Cisco Extension Mobility service and click Save to
activate it.
Note
The Cisco Extension Mobility service can be activated on multiple servers
In Cisco Unified Communications Manager Administration, navigate to Systc
Step 4
Service Parameters.
Step 5
From the Serverdrop-down menu, choose ltLv.1.1.
Step 6
From the Service drop-down menu, choose Cisco Extension Mobility.
Step 7
Configure the following parameters:
•
•
Enforce Intra-Cluster Maximum Login Time: True
Intra-Cluster Maximum Login Time: 0:03
It is acommon configuration error to subscribe to the Cisco Extension Mobility service only
Note
at the IP phone and not also at the device profile. In such a situation, you cannot log out of
the phone anymore once you have logged in. By setting the maximum login time to a
relatively low value, you have a back door for this case, because an auto logout is
performed after expiration of the maximum login time. This is a common setup for a lab
environment.
Steps
•
Intra-Cluster Multiple Login Behavior: Auto Logout
•
Remember the Last User Logged In:True
•
Clear Call Log on Intra-Cluster EM: True
Click Save.
Activity Verification
You have completed this task when vou attain these results:
• The Cisco Extension Mobility service parameters in System >Service Parameters tire
configured as described in the lask.
Note
80
Further venfication will bedone in a later task of this lab exercise.
Implementing Cisco Unrfied Communications Manager, Part 2(CIPT2) ve.O
©2010 Cisco Systems, Inc
Lab 4-2: Implementing Cisco Extension Mobility
Complete this lab activity to practice what you learned in the related module.
Activity Objective
In this activity. you will implement Cisco Extension Mobility for roaming users. After
completing this activity, you will be able to meet these objectives:
• Configure the sen ice parameters for Cisco Fxtension Mobility
•
Create a device profile for a user
•
Add an end user
•
Associate the end user with the user deviceprofile
• Subscribe IP phones and device profiles to the IP phone service for Cisco Extension
Mobility
Visual Objective
The figure illustrates what vou will accomplish in this activity.
Lab 4-2: !rn
Mobility
=hpne'-<
PI>one2
J
Allow roaming
users to log in to
HCP ^T^DHCP
any phone and
have personal
settings applied
Frame ^Sj PSTN
Relay
Allow roaming users to log
in to any phone and have
personal settings applied
Required Resources
These are the resources and equipment that arc required to complete this activity:
•
One Cisco Unified Communications Manager cluster
•
Student PC
•
Cisco IP Phones
•
Cisco IOS MGCP gateway
Lab Guide
© 2010 Cisco Systems. Inc
79
The local route group ofaroaming phone should be updated. Perform the following steps
for verification:
Stepl
Nav igate to Device > Phone and click Find.
Step 2
Choose Phone2-.v.
Step 3
At the phone configuration window, click the View Current Device Mobility
Settings link next to the Device Mobility Mode parameter. Awindow will pop up,
show ing ihecurrent configuration of the phone.
Note
Step 4
The local route group is nol shown in the pop-up window. You cannot verify that the local
route group has been updated by using the View Current Device Mobility Settings link.
Log in to the HO-.v router and enable ISDN debugging with the debug isdn q931
command. Make sure to turn onmonitoring with the terminal monitor command.
Step 5
Shut down the ISDN interface at the branch router so that the gatcwav cannot be
used for TEHO calls to the BR PSTN.
Step 6
From the Phonc3-.v (which is currently roaming to the HQ site) place acall to the
BR PSTN by using the home dial rules (for example, 9 5554444). 'fhe call uses the
IIQ-.v gateway. This indicates that the local route group ofthe phone was updated by
the one configured in the roaming device pool.
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Implementing Cisco Unified Communicatiot ;Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems, Inc.
Step 17
Click Save.
Step 18
Click Add New and enter the following parameters in the Device Mobility Into
Configuration window:
Step 19
•
Name: BR_dmi
•
Subnet: KU.4.0
•
Subnet Mask: 24
Mov ethe Branch device pool from the Available Device Pools list to the Selected
Dev ice Pools list.
Step 20
Click Save.
Enable Device Mobility Mode for the Cluster
Step 21 Navigate to System>Service Parameters.
Step 22
from the Server drop-down list, choose Cisco Unilied Communications Manager
10-v.l.l.
Step 23
From die Service drop-down list, choose Cisco CallManager
Step 24
In Clusierwide Parameters (Device- Phone), set the Device Mobility Mode to On.
Activity Verification
You have completed this task when vou attain these results:
• Phones can register at different physical locations. Perform the following steps tor
verification:
Step 1
Disconnect Phonc2-.r and Phone3-.v from the switch port.
Step 2
Step 3
Reconnect Phone2-.r to the port of Phone3--v.
Reconnect Phone3-.v lothe port of Phone2-„v.
Step 4
Phone2-.v should register in the branch with its directory number 2002, and Phone3-A
should register in the headquarters with its directory number 300I.
Step 5
Place calls between any ofthe three phones.
• Roaming phones have their roaming-sensitive settings updated based on the configuration
in the roaming dev ice pool. Perform the following steps for verification:
Step 1
Nav igate to Device >Phone and click Find.
Step 2
Choose Phone2-.v.
Step 3
At the phone configuration window, click the View Current Device Mobility
Settings link next to the Device Mobility Mode parameter. Awindow is dtsplaved.
which showsthe currentconfiguration of the phone.
Step 4
Verifv that the headquarters phone adapted lo its new physical location (branch) by
changing the roaming-sensitive settings (such as the location, region, and SRS 1
reference).
Step 5
In order to sec that these updated settings are active, place acall from I'hone2-* lo
Phonel -.v. Press the ?button twice at both phones. The codec that is used tor the cal
should be G.729.
Note
When it is in the home location, Phone2-x uses G.722 for calls to Phone1-x.
Lab Guide
i 2010 Cisco Systems. Inc
Task 1: Configure Device Mobility
In this task, vou will configure the Device Mobility feature so that roaming users can use their
home dial rules, but can aiso use the local route group as abackup path for TEHO PSTN calls.
Activity Procedure
Complete these steps:
Configure Physical Locations
Step 1
Nav igate to System > Physical Location and click the Add New button.
Step 2
In the Physical location Configuration window, enter the following parameters:
Step 3
•
Name: HQpl
•
Description: Headquarters
Click Save.
Click Add New and configure another physical location for the branch office, with
Step 4
the following parameters:
Step 5
•
Name: BR pi
•
Description: Branch
Click Save.
Note
No DMGs are required. When no DMG is set at the roaming device pool and at the home
device pool, the device-mobility-related settings are not updated. In this lab, globalized call
routing is used. Therefore, there is no need to change the device CSS, AAR group, or AAR
CSS,as theyare the same at allphones.
Configure Device Pools
Step 6
Navigate to System > Device Pool andclickthe Find button.
Step 7
Choosethe device pool Default.
Step 8
From the Physical Location drop-down menu, choose IIQ_pl.
Step 9
Click Save and then reset the dev ice pool.
Step 10
In the Related finks pane, select Back To Find/List and then click Ihe Go button.
Step 11
Choosethe device pool Branch.
Step 12
From the Physical Location drop-down menu, choose BR__pl.
Step 13
Click Save and then reset the device pool.
Create Device Mobility Infos
Step 14
Step 15
Nav igate to System >Device Mobility >Device Mobility Info and click Add New.
Enter the following parameters in the Device Mobility Info Configuration window:
•
Step 16
Name: HQ_dmi
•
Subnet: 10_v.2.0
•
Subnet Mask: 24
Move the Default device pool from the Available Device Pools list lo ihe Selected
Device Pools list.
76
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v*0©2010 Cisco Systems. Inc.
Lab 4-1: Implementing Device Mobility
Complete this lab activity to practice what you learned in the related module.
Activity Objective
In this activitv. vou will enable the Device Mobility feature to help mobile users who roam
awav from their home location. After completing this activity, you will be able to meet these
objectives:
•
Configure Device Mobility
Visual Objective
The figure illustrates what vou will accomplish in this activity.
Lab 4-1; Implementing Device
Required Resources
These are the resources and equipment that arc required to complete this activitv
•
Cisco Unified Communications Manager
•
Student PC
•
Cisco IP Phones
•
Cisco IOS MGCP gateway
•
H.323 gateway
•
PSTN with PSTN phone
Lab Guide
© 2010 Cisco Systems, Inc
75
Note
The SIP trunk does not need to have access to the RSVP Agent media resource. Therefore,
MRGs and MRGLs do not have to be modified.
Step 4
Step 5
Nav igate to Device >Device Settings >SIP Profile and click the Add button.
In the Name field, enter SIPPrecondition, and scroll down lo the Trunk Specific
mm
Configuration section.
^
Step 6
From the Reroute Incoming Request toNew Trunk Based On drop-down menu,
select Never.
'
|L
Step 7
From the RSVP Over SIP drop-down menu, select E2E and at theSIPRel IXX
Options, choose Send PRACK if Ixx Contains SDP.
'mt
Step 8
Go to the configuration page ofthe existing trunk called SIPJI'runk.
Step 9
From the SIP Profile drop down menu, select SIP_Precondition and click Save and
Apply.
Configure the RSVP Bandwidth at the IP WAN Interface That Connects to the Other Pod
Step 10
At the subinterface that interconnects the headquarters and the other pod, configure
**
"**
jjL
the bandwidth thatcan bereserved by RSVP as follows:
interface Serial...
s«-
ip rsvp bandwidth 4 0
Test RSVP SIP Preconditions CAC
Note
Step 11
'L*
Check that the configuration of the other pod is also completed before continuing.
Establish one call between your local headquarters phone and the other headquarter
phone at the other pod and keep the call open. Try to set up asecond call by calling
the second headquarter phone for the other pod. The second call should be rerouted
over the PSTN Use the debug isdn q931 command toverify thai the call is sent
through the PSTN. The show seep connections rsvp command can be used toshow
the currently active connections atthe RSVP agent.
Activity Verification
in
You have completed this task when you attain these results:
• You configured SIP Preconditions between the two pods as described in the activity
procedure.
You can place one call between the two pods over the SIP trunk and end-to-end RSVP is
^
When placing an additional call, the PSTN is used as abackup as described in the activity
jk
used for that call as described inthe activity procedure.
procedure.
Note
Make sure lo turn off all of the debug commands at all of the routers (use no debug alt
command).
2.
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) V8.0
©2010 Cisco Systems, Inc.
Step 19
Repeat the previous steps for Phonc2-* and Phone3-x
Modifying the RSVP Bandwidth So That All Calls Fail
Step 20
Reconfigure the RSVP bandwidth on one of the two routers to avalue below 40
kb/s. This will make all calls between branch and headquarters fail.
Step 21
Try calls between the headquarters and branch. The calls should be rerouted through
ihe PSTN.
Configure One Phone for an Alternate CFNB Destination
Step 22
Change the configuration of the Phonel-.v line so that calls arc forwarded to aPSTN
number if2001 cannot he reached because ofCAC. Set the AAR destination mask
to +556065554444.
Step 23
Try to call 2001 from the branch phone. The call should be sent to the PSTN phone
using the BR gateway.
Allowing Calls over IP WAN Again
Step 24
Step 25
Change the RSVP bandwidth back to40 on both routers.
Verify that one call between the headquarters and branch uses the IP WAN. Keep
that call open. Tr> calling the branch phone from another phone in the headquarters.
You should see an incoming call at the branch phone (which is still in a call) coming
through the PSTN. You can use the debug isdn q931 command to verify that calls
arc using the PSTN path. The show seep connections rsvp command can be used to
s,hou the currently active connections at the RSVP agent.
Cleanup
Step 26
Remov ethe (TNB setting at Phone 1-x by clearing the AAR destination mask at
line 1.
Activity Verification
You have completed this task when vou attain these results:
• When intracluster calls are rejected because there is no available bandwidth, the calls are
rerouted over the PSTN as described in the activity procedure.
Note
Make sure to turn off all of the debug commands atall of the routers (use the no debug all
command)
^
Task 4 (Optional): Configure SIP Preconditions
In this task, you will modify the previously implemented RSVP-bascd CAC to implement endto-end CAC for calls between pod 1 and pod 2.
Activity Procedure
Complete these steps:
Enable End-to-End RSVP to Be Used for Calls Between the Two Pods Using the SIP Trunk
SIP Preconditions-based CAC should allow one Ci.729 call between the two pods.
Step 1
Step 2
Step 3
Cio to the configuration page of the existing location called Trunk.
Prom fhe Modifv Settings) to Other Locations pane, select the Iliib_None location,
and from the RSVP Setting drop down menu, choose Mandatory (V ideo Desired).
Click Save.
Lab Guide
© 2010 Cisco Systems. Inc
73
Task 3: Configure AAR and CFNB to Route Calls over the PSTN
If They Are Not Admitted by the Deployed CAC Methods
In this task, you will configure a backup path for calls that are rejected by the previously
implemented CAC methods. These calls will be rerouted over the PSTN using AAR and
CFNB.
Activity Procedure
Complete these steps:
Enable AAR
In the following steps, you will enable AAR by setting the Cisco CallManager service
parameter Automated Alternate RoutingEnabled to True.
Step 1
Navigate to System > Service Parameters and choosethe Cisco Unified
Communications Manager(IOjc.1.1).
Step 2
From the Service drop-down menu, choose Cisco CallManager.
Step 3
Locate the Clusterwide Parameters(System—CCM Automated Alternate
Routing) pane.
Step 4
Set the Automated Alternate Routing Enable parameters toTrue.
Step 5
Click Save.
Configure an AAR Group
Step 6
Navigate to Call Routing> AAR groups and click Add New.
Step 7
Enterthe name System_aar.
Step 8
Click Save.
Configuring Phones for AAR
Step 9
Navigate to Device > Phone and click Find.
Step 10
Choose Phonel-x.
Step 11
Click Line 1toget tothe Director) Number Configuration window.
Step 12
In the AAR Settings pane, choose System_AAR for the AAR Group.
Step 13
Verify that the external phone number mask isin globalized format.
Step 14
Click Save.
Step 15
From the Related Links, choose Configure Device and click <;».
Step 16
At the Phone Configuration window, choose Global ess for the AAR CSS.
Note
When acall between the HQ and BR sites is not admitted, AAR will be used to place the call
over the PSTN The AAR call will match the Wtranslation pattern first, and then the TEHO
pattern of the local pod. The first option of the route list that is applied to the TEHO pattern is
the TEHO gateway. This, however, cannot be used, because there Is not enough bandwidth
available between the HQ and BR sites. Therefore, the second option of the route list is
used—the local route group.
72
Step 17
Click Save and then OK in the pop-up window.
Step 18
Reset the phone.
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems, Inc
Step 19
Click Save.
Put the RSVP Agents Into MRGs
Step 20
Add the HQ-RSVP RSVP agent to the HQ_mrg MRG.
Step 21
Add the BR-RSYP RSVP agent tothe BR-SW-CFB_mrg MRG.
Test RSVP CAC
Note
The RSVP configuration has to be performed onboth sides. Unless both sides are
configured for RSVP, the call will fail.
Step 22
Step 23
Place a call between aheadquarters phone and a branch phone. The call should fail.
Use the debug ip rsvp signaling command on the I\Q-x router to see why the
reservation fails. Ilow much bandwidth do you expect to be reserved? How much is
aetuallv reserved0
Note
During the call setup phase, the RSVP agents always attempt to reserve an additional 16
kb/s (for signaling). Therefore, in this case, the RSVP bandwidth atthe interface must allow
40 kb/s for the call togothrough. The extra 16 kb/s that that the RSVP agents attempt to
reserve during call setup areimmediately released once the call issetup end-to-end.
Step 24
Change the RSVP bandwidth at the IP WAN interfaces on the IIQ-jc and on the BR.i routers to 40 kb/s. Now the calls should go through. If you want, you can retry
with 39kb/s tomake sure that 40 kb/s isthe absolute minimum for one G.729 call to
be allowed. Make surethat youset it back to 40 kb/safterward.
interface serial...
ip rsvp bandwidth 4 0
Note
The RSVP bandwidth command has tobemodified on both sides. Unless both sides permit
enough bandwidth for RSVP, thecall will fail.
Step 25
. ^
I stablish one call between a headquarters phone and the branch phone and keep the
call open. Try to set up asecond call by calling the branch phone directory number
from the other phone inihe headquarters. The call should fail.
Activity Verification
You have completed this task when you attain these results:
• You modified Cisco Unified Communications Manager CAC between the IIub_None and
the branch locations to use RSVP asdescribed intheactivity procedure.
• RSVP permits one G.729 call between these two locations. Additional calls fail, because of
a lack ofavailable bandwidth asdescribed inthe activity procedure.
Note
Make sure to turn off all of the debug commands at all of the routers (use the no debug all
command)
^
Lab Guide
) 2010 Cisco Systems. Inc
dspfarm profile 2 mtp
codec pass-through
rsvp
maximum sessions software 2
associate application SCCP
no shutdown
Configure the RSVP Bandwidth atthe IP WAN Interfaces ofthe Routers
Steps
On the main Frame-Relay serial interface, enable fair queuing, as follows:
interface Serial...
bandwidth 2000
fair-queue
Note
Use the main interface that is connected to the Frame Relay network (PSTN router).
Step 9
At the subinterface that interconnects the headquarters and the branch router,
configure the bandwidth that is allowed to be reserved by RSVP as follows:'
interface Serial...
ip rsvp bandwidth 24
Step 10
Step 11
Save the configuration to NVRAM.
Repeat the above steps (configure Cisco IOS routers to provide RSVP agent MTP
resources and configure the RSVP bandwidth onthe IPWAN interfaces of the
routers) at your BR-x router. When configuring the media resource, use the name
BR-RSVP instead ofHQ-RSVP in the associate profile command.
Add the RSVP Agents (MTPs) in Cisco Unified Communications Manager
Step 12
In Cisco Unified Communications Manager Administration, navigate to Media
Resources >Media Termination Point and click Add New.
Step 13
Verify that the Media Termination Point Type value isCisco IOS Enhanced
Software Media Termination Point.
Step 14
Note
Enter HQ-RSVP for the Media Termination Point Name value.
Tne name ofthe media resource iscase-sensitive.
Step 15
Hnter HQ-* RSVP Agent for the description.
Step 16
Choosethe device pool Default.
Note
The location Hob^None and region HQ are applied to the HQ-RSVP media resource
through the device pool Default.
Step 17
Step 18
Note
Click Save.
Click Copy and change the name to BR-RSVP, the description to HR-a RSVP
Agent, and device pool to BR.
The location Branch and region BR are applied to the HQ-RSVP media resource through the
device pool Branch.
70
implementing Cisco Unmed Communications Manager. Part 2(CIPT2, ,8.0
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©2010 Cisco Systems, Inc.
Step 12
Repeat the previous steps lo apply the llub_Nonc location to the MOH.Only
device pool.
Step 13
Repeat the previous steps to apply the Branch location to the BR device pool.
Step 14
Nav igate lo Device >Trunk and click the Find button.
Step 15
Choose the SIP_Trunk.
Step 16
From the location drop-down menu, choose Trunk.
Step 17
Click Save and reset the trunk.
Activity Verification
You have completed this task when you attain these results:
• You have configured and applied locations as specified in the activity procedure.
• You cannot place more than one call to the other cluster using the SIP trunk (by dialing
851 v-2001. 85 ly-2002. or 852v-300l). The G.729 codec should be in use lor ibis call.
• You cannot place more than one call to the branch phone (by dialing 3001). The (G.729
codec should be in use for this call.
Task 2: Configure RSVP-Enabled Locations
In this task vou will change the previously implemented locations-based CAC to use RSVP in
the IP WAN for calls between the Hub^Nonc and Branch locations. 1his will be done bv
deploying RSVP agents at the headquarters and branch sites.
Activity Procedure
Complete these steps:
Enable RSVP to Be Used Between the Hub_None and the Branch Locations
In the following steps, vou will change the branch location so that it uses RSVP toward the
1lubNone location. RSVP should be mandatory between these two locations.
Step 1
Nav igaie to System >Locations and click the Find button.
Step 2
Click Branch.
Step 3
In the Modifv Sctting(s) to Other Locations pane, from the RSVP Setting drop-down
Step 4
Click Save. You should see the changes in the Location RSVP Settings pane.
Click the Resync Bandwidth button to reset all CAC bandwidth usage, and click
Step 5
menu, choose the Hub_Nonc location and choose M»ndatory(Vidco Desired).
OK in the pop-up window.
Note
RSVP is configured per pair of locations. The setting applies to both directions. Therefore,
the configuration that you apply to one location automatically updates the other location
accordingly
.
Configure Cisco IOS Routers to Provide RSVP Agent MTP Resources
StepG
Connect to your HQ-jc router.
Step 7
Configure router IIQ-.v asfollows:
seep ccm group 1
associate profile 2 register HQ-RSVP
Lab Guide
© 2010 Cisco Systems. Inc
69
H.323 gateway
PSTN with PSTN phone
Job Aids
This job aid is available to help you complete the lab activity.
i»
Location Configuration
Name
Allowed Bandwidth
Applied To Device
Applied To Device
Pool
Hub_None
Unlimited
Default
MOH_0nly
Branch
24 kb/s
Trunk
24 kb/s
Branch
SIPJTrunk
Task 1: Configure Locations
In this task you will configure locations-based CAC for calls between the headquarters, branch
and SIP trunk.
Activity Procedure
Complete these steps:
Add Locations to Cisco Unified Communications Manager
Create new locations as described in the "Location Configuration" table in the .lob Aids section.
Step 1
Navigate to System > Location and click the Find button.
Step 2
Click the location name Hub_None to enter the Location Configuration window.
Step 3
Make sure that the Hub^None location has unlimited audio bandwidth.
Step 4
Click the Add New button.
Step 5
Configure anew location with the following parameters:
•
Name: Branch
•
Audio bandwidth should be limited to 24 kb/s
Step 6
Click Save.
Step 7
Repeat the previous steps to configure the remaining location with the location name
Trunk as described in the "Location Configuration" table in Ihe Job Aids section.
Apply Locations to Devices
Apply the newly created locations to devices, through the device pool or directly, as described
inthe Location Configuration" table in the Job Aids section.
Steps
Navigate to System > Device Pool and click the Find button.
Steps
Choose Default to enter the Device Pool Configuration window for device pool
Default.
68
Step 10
From the Location drop-down menu, choose HubJSonc.
Step 11
Click Save and reset the device pool.
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
•
>2010 Cisco Systems, Inc.
Lab 3-2: Implementing CAC
Complete this lab activity to practice what you learned in the related module.
Activity Objective
In this ael.v itv. vou will implement CAC by configuring locations and deploying RSVP agents
to prevent WAN bandwidth oversubscription. You will implement AAR to route calls over the
PSTN ifthev were not admitted bv locations-based CAC. Then you will mplement SIP
Precondition's in order to implement end-to-end CAC for the SIP trunk to the other pod. After
completing this activity, you will be able lo meet these objectives:
•
Configure standard locations
•
Configure RSVP-enabled locations
i Configure AAR to route calls over the PSTN ifthey arc not admitted by the deployed CAC
methods
•
Configure SIP Preconditions
Visual Objective
fhe figure illustrates what vou will accomplish in this activity.
Configure
locations. RSVP
agents. AAR, and
' _^- SIP Preconditions
HQ-.
;ۤ
Required Resources
These arc the resources and equipment that are required to complete this activity:
•
Cisco Unified Communications Manager
•
Student PC
•
Cisco IP Phones
•
Cisco IOS MGCP gateway
Lab Guide
2010 Cisco Systems. Inc
Enable Multicast MOH from Branch Router Flash at the BR-x SRST Router
Step 51
Configure the branch router as follows:
Hi
telephony-service
multicast moh 239.1.1.1 port 16384 route 10.x.4.102
Note
El
The moh moh-file-name command that is used to enable unicast MOH in SRST mode already configured in an earlier lab exercise.
Step 52
Saveyour configuration to NVRAM.
Verify That Multicast MOH from Branch Router Flash Works
Step 53
Place acall from Phonel-* to Phone3-*, and at Phonel-*, put the call on hold
Phone3-* should play MOH. Keep the call in this state.
Step 54
Pressthe Settings button at Phone3-x
Step 55
Step 56
Choose option 2 Network Configuration.
Press 6orscroll down toget tooption 6 IPAddress.
Step 57
Write down the IP address ofPhone3-*: 10.
Step 58
Using a web browser, browse tothe IP address of Phone3-.v.
Step 59
Click the Stream 1link to see details about the current RTP stream.
Step 60
.4.
The local address should be 239.1.1.1/16384, which indicates thai the phone listens
to the multicast MOH stream.
Note
The phone now plays the locally generated multicast MOH stream.
'
'
.
___
-•*&•
Activity Verification
You have completed this task when you attain these results:
Si
• Branch phones can play MOH created by the local SRST router as described in the activity
procedure.
Note
J
e
^
More detailed verification was pan of the Activity Procedure.
Hm%
m*
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©2010 Cisco Systems, Inc.
Step 42
Step 43
Press the Settings button atPhone3-.v.
Choose option 2 Network Configuration.
Step 44
Step 45
Step 46
Press 6or scroll down to get to option 6. IP Address.
Write down the IP address ofPhone3-.v: 10. .4.
Using aweb browser, browse to the IP address of Phone3-.r.
Note
If you did not enable web access to the phone earlier, you need to enable it now, in order to
be ableto browse to the phone.
Step 47
Click the Stream 1link to see details about the current RTP stream.
Step 48
The local address should be 239.1.1.1/16384, which indicates that the phone listens
to the multicast MOH stream. Keep the call in this stale so that you hear MOH.
PreventMulticast MOH from Being Sent overthe IP WAN
Step 49
At router HQ-.t. disable multicast routing toward the branch by entering the
following commands:
interface Serial...
no ip pim sparse-dense-mode
Note
Use the interface that connects to the BR-x router (IP WAN).
Note
As soon as you enter the above commands, Phone3-x should not play MOH anymore Nor
will it play TOH, because Cisco Unified Communications Manager is unaware that the phone
nolonger receives theMOH audio stream
_____
Step 50 At router BR-.r. disable multicast routing by entering the following commands:
interface Serial...
no ip pim sparse-dense-mode
Note
Use the interface that connects to the HQ-x router (IP WAN)
interface FastEthernet...
no ip pim sparse-dense-mode
no ip multicast-routing
Note
Use the interface that connects to the branch phones.
Note
Mullicast MOH in SRST does not require multicast routing. It simply streams permanently at
the interface that is configured to be used by SRST. If the stream is requ.red on adifferent
mterface (for example, when using aloopback for SRST) the interface or interfaces can be
specified using the route option of the multicast moh command (as shown in the next step)
Lab Guide
© 2010 Cisco Systems, Inc.
65
At this point, the MOH server shares the same region with all other headquarters devices To
m
limit calls to these other devices to G.729 but allow G.7I Ibetween the MOH server and the
branch phones, the MOH server needs to be placed into aseparate, dedicated region In
addition, the multicast MOH stream that is generated by the MOH server has to be blocked
from the IP WAN. Then multicast MOH can be enabled at the BR-* SRST router.
Allow G.711 Between the MOH Server and Branch Phones
Step 29
In Cisco Unified Communications Manager Administration, navigate to System >
Region and click Add New.
Step 30
k
Enter MOH for the name and click Save.
Step 31
Using the Modify Relationship to other Regions pane, allow the G.722 and G711
audio codecs to be used to the region HQ by highlighting the region HQ in the
Regions list and choosing G.722/G.711 from the Audio Codec drop-down menu
Step 32
Using the same technique, also allow the G.722 and G.711 audio codecs for calls
m\
You must click Save after each change in the Modify Relationship to Other Regions pane
By limiting the audio codec to G.729 for calls between regions Trunks and MOH you
effectively disable MOH for these calls. The reason is that the MOH server is only streaming
G.711 multicast MOH, and a multicast stream cannot be transcoded (which would be
required toward the region Trunks). This is desired, because G.729 MOH has only poor
quality, and G.711 must not be sent over the IP WAN (used by the trunks).
Step33
Navigate to System > Device Pool andclickAdd New.
Step 34
Configure the following parameters for the new device pool:
•
Device Pool Name: MOH_Only
• Cisco Unified Communications Manager Group: Default
Step 35
•
DateATime Group: CMLocal
•
Region: MOH
•
SRST Reference: Disable
Click Save.
Step 36
Navigate to Media Resources >Music On Hold Server and click lind.
Step 37
Choose HQ-SVV-MOH.
Step 38
Change the device pool from Default to MOH_Only.
Step 39
Click Save.
Step 40
Reset the MOII server.
Verify That Multicast MOH Now Works to Branch Phones
Step 41
64
£.
mt
The changes will then appear in the Region Relationships pane.
Note
&
Click Save.
between the region MOH and the region BR and for calls within the region MOH.
Allow G.729 only for calls between the regions MOH and Trunks.
Note
it
Place acall from Phone I,r to Phone3-x, and at Phone I-x put (lie call on hold
Phone3-.v should play MOH. Keep the call inthis state.
Implementing Cisco Unified Communications Manager. Par. 2(CIPT2) v8.0
©2010 Cisco Systems, Inc.
m
Verify That Multicast MOH Is Used
Step 20
Place acall from Phonel-* to Phone2-x. and at Phoncl-x. put the call on hold.
Phone2-.v should play MOH. Keep the call inthis slate.
Step 21
Step 22
Press the Settings button at Phonc2-.v.
Choose option 2 Network Configuration.
Step 23
Step24
Step 25
Press 6or scroll down to get to option 6, IP Address.
Writedown the IP address of Phonc2-.x: 10. .2. _
Using aweb browser, browse to the IP address ofPhone2-v.
Note
If you did not enable web access to the phone earlier, you need to enable it now, in order to
be able to browse to the phone.
Step 26
Click the Stream I link to see details about the current RIP stream.
Step 27
The Local Address should be 239.1.1.1 /16384. which indicates that the phone listens
lo the multicast MOH stream.
Step 28
Place acall from Phone I-x to Phone3-.v. and at Phone I-x put the call on hold.
Phonc3-.v will play lone on hold only.
Note
The reason that Phone3-x will play tone on hold instead of MOH is that the MOH server is
configured for G.711 MOH only (this is the default configuration). However, before changing
to multicast MOH, Phone3-x played MOH. This was possible by using the transcoder media
resource The MOH server is configured with the device pool Default, which applies region
HQ and MRGL HQ_mrgl. This MRGL allows the MOH server to access the transcoder
Such a configuration is not recommended, because if MOH with the G.729 codec should be
permitted, it can be directly enabled on the MOH server (by using the Supported MOH
Codecs service parameter of the Cisco IP Voice Media Streaming Application service)
Using G.729 for MOH, however, is not recommended, because the G.729 codec audio
quality for music is poor; G.729 is designed and optimized for human speech, and does not
work well with music.
Multicast audio streams cannot be transcoded, sobranch phones do not hear MOH
anymore, because the MOH server was configured to use multicast MOH instead of unicast
MOH You will solve this problem by implementing multicast MOH from branch router flash.
Enable Multicast MOH from Branch Router Flash
When using multicast MOH from branch router flash, the branch router locally generates a
multicast MOH stream. This stream must use attributes (destination address -that is. multicast
group address-port numbers, and codec) that are identical to the attributes tor the multicast
MOH stream that is generated by the Cisco Unified Communications Manager MOH server
that is located at the headquarters. This is required because neither Cisco Unified
Communications Manager nor the branch IP phones are aware that the phones listen to astream
generated bv the local SRST gateway. Cisco Unified Communications Manager tel sthe phone
lo listen to its stream (providing the attributes that were mentioned belore) in signaling
messages, and. therefore, the locally generated stream has to look exactly that way.
Because SRST MOH supports only G.711, Cisco Unified Communications Manager also has to
instruct the phone to listen to aG.711 MOH stream. Consequently. G.711 must be enabled
between branch phones and the MOH server in region configuration.
Lab Guide
© 2010 Cisco Systems. Inc
63
Update MRGs to Use Multicast
Step 14
Navigate to Media Resources >Media Resource Group and click the Find button.
Step 15
Choose the Generalmrg MRG.
Step 16
Ifat least one multicast HQ-SW-MOI Iresource is available, cheek the Use
Multicast for HQ-SW-MOH Audio check box in the Media Resource Group
Configuration window.
Step 17
Click Save.
Enable Multicast Routing in the Network
Step 18
At router HQ-*, enable multicast routing using the following commands:
ip multicast-routing
interface FastEthernet...
ip pim sparse-dense-mode
Note
Use the interface that connects to the voice server network (CUCM-x).
interface FastEthernet...
ip pim sparse-dense-mode
Note
Use the interface that connects to the headquarters phones.
interface Serial...
ip pim sparse-dense-mode
Note
Step 19
Use the interface that connects to the BR-x router (IP WAN).
At router BR-*, enable multicast routing using the following commands;
ip multicast-routing
interface Serial...
ip pim sparse-dense-mode
Note
Use the interface that connects to the HQ-x router (IP WAN).
interface FastEthernet...
ip pim sparse-dense-mode
62
Note
Use the interface that connects toIhe branch phones.
Note
Multicast routing is now enabled for the voice server network, the headquarters phone
network, the branch phone network, and the link between HQ-x and BR-x (IP WAN).
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
12010 Cisco Systems, Inc.
Stepl
Step 2
Initiate an ad hoc conference from Phone3-jr (adding Phonel-* and Phone2,v>.
At each IP phone, press the ?button twice. Phonel-.v and Phonc2-.v should use the
G729 codec and Phone3-j should use the G.711 codec. This is because the
conference bridge is at the branch and G.7I I is only allowed locally at the branch
butnotbetween branch and headquarters. Keep the call open.
Step 3
At BR-.r. enter the show dspfarm cisp all command. You should see three used
connections representing the three conference participants.
Step 4
r.nd the conference. Verifv thai all DSP resources arc freed by entering the sho»
dspfarm dsp all command again.
Step 5
Repeat the previous steps but initiate the conference from Phone I-x or Phonc2-.v.
1his time Phonel-.v and Phone2-Jc should use G.7I I and Phone3,v should use G.729.
The show dspfarm dsp all command will indicate that no conference resources are
used at BR-.v. Issue the same command at I\Q-x and you will see atranscoder
session for the connection of Phone.V.v to the conference bridge.
Task 5: Implement Multicast MOH from Branch Router Flash
In this task vou will first implement multicast MOII. Then you will implement multicast MOII
from branch'routcr Hash. This allows branch users to listen to MOH. but prevents the MOII
streams from being sent over the IP WAN.
Activity Procedure
Complete these steps:
Enable Multicast MOH in Cisco Unified Communications Manager
Step 1
In Cisco Unified Communications Manager Administration, navigate to Media
Resources >Music On Hold Audio Source and click the Find button.
Step 2
Choose the onlv available audio source and verify that the Play Continuously check
box is checked.
Step 3
Check the Allow Multicasting check box.
Step 4
Click Save.
Step 5
Nav igate to Media Resources >Music On Hold Server and click ihe Find button.
Step G
Click the only available MOI Iserver (HO-SW-MOH).
Step 7
Under Multicast Audio Source Information, check the Enable Multicast Audio
Sources on This MOH Server check box.
Steps
Click Save.
Step 9
Verify the following parameters under Multicast Audio Source Information:
Step 10
•
Base Multicast IP Address: 239.1.1.1
•
Base Multicast Port Number: 16384
Change the Increment Multicast value from Port Number to IP Address.
Step 11 Under Selected Multicast Audio Sources, set the Max Hops value for the multicastenabled audio source to 3.
Step 12
Click Save.
Step 13
Reset the MOII server.
Lab Guide
© 2010 Cisco Systems, Inc.
Step 17
Enter HQ_mrg for the name of the MRG.
Step 18
Enter HQ SW Conference Bridge for the description.
From the Available Media Resources pane, add HQ-SW-CFB to the Selected Media
Step 19
Resources list.
Step 20
Click Save.
Step 21
Repeat the previous steps to add the other two MRGs, as described in the "Media
mh
Resource Group Configuration" table in theJobAids section.
Create MRGLs
In these steps, you will configure dilTerenl MRGLs that allow IP phones to use their local
t*
Step 22
Navigate to Media Resources> Media Resource Group List.
**
Step 23
Click the Add Newbutton.
__
Step 24
Enter HQ_mrgl for the name of the MRGL.
conference bridge.
Step 25
From the Available Media Resource Groups pane, add the HQ-SW-CFB and
m,
Step 26
Click Save.
»
Step 27
Repeat the previous steps to add the other MRGL, as described in the "Media
Resource Group List Configuration1" table in the Job Aids section.
H
General_mrg MRGs to the Selected Media Resource Groups list.
Assign MRGLs to Devices
In these steps, you will assign the newly created MRGLs to devices (phones, trunks and
gateways) by configuring the appropriate MRGL in the available device pools.
Step 28
Navigate to System > Device Pooland click Find.
Step 29
Choosethe device pool Default.
'
m,
tt
Step 30
From the Media Resource Group List drop-down menu, choose IIQ_mrgl.
Step 31
Step 32
Click Save.
Reset the device pool.
Step 33
"*
W
m
Repeat the previous steps for the other two device pools, assigning the MRGLs as
described in the "Device Pool Configuration" table in the Job Aids section.
M
Activity Verification
You have completed this task when you attain these results:
•
• BR-.v provides aconference hardware media resource (BR-HW-CFB) lhat is registered
with Cisco Unified Communications Manager. Perform the following steps at router BR-v
m
""
to verity the hardware media resource configuration and status-
Step 1
Step 2
Enter the show seep command. Verify that the Conferencing Oper Stale is ACTIVE
and that the TCP Link Status is CONNECTED.
m
Enter the show dspfarm profile 1command. Verify the stains, number ofavailable
resources, and the listof supported codecs.
m
• Conferences that are initiated by headquarters phones use the software eonlerenee bridge
that ,s located at the headquarters; branch phones use the hardware conference bridge that
is located at the branch. Verify this by performing the following steps-
60
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Imptemenflng Cisco Unified Communications Manger, Part 2(CIPT2) v8.0
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"
Step 3
To configure the router DSP resources that are to be used as ahardware conference
bridge, enter this sequence ofcommands:
voice-card 0
dspfarm
dsp services dspfarm
seep local loopbackO
seep ccm 10.x.1.1 identifier 1 version 7+
seep
seep ccm group 1
associate ccm 1 priority 1
associate profile l register BR-HW-CFB
1
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 2
associate application SCCP
no
Step 4
shutdown
Save the configuration to NVRAM.
Add the Cisco IOS Hardware Conference Bridge to Cisco Unified Communications Manager
for the Branch
Step 5
Nav igate to Media Resources >Conference Bridge.
Step 6
Click Add New
Step 7
From the Conference Bridge Type drop-down menu, choose Cisco IOS Fnhanced
Conference Bridge.
Step 8
Note
Step 9
Inter BR-HW-CFB for ihe name ofthe media resource.
Themedia resource name is case-sensitive.
Lnter Hardware Conference Bridge at BR-a" for the description.
Step 10 Apply device pool BR to the transcoder. This device pool is configured with the
region BR.
Step 11
Set the device security mode to Non Secure Conference Bridge.
Step 12
Click Save.
Step 13
Reset the newly created conference bridge.
Step 14 Verily the registration status. It should say Registered with Cisco Unified
Communications Manager 10..V.L1.
Create MRGs
Step 15
Nav igate to Media Resources >Media Resource Group.
Step 16
Click the Add New button.
Lab Guide
© 2010 Cisco Systems. Inc
59
Step 6
From the Transcoder Type drop-down menu, choose Cisco IOS Knhanced Media
Termination Point.
Step 7
Enter Hardware Transcoder at HQ-x for the description.
Step 8
Enter HQ-HVV-XCD forthename of themedia resource.
Note
Step 9
The media resource name is case-sensitive.
Apply the device pool Default to the transcoder. This device pool is configured with
^
Step 10
Click Save.
j*
Step 11
Reset the newly created transcoder.
Step 12
the region HQ.
Verify the registration status. Itshould say Registered with Cisco Unified
""
Communications Manager 10jr. I. I.
Activity Verification
**
You have completed this task when you attain these results:
• HQ-* provides atranscoding hardware media resource (HQ-HW-XCD) that is registered
with Cisco Unified Communications Manager. Perform the following steps at router HQ-*
H
Step 1
Bl
toverify the hardware media resource configuration and status:
Enter the show seep command. Verify that the Transcoding Oper State is ACTIVE
§^
and that the TCP Link Status is CONNECTED.
Step 2
Enter the show dspfarm profile I command. Verify the status, number of available
&
resources, and the list of supported codecs.
• Branch phones can now join conferences on the G.711-only software conference bridge
even though they are not allowed to use the G.711 codec over the IP WAN. Verify this by
performing the following steps:
Step 1
1*
m
Set up an ad hoc conference with Phonel-*, Phone2-*, and Phone3-.v as members.
Step 2 At each IP phone press the ?button twice. Phonel-* and Phone2-.r should show the
^
Step 3
*
G.711 codec being used for the call, while Phone3-* shows G.729.
At HQ-.v. enter the show dspfarm dsp all command. You should see two used
connections representing the two call legs ofthe transcoder (G.711 tothe software
conference bndge and G.729 to Phone3-*).
Task 4: Implement a Hardware Conference Bridge
§fc
m
In this task, you will configure alocal hardware conference bridge at the branch You will
implement MRGs and MRGLs to ensure that IP phones use the local conference media
Mi.
resource.
am
Activity Procedure
jfc
Complete these steps:
Configure aCisco IOS Router as aHardware Conference Media Resource for the Branch
58
Step 1
Connect to your BR-*.
Step 2
Enter configuration mode.
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
f&
12010 Cisco Systems, Inc.
Task 3: Implement Transcoders
In this task vou will implement atranscoder at the headquarters in order lo allow branch users
to join confe'renccs on aG.711-only software conference bridge, even though branch phones
are not allowed to use G.711 over the IP WAN. The transcoder will transcode the (..729 audio
stream that is received from branch phones to aG.711 stream toward the software conference
bridge and vice versa.
Activity Procedure
Complete these steps:
Configure a Transcoder Media Resource in Cisco IOS Software
Configure the IIQ-.v router as a transcoder resource.
Step 1
Connect to \our HQ-.* and enter configuration mode.
Step 2 To configure router DSP resources to be used as atranscoder. enter this sequence of
commands:
voice-card 0
dspfarm
dsp services dspfarm
i
seep local loopbackO
seep ccm 10.x.1.1 identifier 1 version 7+
Note
The highest possible SCCP version that can be specified in the seep ccm command
depends on the Cisco IOS Software release that is used on the router.
seep
seep ccm group 1
associate ccm 1 priority 1
associate profile 1 register HQ-HW-XCD
dspfarm profile l transcode
codec g7llulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 2
associate application SCCP
no shutdown
Step 3
Save the configuration to NVRAM.
Add theTranscoder to Cisco Unified Communications Manager
Add the transcoder to Cisco Unified Communications Manager and assign adevice pool that
uses the region HQ.
Step 4
Navigate to Media Resources >Transcoder.
Steps
Click Add New.
Lab Guide
)2010 Cisco Systems. Inc
Step 35
Change thedevice pool from Default to BR.
Step 36
Click Save.
Step 37
Reset the gateway in Cisco Unified Communications Manager and reset the MGCP
process at the BR gateway by entering the no mgcp command, followed by the
morn command.
rnmmnniH
mgcp
I*
You already verified the device pool configuration (and hence the region assignment) of the
software media resources in the previous task. All software media resources are configured
Note
with the device pool Default
Activity Verification
You have completed this task when you attain these results:
• Place test calls between the following phones and while on acall press the 7button on the
IP phone two times. The IP phone will display call information that includes the codec thai
is used for the call:
—
Phone 1,r or Phone2-* and the PSTN (for example, 0 112): This call should use
G.711.
—
Phone I-x or Phone2-j; and any phone that is located in the other pod (for example
dial851y2001):ThiscallshoulduseC..729.
'
—
Phonel-j and Phone2-.r: This call should use G.722.
—
—
—
Phonel-.t orPhone2-;r and Phone3-*: This call should use G.729.
Phone3-.v and the PSTN (for example, dial 9 911): This call should use G.711.
Phone3-.v and any phone that islocated in the other pod (for cxumnle dial 851 v
2001 ):This call should use G.729.
Tip
You can also view information about active calls of an IP phone by using aweb browser to
browse to the IP address of the IP phone. The built-in web server of the phone provides
information about active RTP streams.
The built-in web server is disabled by default. You need to enable it when you want to
examine the information that is provided by the built-in web server. The built-in web server
can be enabled at the phone configuration page: setthe Web Access parameter to
Enabled
Note
You cannot add Phone3-x to aconference anymore. The only available conference bridge
(HQ-SW-CFB) is asoftware conference bridge running on Cisco Unified Communications
Manager. This software conference bridge supports G.711 only, Because Phone3-x is in
region BR and Ihis region is not permitted to use G.711 to region HQ (where the software
conference media resource is in), Phone3-x cannot join conferences anymore
56
implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems. Inc
&
Step 17
Verify that the region issetto HQ.
Step 18
At the relaled links, choose Back To Find/List and click Go.
Step 19
Choose the Branch device pool.
Step 20
Change the region lo BR.
Step 21
Click Save.
Step 22
Reset the device pool.
Step 23
Click the Add New button and configure anew device pool with following
parameters:
•
Name: Trunks
•
Cisco Unified Communications Manager Group: Default
•
Local Route Group: HQrg
Note
The local route group at the trunk is required in order to allow received TEHO calls that are
to besent to the BR gateway to bererouted via the backup path (standard local route
group). Until now, the trunk had the device pool Detault applied, which also has the local
routegroup HQ_rg configured.
Step 24
•
Date-TimeGroup: CM Local
•
Region: Trunks
•
SRST Reference: Disable
^^_
Click Save.
Apply Device Pools to Devices
In these steps vou will applv device pools to devices as speci tied in the "Device Pool
Configuration"' table in the Job Aids section. This will assign the appropriate regions to the
devices.
Step 25
Nav igate to Device >Phone and click Find.
Step 26 At the list of phones, verify that Phonel* and Phonc2-x are listed with device pool
Default and Phone3-.v is listed with device pool Branch.
Note
Regions cannot be directly applied to devices. You have to create different device pools with
regions and then apply the appropriate device pools to the devices.
Step 27
Navigate to Device >Trunk and click Find.
Step 28
Choose the SIP-Trunk.
Step 29
Change the dev ice pool from Default to Trunks.
Step 30
Click Save.
Step 31
Reset the trunk.
Step 32 Nav igate to Device >Gateway, choose the option to Show endpoints. and click
Find.
Step 33 Verifv that the HO-.vgaleway(lO^.l.lOl) is listed with the device pool Default.
Step 34 Choose the MGCP endpoinl ofgateway BR-x (SO/SUO/OSl-OfflJBR-* or similar).
Lab Guide
© 2010 Cisco Systems, Inc
55
• You can establish ad hoc conferences using the HQ-SW-CFB conference resource Verily
this by creating an ad hoc conference with Phonel*, Phone2-j, and Phone3-* as members.
• When acall is put on hold, the caller hears MOH. Verify this by establishing acall and then
putting the call on hold.
Task 2: Configure Regions
In this task, you will configure regions in order to prevent audio streams that are sent over the
IP WAN from using htgh-bandwidth codecs. Only G.729 will be permitted tor streams that
traverse the IP WAN. Refer to the table -Region Configuration" in the Job Aids section for
more details. The regions are then applied to devices using device pools.
Activity Procedure
Complete these steps:
Create and Configure Regions
Step 1
Navigate to System > Regionandclick Find.
Step 2
Choose the region Default.
Step 3
Change the name to HQ.
Step 4
Verify that G.722/G.7I Iaudio codecs are allowed for calls within region
headquarters.
Step 5
Click Save.
Step 6
Click the Add New button.
Step 7
Enter Trunks for the name ofthe new region and click Save.
Step 8
Using the Modify Relationship to other Regions pane, allow the G.729 audio codec
for calls within region Trunks by highlighting Trunks in the Regions list and
TJ-.»" /Villi- llJltliin Hr^x^-^^.^- T
1.1
I •
II-
• .•
»»,
-
choosing G.729 from the Audio Codec drop-down menu. Click Save.
Step 9
Using the same technique, allow the G.729 audio codec for calls between region
Trunks and region HQ.
Note
You must click Save after each change in the Modify Relationship to Other Regions pane
The changes will then appear in the Region Relationships pane.
Step 10
Click Add New.
Step 11
Step 12
Enter BR for the name ofthe new region and click Save.
Allow G.722/G.711 for calls within region BR.
Step 13
Allow G.729 for calls between regions BR and Trunks.
Step 14
Allow G.729 for calls between regions BR and HQ.
Create and Configure Device Pools
In these steps, you add anew device pool for the trunks and update the existing device pools
with the new regions, as described in the "Device Pool Configuration" table in .he Job Aids
otciions.
54
Step 15
Navigate to System >Device Pool and click the Find button.
Step 16
Choose the Default device pool.
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
>2010 Cisco Systems. Inc.
•to
Task 1: Enable Software Media Resources on Cisco Unified
Communications Manager
In this task vou will enable the Cisco IP Voice Media Streaming Application service, which
provides several software media resources running on Cisco Unified Communications
Manager. You will change the default names and descriptions of these media resources.
Activity Procedure
Complete these steps:
Verify That the Cisco IP Voice Media Streaming Application Service Is Activated
Step 1
Log in to Cisco Unified Serviceability and navigate to Tools >Control Center-
Step 2
Verify that the Cisco IP Voice Media Streaming App service is activated and
Feature Services.
running at CUCMl-.v.
Note
This service provides the following software media resources: Annunciator, Conference
Bridge, Media Termination Point, and Music on Hold Server.
Verify and Configure the Software Media Resources
Step 3
Iog in to Cisco Unified Communications Manager Administration and navigate to
Media Resources >Annunciator and click Find.
Step 4
Verify the status of the media resource. It should be registered with your Cisco
Step 5
Click the shown Annunciator media resource (ANN_2).
Step 6
Change the Name to HQ-SW-ANN and the description lo Software Annunciator
Unitied Communications Manager CUCMI-x (IO.r. I.I).
atCICMl-.Y.
Step 7
Verifv that the Device Pool is Default.
Step8
Click Save.
Step9
Reset the media resource.
Step 10 Repeal the previous steps for the following media resources using the specified
values:
Media Resource
Name
Description
Conference Bridge
HQ-SW-CFB
Software Conference Bridge at
Media Termination
HQ-SW-MTP
Software Media Termination
HQ-SW-MOH
Software MOH Server at
CUCMI-x
Point at CUCMI-x
Point
MOH Server
CUCMI-x
Activity Verification
You have completed this task when you attain these results:
• The Cisco IP Voice Media Streaming Application service is activated.
. The following software media resources are registered with CUCMl-.v: Annunciator.
Conference Bridge. Media Termination Point, and MOH Server.
Lab Guide
J2010 Cisco Systems. Inc.
•
H.323 gateway
•
PSTN with PSTN phone
Job Aids
These job aids are available to help you complete the lab activity.
Region Configuration
HQ
Trunks
HQ
G.711
G.729
G.729
Trunks
G.729
G.729
G.729
BR
G.729
G729
G.711
Note
BR
Region Default, which exists by default, will be renamed to HQ.
Media Resource Group Configuration
Name
Description
HQ_mrg
HQ SWConference Bridge
HQ-SW_CFB
BR_mrg
BR HWConference Bridge
BR-HW-CFB
General_mrg
Included Media Resource
&
HQ SW Annunciator, MOH and
HQ-HW-XCD
MTP; HQ HW Transcoder
HQ-SW-ANN
HQ-SW-MOH
HQ-SW-MTP
Media Resource Group List Configuration
Name
Contains I
ContainsMedia
Resource Groups
HQ_mrgl
HQ_mrg
General_mrg
BR_mrgl
BR_mrg
Generaljnrg
Device Pool Configuration
Device Pool
Region
Media Resource List
Applied to Device
Default
HQ
HQ_mrgl
Phone 1-x
Phone2-x
HQ-x
HQ-SW-CFB
HQ-SW-MTP
HQ-SW-MOH
HQ-SW-ANN
HQ-HW-XCD
Trunks
Trunks
HQ_mrgl
BR
BR
BR_mrgl
m
SIP_TRUNK
Phone3-x
BR-x (MGCP endpoint)
BR-HW-CFB
52
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems. Inc.
Lab 3-1: Implementing Bandwidth Management
Complete this lab activity to practice what you learned in the related module.
Activity Objective
In this activitv. vou will configure multicast MOH from branch router flash, regions, local
conference bridges, and Iranseoders to reduce bandwidth requirements on the IP WAN. After
completing this activ ity. you will be able to meet these objectives:
•
fnable software media resources on Cisco Unified Communications Manager
•
Configure regions
•
Implement transcoders
•
Implement a hardware conference bridge
• Implement multicast MOH from branch router flash
Visual Objective
The figure illustrates what vou will accomplish in this activity.
Lab 3-1: Implementing Bandwidth
Management
Implement
conference bridge
and transcoder
Use low-bandwidth
codecs only in WAN
Use low-bandwidth
codecs only in WAN.
Implementlocalconference bndge;
npiement multicast MOH from branch
router flash
Required Resources
These arc the resources and equipment that are required to complete this activity:
•
Cisco Unified Communications Manager
•
Student PC
•
Cisco IP Phones
•
Cisco IOS MGCP gateway
Lab Guide
© 2010 Cisco Systems. Inc
When prompted for the IP address of the FTP server, enter the IP address you wrote
Step 4
down in Step 2.
Step 5
For the source filename, enter moh.au.
Steps
Confirm the destination filename (moh.au) and wait for the file to be copied.
Step 7
Verify that the moh.au file isstored in flash by entering the show Hash command.
Enable MOH in SRST Mode
Step 8
Enable MOH at the branch by entering the following commands at BR-.v (in
configuration mode):
telephony-service
moh moh.au
Step 9
Savethe router configuration to NVRAM.
Activity Verification
You have completed this task when you attain these results:
• Branch phones can listen to MOH when put on hold. This can be verified by performing
the following steps:
Step 1
Place a call between aheadquarters phone and the branch phone.
Step 2
At the branch phone (Phone3-^). putthecall on hold.
Step 3
The headquarters phone should play MOH coming from the branch router.
Note
When you are finished, make sure to remove the access-list that you entered in an earlier
task to break the connection between BR-x and CUCMI-x from the serial interface atrouter
HQ-x. Verify that the Phone3-x re-registers with Cisco Unified Communications Manager.
&
50
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
>2010 Cisco Systems, Inc.
Activity Verification
You have completed this task when you attain these results:
• To verify the SRST Fallback configuration, enter the show telephony-service command on
the branch router BR-.v.
• To verity that the current flies are accessible to IP phones, enter the show telephonyservicetftp-bindings command.
•
To verify that SRST isworking, follow these steps:
Step 1
Break connective lo Cisco Unified Communications Manager by reapplying access
list 100 in the incoming direction atthe interface ofthe HQ-.r router that connects to
the BR-.v router.
Step 2
Place acall from aheadquarters phone to the branch phone (300I). The call should
work: the calling party number should be the 10-digit PSTN number ofthe
headquarters phone.
Step 3
Place acall from Phone3-.r to aheadquarters phone (use internal dialing: 2001 or
2002). The call should work: the calling party number should be the I0-digit PS IN
number of the branch phone.
•
lo verifv that ihe learned configuration was saved, display the configuration ofthe router:
— Enter the show running-config command and verify that you see an cphone-dn and
cphonc in the configuration.
Note
The next time the phone registers with Cisco Unified Communications Manager Express.
Cisco Unified Communications Manager Express uses the stored configuration instead of
learning the phone configuration using SNAP. In order to configure a phone with features
that cannot be learned by SRST. you can preconrigure the ephone-dn only (and then the
ephone is learned) or ephone-dn and ephone
_
Task 2: Configure MOH on Cisco Unified Communications
Manager Express
In this task, vou will configure Cisco Unified Communications Manager Express to provide
MOH lo Cisco IP phones.
Activity Procedure
Complete these steps:
Copy an MOH Audio File to the SRST Router
To use MOH with Cisco Unified Communications Manager Express in SRST mode, the MOII
file must be stored on the router flash.
Step 1
At PC-v. navigate to Start >Run and enter the cmtl command.
Step 2
In the command-line window, enter the ipconflg command to find out the IP address
of the PC. The IP address must be in network 10^.3.0/24. Write down the IP address
here:
10.
Step 3
-3.
Vour instructor reinstalled and preconfigured an FTP server at PC-*. AMOH audio
file was stored at PC-.v and made accessible via FTP for anonymous user. Copy this
tile to the Hash of vour BR-.v router by entering the copy ftp flash command:
Lab Guide
2010 Cisco Systems, Inc
•
H.323 gateway
•
PSTN with PSTN phone
Task 1: Configure Cisco Unified Communications Manager
Express in SRST Fallback Mode
In this task you will change from standard SRST to Cisco Unified Communications Manager
Express in SRST fallback mode.
Activity Procedure
Complete thesesteps:
Remove SRST Configuration from the Branch Router
Step 1
Step 2
Log in tothe BR-j router and enter configuration mode.
Delete the SRST command by entering the following command:
no call-manager-fallback
Note
The dial peers and translation profiles that are configured in the standard SRST lab will be
reused for Cisco Unified Communications Manager Express in SRST Fallback mode.
Configure Cisco Unified Communications Manager Express in SRST Mode on the Branch
Router
In these steps, you will configure Cisco Unified Communications Manager Lxpress in SRST
mode for the branch router.
Step 3
Enable Cisco Unified Communications Manager Express in SRSI mode by entering
the following commands:
telephony-service
ip source-address 10.x.250.102 port 2000
system message CUCME in SRST Mode
max-ephones 5
max-dn 5
srst mode auto-provision all
Note
The keyword all at the end of the srst mode auto-provision command causes the router to
save the learned ephone and ephone-dn configuration.
srst dn line-mode dual
srst ephone description SRST learned
create cnf-files
Note
The create cnf-files command makes the router generate configuration files that,
required by SCCP phones.
end
Step4
48
Save the router configuration.
Implementing Cisco Unrfied Communications Manager. Part 2(CIPT2>v8.0©2010 Cisco Systems, Im,
Lab 2-2: Implementing Cisco Unified
Communications Manager Express in SRST
Mode
Complete this lab activity to practice what you learned in the related module.
Activity Objective
In this activitv vou will configure Cisco Unified Communications Manager Express in SRST
mode to provide basic telephony services lo phones that lost the connection to Cisco Unified
Communications Manager. In addition, you will enable MOII. After completing this activity,
you will be able to meet these objectives;
• Configure Cisco Unified Communications Manager Express in SRST fallback mode
• Configure MOI 1on Cisco Unified Communications Manager Express
Visual Objective
The figure illustrates what you will accomplish in this activity.
Lab 2-2: Implementing Cisco Uni fsed
Manager Express in SRST Mode
Communications
Required Resources
These are the resources and equipment that arc required to complete this activity:
•
Cisco Unified Communications Manager
•
Student PC
•
Cisco IP Phones
•
Cisco IOS MGCP gatewa;
Lab Guide
& 2010 Cisco Systems, Inc.
Step 16
Step 17
Place atest call to the branch phones of the other pod using site-code dialing (852v
300I).
Save your configuration changes.
Activity Verification
You have completed this taskwhen you attain these results:
• You can receive calls at Phone3-jr when the calls are placed to the PSTN number of
Phone3-.v.
• You can place outgoing calls to the PSTN from Phone3-x The calling party number should
always be shown as 10-digit PSTN number at the PSTN phone. Make sure to place test
callsto the following types of destinations:
—
—
—
Local destinations, for example by dialing 9-555-5678
National destinations, for example by dialing 9-1-606-555-1234
International destinations, for example by dialing 9-011-44-555-666-7777
—
Emergency (911 and 9-911)
• You can call Phone3-.r from headquarters phones by using the internal directory number of
Phone3-.T(300I).
Note
CFUR, which is required at the main site in this scenario, was already configured in the
previous task. In this task you enabled incoming PSTN calls sothat received CFUR calls
can be routed to the internal number of Phone3-x.
From Phone3,v. you can use the internal numbering plan to reach sites within the pod .
the other pod asdescribed in the activity procedure.
Note
Caution
Detailed verification was part of the activity procedure.
M»
When you are finished, make sure to remove the access list at HQ-x that you entered in an
earlier task to break the connection between BR-x and CUCMI-x from the serial interface at
router HQ-x. Verify that the Phone3-x re-registers with Cisco Unified Communications
Manager.
46
implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems. Inc.
Si
Configure Digit Manipulation at the Branch SRST Router for Outgoing Calls
In these steps, you will configure digit manipulation to ensure that PSTN format is used for the
calling partv numbers onoutgoing calls.
Step 7
Configure atranslation profile, which adds 52r555 to the four-digit directory
numbers starting with 3by entering the following commands:
voice translation-rule 2
rule 1 /'3/ /52X5553/
exit
voice translation-profile pstn-out
translate calling 2
exit
Step 8
Bind the translation rule to the voice port that connects to the PSTN by entering the
following commands:
voice-port 0/0/0:23
translation-profile outgoing pstn-out
exit
Step 9
Save your configuration changes.
Step 10 Verily that the calling number ofthe outgoing calls is now placed with the PSTN
number of Phone3-.v (52r 555 3001).
Configure Digit Manipulation at the Branch SRST Router to Allow Internal Dialing for
Intersite Calls
In these steps, vou will configure the SRST gateway in such away that users can place calls to
other sites using internal numbers instead ofPSTN numbers.
Step 11
Configure an outgoing dial peer that matches the internal directory numbers of
headquarters phones and adds the appropriate prefix:
dial-peer voice 2000 pots
destination-pattern 2...
port 0/0/0:23
prefix 0115551x5552
Step 12
Place atest call to one ofthe headquarters phones using its four-digit internal
directory number.
Step 13 Configure an outgoing dial peer that matches internal site-code dialing toward the
headquarters of the other pod and modifies the called number appropriately:
dial-peer voice 8y2 pots
destination-pattern 851y2...
port 0/0/0:23
prefix 0115551y5552
Step 14 Place atest call to one ofthe headquarters phones ofthe other pod using site-code
dialing (851 v2001 or851 v2002).
Step 15 Configure an outgoing dial peer that matches internal site-code dialing toward the
branch ofthe other pod and modifies the called number appropriately:
dial-peer voice 8y3 pots
destination-pattern 852y3...
port 0/0/0:23
prefix 152y5553
Lab Guide
© 2010 Cisco Systems. Inc
at
Configure the Branch SRST Router toAllow Outgoing Calls
In these steps, you will add adestination pattern to the existing POTS dial peer Iin order to
S*
Step 5
•
allowoutgoing PSTN calls.
At BR-.t. enter the following commands in configuration mode:
dial-peer voice 2 pots
destination-pattern 9011T
Ifc
prefix Oil
port 0/0/0:23
Jfe
^L
dial-peer voice 3 pots
destination-pattern 911
m
prefix 911
port 0/0/0:23
dial-peer voice 4 pots
destination-pattern 9911
,»,
prefix 911
••
port 0/0/0:23
dial-peer voice 5 pots
destination-pattern 91[2-9] .. [2-9]
K
prefix 1
port 0/0/0:23
||L
dial-peer voice 6 pots
?*
destination-pattern 9 [2-9]
port 0/0/0:23
a*.
interface serial 0/0/0:23
isdn map address "Oil* plan unknown type unknown
Note
The ISDN switch type that is used at BR-x is primary-ni. This switch type automatically sets
the number type to international when the called number starts with 011 and has 12 more
digits, which can be the case in this lab. The PSTN, however, does not allow the type of
number to be used atthe BR site; only prefixes should be used. The shown isdn map
address command instructs the BR-x gateway not to automatically set the type of number to
international.
Step 6
Verify that outgoing calls are working by calling 9011 55 Six 5552001 from
Phone3-.r. Also try placing acall to the PSTN phone by dialing any valid PSTN
number (for example, 91606 555 4444). Note that the calling party number
displayed on the PSTN phone is the four-digit internal directory number olThoncl-v
Note
At this stage, branch phones are able to place calls to the PSTN. This includes calls to
headquarters phones if the headquarters phones are dialed by their PSTN numbers The
caU'ng party numbers of outbound PSTN calls use internal directory number format
44
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
©2D10 cisco Sys(emSi ,nc
m.
Step 2
Verifv using the debug isdn q931 command whether the call hits the HR-* gateway.
The call arrives at BR-.v. but DID is not enabled and the called party number is a 10digit number and not a4-digit director) number.
Note
Gateway BR-x accepts the call and, because DID is not enabled, it waits for dialed digits
(two-stage dialing) If you manually enter 3001 at this stage, Phone3-x will ring
Task 4- Implement a Dial Plan at the SRST Gateway Supporting
Inbound and Outbound Calls When in MGCP Fallback or in
SRST Mode or Both
In this task vou will configure adial plan at the SRST gateway that allows incoming calls that
are placed to the F 164 PSTN number of the branch phones to be sent to the appropriate
directorv number. In addition, you will implement PSTN access for branch users, finally, vou
will allow branch users to place calls to headquarters phones over the PS INby dialing internal
director, numbers.
Activity Procedure
Complete these steps:
Configure the Branch SRST Router toAllow Incoming Calls
In these steps. >ou will configure an inbound dial peer to allow incoming calls to be routed
correcllv.
Step 1
l.nable DID for the voice port that connects to the PSTN by entering the following
commands:
dial-peer voice 1 pots
incoming called-number 52x5553...
direct-inward-dial
port 0/0/0:23
Step 2
Configure atranslation profile to manipulate the incoming called number from the
PSTN (the complete PSTN number 52.x 55530(11) to the four-digit directorv number,
bv entering the following commands:
voice translation-rule 1
rule 1 /*52x5553/ /3/
exit
voice translation-profile pstn-in
translate called l
Step 3 Bind the newly created translation profile to the voice port that connects lo the
PSTN bv entering the following commands:
voice-port 0/0/0:23
translation-profile incoming pstn-in
Note
At this stage, branch phones are reachable by their PSTN numbers. CFUR from the
headquarters should now work
Step 4 Verifv that incoming calls are working by calling 3001 from PhoneKt or Ph0nc2,v.
Note 'that the calling parly number that is displayed at Phone3,v is the PS IN number
of the calling phone.
Lab Guide
© 2010 Cisco Systems. Inc
Task 3: Implement a Dial Plan in Cisco Unified
Communications Manager Supporting Outbound Calls Durina
SRST Mode
y
In this task, you will configure CFUR for remote phones to allow phones that are located at the
main siteto call remote phones viathe PS'fN in SRST mode.
Activity Procedure
Complete these steps:
Adjust CFUR Service Parameters
In these steps, you will adjust the CFUR-Max-Hop-Counter service parameter.
Step 1
Navigate to System >Service Parameters and choose the Cisco Unilied
Communications Manager publisher (IOjc.1.I).
Step 2
Step 3
From the Service drop-down menu, choose the Cisco CalfManager service.
In the Clusterwide Parameters (Feature—Forward) pane, change Ihe Max Forward
UnRegistered Hops toDN parameter to2 (default is0).
Step 4
Click Save.
Configure CFUR for the Branch Phone
Step 5
Navigate toCall Routing >Directory Number and click the Find button.
Step 6
From the result list, click directorv number3001, which is in the Internal partition.
Scroll to the Call Forward and Call Pickup Settings pane and enter the following
parameters in the Forward Unregistered Internal and in the Forward Unregistered
Step 7
Fxternal rows:
•
Destination: +6651x5553001
•
CSS: GlobaI_css
Step 8
Click Save.
Step 9
Reset the directory number.
Note
SRST mode will be active when IP connectivity between HQ and BR sites is broken. Calls
from HQ to BR will use the configured CFUR settings (destination and CSS). The CFUR
destination will match the VH translation pattern first, and then the \+6652x[2-9]XXXXXX
TEHO pattern. The first option of the route list that is applied to the TEHO pattern is the BR
gateway This, however, cannot beused, because IP connectivity between HQ and BR is
broken {the MGCP gateway isdown) Therefore, the second option of the route list is
used—the local route group.
Activity Verification
You have completed this task when you attain these results:
• To verify that Cisco Unified Communications Manager routes calls to the unregistered
numbers of phones that are in SRST mode, perform the following steps:
Step 1
42
Place acall from one ol'your headquarters phones to 300I. The call will fail.
Implementing Cisco Unified Communications Manager, Part 2<CIPT2)v8.0
©2010 Cisco Systems, Inc.
Step 2
Enter the following commands to enable and configure the SRST feature:
call-manager-fallback
ip source-address 10.x.250.102
max-dn 1 dual-line
raax-ephones 1
Configure the Gateway for MGCP Fallback Support
Step 3
hntcr the following command to enable the gateway fallback feature:
ccm-manager fallback-mgcp
Step 4
Specify with the following command that the default voice application (H.323) takes
over if the MGCP application is notavailable:
application
global
service alternate Default
Step 5
Save vour configuration changes using the copy running-config startup-con fig
command.
Activity Verification
You have completed this task when you attain these results:
• To verifv that SRSI' isworking on your branch router, perform the following steps:
Step 1
i; nter the debug ephone register command to start debugging.
Step 2
Fnter the terminal monitor command.
Step 3
To break connectivity to Cisco Unified Communications Manager, enter the
following commands at HQ-.v in global configuration mode:
access-list 100 deny ip any host 10.x.1.1
access-list 100 permit ip any any
interface serial
...
ip access-group 100 in
Note
Step 4
Use the interface that connects the HQ-x route with the BR-x router
Your Cisco IP phone in the branch should register with the BRI-jc SRST router. This
is indicated by the text "CM Fallback Service Operating" at the bottom of the phone
display.
Step 5 At the BR-.t gateway, you will see debug output indicating that the phone registered
with the SRST gateway. The last message should be "ephonel|l]:SkinnyCompleteRegistration."
Step 6
When you are finished, turn off all ofthe debug commands at all of the routers
using the nodebug all command.
Lab Guide
© 2010 Cisco Systems, Inc.
•
HJ23 gateway
•
PSTN with PSTN phone
Task 1: Configure SRST Gateways in Cisco Unified
Communications Manager
In this task, you will add an SRST reference, configure adevice pool with the SRST reference
and apply the device pool toremote phones.
Activity Procedure
Complete these steps:
Add aNew SRST Reference in Cisco Unified Communications Manager
Step 1
Create anew SRST reference with the following parameters.
•
Name: BR-x
•
Port: 2000 (default)
•
IP Address: 10jt.250.102 (loopbackO)
Configure the BR Device Pool withthe New SRST Reference
Step 2
At the BR device pool, set the SRST reference to BR-x.
Step 3
Reset the device pool.
Activity Verification
3j£
You have completed this task when you attain these results:
• Anew SRST reference is configured in System >SRST as described in the activity
m,
• The SRST reference is assigned to the branch IP phone. This can be verified at Ihe phone
gL
Step 1
Press the Settings button.
fife
Step 2
Choose option 3 Device Configuration.
procedure.
by performing the following steps:
mm
»>
•fr
Step 3
Choose option I CallManager Configuration.
Step 4
The second entry should read "CallManager 2SRST," and when you choose Select
the IP address of the loopback interface ofBR-* (10jc.250.I02) should be shown.
Task 2: Configure a Cisco IOS Gateway for MGCP Fallback and
SRST
h
*
^
In this task, you will configure SRST and MGCP Fallback support at aCisco IOS gateway.
Activity Procedure
lb
Complete these steps:
«.
m
Configure the Gateway for SRST
Step 1
40
Log in to your BR-x router and enter configuration mode.
Implementing Cisco Unified Communications Manager. Part 2(CIPT2) v8.0
M*
©2010 Cisco Systems, Inc.
Lab 2-1: Implementing SRST and MGCP Fallback
Complete this lab activity to practice what you learned in the related module.
Activity Objective
In this activ itv vou will configure Cisco Linified SRST to provide call survivability for Cisco
IP phones, and MGCP fallback for gateway survivability. After completing this activity, you
will be able lo meet these objectives:
• Configure SRSI gateways in Cisco Unified Communications Manager
• Configure aCisco IOS gateway for MGCP fallback and SRST
. Implement adial plan in Cisco Unified Communications Manager supporting outbound
calls during SRST mode
• Implement adial plan at the SRST gateway supporting inbound and outbound calls when in
MGCP fallback or in SRST modeor both
Visual Objective
The figure illustrates what you will accomplish in this activity.
Lab 2-1; implementing SRST and MGCP
Fallback
Required Resources
These are the resources and equipment that are required to complete this activity:
•
Cisco Unified Communications Manager
•
Student PC
•
Cisco IP Phones
•
Cisco IOS MGCP gateway
Lab Guide
>2010 Cisco Systems, Inc
39
Configure Calling Party Transformations for TEHO Calls
Step 2
Create calling party transformation patterns for the HQ gateway. Refer to the
"Localization ofCalling Party During Call Egress for Outbound TE110 Calls" table
of the Task Job Aids.
Note
The gateway isalready configured with a calling party transformation CSS that has access
to the partition that you applied to the newly created transformation patterns.
Place test calls to TEHO PSTN destinations located at the other pod. Use the debug
Step 3
isdn q93I command at the gateways ofthe other pod to verify that the TEHO
gateway is used. Make sure that for TEHO calls through the HQ gateway, the calling
number isthe number ofthe actual caller (in international format ifthe call comes
from the BR site ofthe other pod and in national format ifthe call comes from the
HQ site ofthe other pod) while the calling number for TEHO calls through Ihe BR
gateway isalways the number ofthe BR attendant (52^5553001).
Verify PSTN Backup for TEHO to the Other Pod
Step 4
Step 5
Shut down the serial interface that connects your pod with the other pod.
Repeat placing TEHO test calls from the HQ site and from the BR site to Ihe PSTN
destination located at the other pod. Although IP connectivity is broken, the calls
should still work, because they are rerouted via the second option ofthe route list:
the local route group. Use the debug isdn q931 command to verify that the call is
set up using the local PSTN gateway.
Step 6
Usethe no shutdown command on the serial interface.
Activity Verification
You have completed this taskwhen you attain this result:
• You can place TEHO calls to the other pod as described in the activity procedure.
• When IP connectivity between the two pods is broken, the local gateway is used as a
backup asdescribed intheactivity procedure.
38
Implementing Cisco Unified Communications Manager. Part 2(CIPT2) v8.0
©2010 Cisco Systems. Inc.
Task 6: Implement TEHO Between Pods
In this task, vou will configure TEHO between your pod and the other pod.
Task Job Aids
These job aids arc av ailable to help you complete the lab task.
Route Patterns for Outbound TEHO Calls
Route Pattern
V+5551yf2-9]XXXXXX
Configuration
Partition; System
Description TEHO to +55 51y(remote HQ)
UrgentPriority: deactivated
Gateway/Route List intersite _rl
H6652yf2-9]XXXXXX
Partition: System
Description TEHO to +66 52y(remote BR)
UrgentPriority: deactivated
Gateway/Route List: Intersitejl
Localization ofCalling Party During Call Egress for Outbound TEHO Calls
Calling PartyTransformation Pattern
\+55.51y5552XXX
Configuration
Partition. xfornvcg_HQ-out
Description: HQ (from remote HQ)
Discard DigitsInstructions: PreDot
Number Type: National
\+.6652y5553XXX
Partition: xform-cg__HQ-out
Description: HQ (from remote BR)
Discard DigitsInstructions: PreDot
Number Type. International
Note
The HQ PSTN allows remote calling numbers to be sent. The BR PSTN does not allow
other calling numbers than the number that is assigned to the PSTN line. Therefore, TEHO
calls from the other pod to HQ are configured to use the numbers of the other pod for the
calling number while calls going through the BR gateway have already been configured in
the previous task to use the number of the BR attendant (3001) for the calling number it the
calling number is not in the locally assigned DID range. Further, the HQ PSTN requires
number types to be set, while the BR PSTN expects 10-digit calling numbers without
number types
^
___
Activity Procedure
Complete these steps:
Configure TEHO Route Patterns
Step 1 Create route patterns for TEHO calls. Refer to the "Route Patterns tor Outbound
TEHOCalls" table of the Task Job Aids.
Lab Guide
2010 Cisco Systems. Inc
Configure TEHO Route Patterns
Step 3
Create route patterns for TEHO calls. Refer lo the "Route Patterns for Outbound
TEHO Calls" table of the Task Job Aids.
Configure Calling Party Transformations for TEHO Calls
Step 4
Create calling party transformation patterns for the HQ and BR gateways. Refer to
the••Localization of Calling Party During Call Egress for Outbound TEHO Calls"
table of the Task Job Aids.
Note
The gateways arealready configured with a calling party transformation CSSthathas
access tothe partition that you applied tothe newly created transformation patterns.
Step 5
Place test calls toTEHO PSTN destinations. Use the debug isdn q931 command to
w
verify that the TEHO gateway is used. Make sure that for IEl 10 calls through the
HQ gateway, the calling number is the number ofthe BR phone (in international
mt
wWi
format) while the calling number for TEHO calls through the BR gateway isnot the
IIQ number but the number ofthe BR attendant (52x5553001).
Mb
Verify PSTN Backup for TEHO Within the Pod
Step 6 Shut down the ISDN interface at the IIQ gateway and try placing aTEI10 call from
8
m
the BR phone. Because the primary path (through the TEHO gateway) does not
•"
work, the local gateway (BR) should beused asa backup.
Step 7
Use the no shutdown command on the ISDN interface.
8
Step 8
Shut down the ISDN interface at the BR gateway and try placing aTEHO call from
the HQ phone. Because the primary path (through the TEHO gateway) does not
work, the local gateway (HQ) should be used as a backup.
*&
'*im
Step 9
Use the no shutdown command on the ISDN interface.
m
Activity Verification
w
You have completed this task when youattain this result:
• You can place TEHO calls within your pod as described in the activity procedure.
• When IP connectivity between the two siles is broken, the local gateway is used as a
backup as described in the activity procedure.
36
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems, lire
8
B
%m
Task 5: Implement TEHO Within Your Pod
In this task, you will configure TEHO within your pod.
Task Job Aids
These jobaids arc available tohelp you complete the lab task.
Route Patterns for Outbound TEHO Calls
Route Pattern
Configuration
\+5551x[2-9]XXXXXX
Partition. System
Description TEHO to +55 51x (local HQ)
Urgent Priority deactivated
Gateway/Route List TEHO-HQ_rl
\+6652x[2-9jXXXXXX
Partition: System
Description: TEHOto +66 52* (local BR)
Urgent Priority: deactivated
Gateway/Route List TEHO-BR_rl
Localization of Calling Party During Call Egress forOutbound TEHO Calls
Calling Party Transformation
Configuration
Pattern
\+.6652x5553XXX
Partition: xform-cg_HQ-out
Description: HQ (from BR)
Discard Digits Instructions: PreDot
Number Type. International
Partition xform-cg_BR-out
Description: BR (from HQ)
Calling PartyTransform Mask: 52x5553001
Note
The HQ PSTN allows remote calling numbers to be sent. The BR PSTN does not allow
other calling numbers than the number that is assigned to the PSTN line Therefore, TEHO
calls from BR to HQ are configured to use the BR number as the calling number while calls
going through the BR gateway are configured to use the number of the BR attendant (3001)
for the calling number if the calling number is not of the locally assigned DID range. Further,
the HQ PSTN requires number types to be set while the BR PSTN expects 10 digit calling
numbers without number types
^
Activity Procedure
Complete these steps:
Configure Route Lists with the TEHO Gateways as First Option and the Local PSTN
Gateway as Backup
Step 1
Create aroute list that is called TEHO-HQ_rl and add the HQj-g and the Standard
Step 2
Create aroute list that is called TUIO-BR.rl and add the BR_rg and the Standard
Local Route Group item to the route list. Make sure that the HQ_rg is listed first.
focal Route Group item to the route list. Make sure that the BR_rg is listed first.
Lab Guide
© 2010 Cisco Systems. Inc
35
Step 11
Make sure that the callingnumberis shown with the internally used directorynumber and a site-code dial prefix at the receiving phone (85 ly-2001 and 851v-2002
for calls that are received from HQ phones of the other pod and 852_y-300l for calls
that are received from the BR phone of the other pod).
Verify PSTN Backup for the SIP Trunk
Step 12
When the test calls between your pod and the other pod have finished you can start
testingthe PSTN backup path. Shutdown the serial interface that connects your IIQ
router (HQ-*) to the HQ router of the other pod (HQ-_y).
Note
This is the serial interface at the HQ router that is configured with IP address 10.zx.101 and
subnet mask 255.255.0.0.
As mentioned earlier, x is the number of your pod, and y is the number of your partner pod
(in the same group). Groups are pods 1 and 2, pods 3 and 4, pods 5 and 6, and pods 7 and
8. A z in an IP address stands for the pod numbers of your pod and your partner pod. They
are listed in ascending order. Examples: for pod 1, x=1, y=2, and z=12; for pod 2, x=2, y=1,
and z=12; for pod 3, x=3, y=4, and z=34, for pod 4, x=4, y=3, and z= 34, and so on.
Step 13
Continue placing test calls from the HQ site and from the BR site to the other pod
using intersite dialing, Although IP connectivity is broken, the calls should still
work, because they are rerouted via the second option of the route list: the local
route group. Use the debug isdn q931 command to verify that the call is set up
using the local PSTN gateway. Verify that the calling number is still shown with the
internally used directory number and a site-code dial prefix.
Step 14
Use the no shutdown command on the serial interface.
Activity Verification
You have completed this task when you attain this result:
•
You can place calls to and receive calls from the other pod using site-code dialing as
described in the activity procedure.
•
When IP connectivity between the two pods is broken, the PSTN is used as a backup as
described in the activity procedure.
•
The calling number is always shown with the intemally used directory number and a sitecode dial prefix as described in the activity procedure.
i.
34
Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0
© 2010CiscoSystems.Inc.
Configure Route Patterns for Globalized Intersite Calls Using the Previously Created
Intersite Route List
Step 4
Create route patterns for intersite access. Refer to the "Route Patterns for Outbound
Intersite Calls" tabic of the Task Job Aids.
Configure Translation Patterns for Inbound Calls
Note
At this stage, you are ready to send calls to the other pod using globalized called and calling
numbers, in order to process the calls that are received from the other pod, you have to
change the received called number from globalized format to the internally used directory
number
This called number format change could be done with significant digits set to 4, configured
at the SIP trunk. However, as you want to use the SIP trunk also for TEHO in a later lab
task, the called number cannot be reduced to a four-digit number in general, only if the call
was placed to an internal phone and not when the call was placed to a TEHO destination.
Therefore, you will use translation patterns to modify the called number of inbound calls that
are received through the SIP trunk.
Step 5
Create translation patterns for received intersite calls. Refer to the "Changing the
Called Number of Calls Received Through the SIP Trunk" table of the Task Job
Aids,
Configure the SIP Trunk with a CSS That Has Access to Internal Phones
StepS
Apply the CSS Ininkjjss to the SIP trunk.
Step 7
Reset the trunk.
Step 8
When the configuration of the other pod has finished, you can start placing test calls
using intersite dialing. Dial 8-51v-2001 and 8-5ly-2002 to reach the HQ phones of
the other pod. dial 8-52v-300l to reach the BR phones of the other pod.
Note
When receiving calls from the other pod, you will see the calling number in globalized format
(+55-51y-555-2001 and +55-51y-555-2002 for calls from the HQ phones of the other pod
and +66-52/-555-3001 for calls from the BR phone of the other pod).
In order to indicate that the call is coming from an interconnected site, the configuration will
be changed in the next steps so that the calling number is shown with the internally used
site code.
Configure the Calling Number of Intersite Calls to Be Shown with Site Codes
Step 9
Create calling party transformation patterns for IIQ and BR phones. Refer to the
"Localization of Calling Party DuringCall Egress for Inbound Intersite Calls" table
of the Task Job Aids.
Note
Step 10
© 2010 Cisco Systems. Inc
The phones are already configured with a calling partytransformation CSS that has access
to the partition that you applied to the newlycreated transformation patterns.
Place lest callsbetween thetwo pods using intersite dialing. Make surethatyou
place calls in both directions. Dial 8-5!v-2001 and8-5l.y-2002 to reach the IIQ
phones of the otherpod. dial 8-52v-300l to reach the BRphones of the otherpod.
Lab Guide
Changing the Called Number of Calls Received Through the SIP Trunk
Translation Pattern
\+5551x555 2XXX
Configuration
Partition: Internal
Description: SIP (to local HQ)
CSS Trunkcss
Urgent Priority: deactivated
Discard Digits Instructions: PreDot
\+6652x555.3XXX
Partition: Internal
Description: SIP (to local BR)
CSS: Trunk_css
Urgent Priority: deactivated
Discard Digits Instructions: PreDot
Localization of Calling Party During Call Egress for Inbound Intersite Calls
Calling Party Transformation Pattern
Configuration
\+5551y555 2XXX
Partition: xform-cgJHQ-phones
Description: HQ-Phones {other pod HQ}
Discard Digits Instructions: PreDot
Prefix: 851y555
\+5551y555 2XXX
Partition: xform-cg_BR-phones
Description: BR-Phones (other pod HQ)
Discard Digits Instructions: PreDot
Prefix 851y555
\+6652y555 3XXX
Partition: xform-cg_HQ-phones
Description: HQ-Phones (other pod BR)
Discard Digits Instructions: PreDot
Prefix: 852/555
\+6652y555.3XXX
Partition: xform-cg_BR-phones
Description: BR-Phones (other pod BR)
Discard Digits Instructions: PreDot
Prefix: 852y555
Activity Procedure
Complete these steps:
Configure a Route Group for the SIP Intercluster Trunk
Step 1
Create route group SIP_rg and add the SIP trunk to the route group.
Configure a Route List with the SIP Trunk as First Option and the PSTN as Backup
Step 2
Create a route list that is called lntersite_rl and add SIP_rg and the Standard I.ocaI
Route Group item to the route list. Make sure that SIP_rg is listed first.
Configure Translation Patterns to Globalize Intersite Calls
Step 3
Create translation patterns for intersite access, Refer to the "Globalization of Called
and Calling Parties During Call Ingress for Outbound Intersite Calls" table of the
Task Job Aids.
32
Implemenling Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
© 2010 Cisco Systems. Inc.
Task 4: Configure Intersite Calling
In this task, you will configure intersite calling over the SIP trunk using site codes.
Task Job Aids
These job aids are available to help you complete the lab task.
Globalization of Called and Calling Parties During Call Ingress for Outbound
Intersite Calls
Translation Pattern
851/2XXX
Configuration
Partition: Global
Description: Intersite to HQ
CSS System__css
Urgent Priority: deactivated
Calling Party: Use external phone number mask: activated
Called Party Discard Digits Instructions: PreDot
Called Party Prefix Digits: +5551/555
852y.3XXX
Partition. Global
Description: Intersite to BR
CSS System_css
Urgent Priority deactivated
Calling Party: Use external phone number mask: activated
Called Party Discard Digits Instructions: PreDot
Called Party Prefix Digits: +6652/555
Route Patterns for Outbound Intersite Calls
Route Pattern
Configuration
\+5551y
Partition: System
5552XXX
Description HQ of other pod
Urgent Priority: deactivated
Gateway/Route List: lntersite_rl
H6652/
Partition System
5553XXX
Description: BR of other pod
Urgent Priority: deactivated
Gateway/Route List. Intersite_rl
© 2010 Cisco Systems, Inc
Lab Guide
Note
The calls fail because a callback is placed to the number as it was before localization took
place at the phone during call egress. Before localization, the number was in global format,
because the calling number received from the PSTN was globalized during call ingress.
In order to allow callbacks to globalized numbers, you have to add a \+! translation pattern.
The translation pattern will not change the called number, but only modifies the calling
number by applying the external phone number mask. Further, the translation pattern has a
CSS that has access to the \+! route pattern.
By adding such a translation pattern, you make sure that the calling number of the outbound
callback is globalized during call ingress and that you then match the \+! route pattern.
Applying the external phone number mask at the route pattern instead of the translation
pattern does not work before the configured global transformations are based on globalized
numbers. Digit manipulation that is configured at the route pattern and at the route list is
ignored by global transformations. Global transformations are based on the pre-transformed
number (that is, the number as it looks when hitting the route pattern), not on the
transformed number {that is, the number as it looks after route pattern or route list digit
manipulation has been applied).
Configure a Translation Pattern for Calls That Are Natively Using Globalized Format
Step 22
Create a translation pattern that is used by the HQ phones with the following
settings:
•
Translation Pattern: \+!
•
Partition: Global
•
Description: Calls to Globalized Numbers
•
CSS: System_css
•
Urgent Priority: activated
Note
Translation patterns are urgent by default. Make sure that you do not disable Urgent Priority.
•
Step 23
Calling Party: Use external phone number mask: activated
Retry placing callbacks from the HQ and BR phones by using the entries of the
received calls list. The calls should work this time. Make sure that the calling
number shown at the PSTN phone is using the correct format. Calls from the two
HQ phones should show a calling number of 5552001 and 5552002: calls from the
BR phone should show a calling number of 52.X5553001.
Activity Verification
You ha\e completed this task when you attain this result:
•
You can receive calls from the PSTN at HQ and BR phonesas described in Iheactivity
procedure.
30
•
PSTN calling numbers are shown in localized format as described in the activity procedure.
•
Callbacks can be placed from \iQ and BR phonesto the PSTN. The calling number for
callbacks is set as described in the activity procedure.
Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0
© 2010 Cisco Systems, Inc.
Verify Inbound PSTN Calls to the HQ Site
Place test calls from the PSTN to an HQ phone. Use the local, national, and inlemalional line of
the PSTN phone for the test calls.
Step 15
Note
When placing a call from the local line of the PSTN phone, the caller ID shown on
the display of the HQ phone should be 5554444.
You can call the HQ phones by any valid PSTN number, regardless of the line of the PSTN
phone that you use Valid numbers are: 555-2001 and 555-2002, 0-51x-555-2001 and 051x-555-2002, 00-55-51X-555-2001 and 00-55-51X-555-2002.
Step 16
When placing a call from the national line of the PSTN phone, the caller ID shoun
on the display of the HQ phone should be 06065554444.
Step 17
When placing a call from the international line of the PS'fN phone, the caller ID
shown on the display of the I IQ phone should be +776065554444.
Verify Inbound PSTN Calls to the BR Site
Place test calls from the PSTN to a BR phone. Use the local, national, and international line of
the PS'fN phone for the test calls.
Step 18
Note
When placing a call from the local line of the PSTN phone, the caller ID shown on
the display of the BR phone should be 5Zy555444.
You can call the BR phone by any valid PSTN number regardless of the line of the PSTN
phone that you use. Valid numbers are: 555-3001, 1-52x-555-3001, and 011 -66-52x-5553001
Step 19
When placing a call from the national line of the PSTN phone, the caller ID shown
on ihe display of the BR phone should be 6065554444.
Step 20
When placing a call from the international line of the PSTN phone, the caller ID
shown on the display of the BR phone should be +776065554444.
Verify Callbacks
Step 21
Place callbacks from the IIQ and BR phones by using the entries of the received
calls list. The calls will fail.
©2010 Cisco Systems, Inc
Step 3
Sa\e \our configuration changes using the copy running-config startu p-con fig
command.
Configure Routing to the Internal Directory Number at the HQ Gateway
Step 4
Configure the HQ gateway with significant digits set to 4, and set the CSS to
Trunkcss.
Configure Globalization of Calling PSTN Number During Call Ingress at the HQ Gateway
Step 5
Configure the HQ gateway with the following incoming calling party prefixes based
on the number type:
Step 6
•
Unknown: +
•
Subscriber: +555U'
•
National: +55
•
International: +
Reset the gateway.
Configure Routing to the Internal Directory Number at the BR Gateway
Step 7
Configure the BR gateway with significant digits set to 4, and set the CSS to
Trunk_css.
Configure Globalization of Calling PSTN Number During Call Ingress at the BR Gateway
Step 8
Configure the BR gateway with the following incoming calling party prefixes based
on the number type:
Unknown: +
Subscriber: +665Xv
National: +66
International: +
Step 9
Reset the gateway. Make sure that you also reset the MGCP process at the branch
router by entering the no mgcp command, followed by the mgcp command.
Configure Localization of Calling PSTN Number During Call Egress
Step 10 Create calling party transformation patterns for HQ phones. Refer to the table of the
Task Job Aids.
Step 11
Apply calling party transformation CSS xform-cg_HQ-phones ess to the IIQ
phones. Make sure that you clear the Use Device Pool Calling Party Transformation
CSS check box.
Step 12
Create calling party transformation patterns for BR phones. Refer to the table of the
Task Job Aids.
Step 13
Apply callingparty transformation CSS xform-eg_BR-phones_css lo the BR phone.
Make sure that you clear the Use Device Pool Calling Party Transformation CSS
check box.
Step 14
28
Reset all phones.
Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
—
—
National call: dial any valid national number, for example 9 1 888 666 444. The
National line at the PS'fN phone should ring: the called number that is shown in the
debug output should be 1888666444. and the number type should be unknow n.
International call: dial any valid international number, for example 9 011 88 222
3.33 4444. The Intemtl line at the PSTN phone should ring; the called number that is
shown in the debug output should be 911882223334444. and the number type
should be international.
Task 3: Configure Inbound PSTN Calls
In this task. >ou will configure inbound PSTN calls.
Task Job Aids
This job aid is a\ailable to help you complete the lab task.
Localization of Calling Party During Call Egress for Inbound PSTN Calls
Calling Party Transformation Pattern
Configuration
\+5551xXXXXXXX
Partition: xform-cg_HQ-phones
Description HQ-Phones (Local)
Discard Digits Instructions: PreDot
\+55 XXXXXXXXXX
Partition: xform-cgJHQ-phones
Description. HQ-Phones (National)
Discard Digits Instructions: PreDot
Prefix: 0
\+66 XXXXXXXXXX
Partition: xform-cg_BR-phones
Description: BR-Phones (Local, National)
Discard Digits Instructions: PreDot
Note
At the HQ site, end users expect to see local PSTN callers seven-digit numbers, national
callers with their national numbers and national access codes (0), and international callers
with a + prefix. Because allcaller IDs are globalizedat ingress, there is no need to modify
the caller ID of international callers when sending the call to the phone.
At the BR site, end users expect to see 10-digit caller IDs for local and national callers, and
see international callers with a + prefix (just like at the HQ site).
Activity Procedure
Complete these steps:
Configure the H.323 Gateway to Support Inbound PSTN Calls
Reconfigure the existing POTS dial peerto support inbound PSTN calls:
Step 1
dial-peer voice 2 pots
direct-inward-dial
incoming called-number
Step 2
.
Reconfigure the existing VoIP dial peerlo support inbound PSTN calls:
dial-peer voice 1
destination-pattern T
session target ipv4:10.x.1.1
©2010 Cisco Systems Inc
Lab Guide
Step19
Reset the BRgateway in Cisco Unified Communications Manager. Make surethat
you alsoresetthe gateway itselfby entering the no mgcpcommand and then the
mgcp command at the BR gateway.
Configure Localization of the Calling Number During Call Egress
Step 20 Create callingparty transformation patterns for the HQ gateway. Referto the
"Localization of Calling Party During Call Egress for Outbound PSTN Calls" table
of the Task Job Aids.
Step 21
Apply callingparty transformation CSS xfonn-cg_HQ-out_css to the IIQ gateway.
Makesure that you clear the Use Device Pool Calling PartyTransformation CSS
check box.
Step 22
Reset the HQ gateway in Cisco Unified Communications Manager.
Step 23
Applycallingparty transformation CSS xform-cg BR-out_css to the BR gateway.
Make sure that you clear the Use Device Pool Called Party Transformation CSS
check box.
Step 24
Reset the BR gateway in Cisco Unilied Communications Manager. Make sure that
you also reset the gateway itself by entering the no mgcp command and then the
mgcp command at the BR gateway.
Activity Verification
You have completed this task when you attain this result:
•
You can place calls to the PSTN from HQ phones. The HQ gateway is used for PSTN
access. Use the debug isdn q931 command for verification of the called and calling
numbers for all types of calls. The calling number should always be seven digits (555 2001
or 555 2002) with number type subscriber. The called number differs per destination:
—
Emergency calls: dial 112 and 0 112. The Emergency line at the PSTN phone
should ring; the called number that is shown in the debug output should be 112, and
the number type should be unknown.
—
Local call: dial any valid local number, for example, 0 333 4444. Ihe I,ocal line at
the PSTN phone should ring; the called number that is shown in the debug output
should be 3334444, and the number type should be subscriber,
—
National call: dial any valid national number, for example, 0 0 888 666 444. The
National line at the PSTN phone should ring; the called number thai is shown in the
debug output should be 888666444, and the number type should be national.
—
International call: dial any valid international number, for example 0 00 88 222
333 4444. The Intemtl line at the PSTN phone should ring; the called number that is
shown in the debug output should be 882223334444, and the number type should be
international.
•
You can place calls to the PSTN from the BR phone. The BR gateway is used for PSTN
access. Use the debug isdn q93l command for verification of the called and calling
numbers for all types of calls. The calling number should always be 10 digits (5Zv555
3001) with no number type set (unknown). 'Ihe called number differs per destination:
26
—
Emergency calls: dial 911 and 9-911. The Emergency line at Ihe PSTN phone
should ring: the called number that is shown in the debug output should be 911, and
the number type should be unknown.
—
Local call: dial any valid local number, for example 9 333 4444. The Local line tit
the PSTN phone should ring: the called number that is shown in the debug output
should be 3334444, and the number type should be unknown.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2)v8.0
© 2010 Cisco Systems, Inc.
Step 4
Save your configuration changes using the copy running-config startup-config
command.
Configure a System Route List Used for PSTN Access
Step 5
Create a route list that is called System_rl and add the Standard Local Route Group
item to the route list.
Configure Site-Specific Route Groups
Step 6
Create route group HQ_rg and add the HQ gateway to the route group.
Step 7
Create route group BR_rg and add the BR gateway to the route group.
Configure Site-Specific Device Pools and Set the Local Route Group
Step 8
Configure the default device pool with local route group HQ_rg.
Step 9
Create a new device pool that is called BR, and configure the device pool with local
route group BR_rg.
Step 10
Apply the ncuK created device pool BR to the BR phone.
Configure Globalization of Called and Calling Number During Call Ingress
Step 11
Configure all phones with an external phone number mask using globalized formal.
Use +555 Lv5552XXX for the HQ phones and +6652.V5553XXX for the BR phone.
Step 12
Create translation patterns for HQ PS'fN access. Refer lo the "Globalization of
Called and ("ailing Parties During Call Ingress for Outbound PS IN Calls Placed
from HQ Phones" table of the Task Job Aids.
Step 13
Create translation patterns for BR PS'fN access. Refer to the "Globalization of
Called and Calling Parties During Call Ingress for Outbound PSTN Calls Placed
from BR Phones" table of the Task Job Aids.
Configure Call Routing Based on Globalized Numbers
Step 14 Create a route pattern with the following sellings:
•
Route Pattern: \+!
•
Route Partition: System
• ( Description: PSTN_Access
•
Urgent Priority: activated
•
Gateway/Route List: System_rl
Configure Localization of the Called Number During Call Egress
Step 15 Create called party transformation patterns for the IIQ gateway. Refer to the
"Localization of Called Party During Call Egress for Outbound PSTN Calls" table
of the Task Job Aids.
Step 16
Apply called party transformation CSS xform-cd_IIQ-out_css to the IIQ gateway.
Make sure that you clear the Use Device Pool Called Party Transformation CSS
cheek box.
Step 17
Reset the HQ gateway in Cisco Unified Communications Manager.
Step 18
Apply called party transformation CSS xform-cd_BR-oul_css to the BRgateway.
Make sure that >ou clear the Use Device Pool Called Party Transformation CSS
check box.
)2010 Cisco Systems, Inc.
Lab Guide
Note
Because the HQ gateway (H.323) has a dial peer with destination pattern OT, PSTN access
code 0 has to be sent to the H.323 gateway; it will be stripped at the H.323 gateway when
the call is sent out to the PSTN. The PSTN expects the HQ gateway to send ISDN number
types; the called numbers must not include any prefixes.
For the BR gateway (MGCP) all digit manipulation is performed in Cisco Unified
Communications Manager. The PSTN expects the BR gateway not to use ISDN number
types; national and international calls have to have the corresponding prefixes.
Localization of Calling Party During Call Egress for Outbound PSTN Calls
Calling Party Transformation Pattern
Configuration
\+5551x5552XXX
Partition: xform-cg_HQ-out
Description: HQ (Local)
Discard Digits Instructions: PreDot
Calling Party Number Type: Subscriber
Partition: xform-cg_BR-out
\+66 52x5553XXX
Description: BR (Local)
Discard Digits Instructions: PreDot
Note
The calling party number that is sent to the PSTN at the HQ site should use the shortest
possible format (subscriber). The type of number must be set appropriately. The calling
party number that is sent to the PSTN at the BR site should be the 10-digit national number.
The type of number must not to be set.
Activity Procedure
Complete these steps:
Configure the H.323 Gateway to Support Outbound PSTN Calls
Step 1
Configure a codec voice class that allows G.711 and G.729 lo be used:
voice
class
codec
l
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
Note
At this time, all calls that are sent to the HQ-x gateway use G.711 only. However, in the next
lab exercise, TEHO will be enabled. At that time, G.729 will be used for TEHO callers.
Therefore all applicable codecs are configured now.
Step 2
Configure a VoIP dial peer for outbound PSTN calls:
dial-peer voice 1 voip
incoming called-number .
voice-class
Step 3
codec
1
Configure a POTS dial peer for outbound PSTN calls:
dial-peer voice 2 pots
destination-pattern OT
port 0/0/0:15
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
>2010 Cisco Systems, Inc.
Note
Only translation pattern 9.911 should be configured with urgent priority. AN other patterns
are unique and therefore urgent priority is not needed. Translation pattern 9.911 could also
match pattern 9[2-9]XXXXXX and is configured with urgent priority so users dialing 9.911
do not have to wait for the interdigit timeout to expire.
Localization of Called Party During Call Egress for Outbound PSTN Calls
Called Party Transformation Pattern
Configuration
\+5551xXXXXXXX
Partition: xform-cd_HQ-out
Description HQ (Local)
Discard Digits Instructions: PreDot
Prefix 0
Called Party Number Type: Subscriber
\+55. XXXXXXXXXX
Partition: xform-cd_HQ-out
Description: HQ (National)
Discard Digits Instructions: PreDot
Prefix: 0
Called Party Number Type: National
\+.!
Partition xform-cd_HQ-out
Description HQ (Intl)
Discard Digits Instructions: PreDot
Prefix: 0
Called Party Number Type: International
\+.112
Partition xform-cd_HQ-out
Description: HQ (Emergency)
Discard Digits Instructions: PreDot
Prefix: 0
Called Party Number Type: Unknown
\+S652x XXXXXXX
Partition xform-cd_BR-out
Description: BR (Local)
Discard Digits Instructions: PreDot
\+66 XXXXXXXXXX
Partition xform-cd_BR-out
Description: BR (National)
Discard Digits Instructions: PreDot
Prefix 1
\+ l
Partition: xform-cd_BR-out
Description: BR (Intl)
Discard Digits Instructions PreDot
Prefix 011
\+.911
Partition: xform-cd_BR-out
Description: BR (Emergency)
Discard Digits Instructions: PreDot
© 2010 Cisco Systems, Inc
Lab Guide
Globalization of Called and Calling Parties During Call Ingress for Outbound
PSTN Calls Placed from BR Phones
Translation Pattern
Configuration
9011.!
Partition: BR_PSTN
Description: BR (Intl, timeout)
CSS System_css
Urgent Priority: deactivated
Calling Party: Use external phone number mask: activated
Called Party: Discard Digits Instructions: PreDot
Called Party Prefix Digits: +
9011 !#
Partition: BR_PSTN
Description: BR (Intl, #)
CSS: System_css
Urgent Priority: deactivated
Calling Party: Use external phone number mask: activated
Called Party: Discard Digits Instructions: PreDot Trailing-*
Called Party Prefix Digits: +
91[2-9)XX
[2-9]XX
XXXX
Partition BR_PSTN
Description: BR (National)
CSS: System_css
Urgent Priority: deactivated
Calling Party: Use external phone number mask: activated
Called Party: Discard Digits Instructions: PreDot
Called Party Prefix Digits: +66
9 [2-9]XX
Partition. BR_PSTN
XXXX
Description: BR (Local)
CSS: System_css
Urgent Priority: deactivated
Calling Party: Use external phone number mask: activated
Called Party: Discard Digits Instructions: PreDot
Called Party Prefix Digits: +6652x
911
Partition: BR__PSTN
Description: BR (Emergency)
CSS: System_css
Urgent Priority: deactivated
Calling Party. Use external phone number mask: activated
Called Party Prefix Digits: +
9.911
Partition BR_PSTN
Description: BR (Emergency)
CSS: System_css
Urgent Priority: activated (default setting)
Calling Party: Use external phone number mask: activated
Called Party: Discard Digits Instructions: PreDot
Called Party Prefix Digits. +
22
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
©2010CiscoSystems, Inc.
Translation Pattern
0.112
Configuration
Partition HQ_PSTN
Description: HQ (Emergency)
CSS: System_css
Urgent Priority: deactivated
Calling Party: Use external phone number mask: activated
Called Party Discard Digits Instructions PreDot
Called Party Prefix Digits: +
©2010Cisco Systems, Inc
Lab Guide
•
You assigned partitions and CSSs to your IP phones. Each IP phone should still be able lo
call the other two IP phones. Verify this by placing test calls.
Task 2: Configure Outbound PSTN Calls
In this task, you will configure outbound PSTN calls using globalized call routing.
Task Job Aids
These job aids are available to help you complete the lab task.
Globalization of Called and Calling Parties During Call Ingress for Outbound
PSTN Calls Placed from HQ Phones
Translation Pattern
000.!
Configuration
Partition: HQ_PSTN
Descnption: HQ (Intl, timeout}
CSS: System_css
Urgent Priority: deactivated
Calling Party: Use external phone number mask: activated
Called Party: Discard Digits Instructions: PreDot
Called Party Prefix Digits: +
000 !#
Partition: HQ_PSTN
Description: HQ (Intl, #)
CSS. System_css
Urgent Priority: deactivated
Calling Party: Use external phone number mask: activated
Called Party: Discard Digits Instructions: PreDot Trailing-*
Called Party Prefix Digits: +
00.[2-9]XX
[2-9]XX
XXXX
Partition: HQ_PSTN
Description: HQ (National)
CSS: System_C5S
Urgent Priority: deactivated
Calling Party: Use external phone number mask: activated
Called Party: Discard Digits Instructions: PreDot
Called Party Prefix Digits: +55
0.[2-9]XX
XXXX
Partition: HQ_PSTN
Description: HQ (Local)
CSS: System_css
Urgent Priority: deactivated
Calling Party: Use external phone number mask: activated
Called Party: Discard Digits Instructions: PreDot
Called Party Prefix Digits: +5551x
112
Partition: HQ_PSTN
a»
Description: HQ (Emergency)
CSS: System_css
Urgent Priority: deactivated
Calling Party: Use external phone number mask: activated
Called Party Prefix Digits. +
20
Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Step 5
Step 6
Enter the following parameters in the Calling Search Space Configuration window:
•
Name: HQ-Phones_css
•
Description: HQ Phones
In the A\ ailable Partitions pane, choose the following partitions and use the down
arrow below the Available Partitions pane to move the highlighted partition to the
Selected Partitions pane. The order of the partitions is not relevant in this lab
exercise:
Step 7
Step 8
•
Internal
•
IIQ_PSTN
Click Save.
Repeat these steps to add the remaining partitions as they are listed in the "CSSs"
tabic of the Task Job Aids section.
Assigning Partitions and CSSs to IP Phones
In these steps, you will assign the appropriate partitions and CSSs to your IP phones.
Step 9
Na\ igate to Device > Phone and click the Find button.
Step 10
Click the IP phone with the directory number 2001 (Phonel-.v) to open the Phone
Configuration window.
Step 11
From the left column, click Line |l]—2001 (no partition) to access the Directory
Number Configuration window.
Step 12
Choose Internal for the Route Partition.
Step 13
Assign CSS HQ_Phoncs_css to the line.
Note
Make sure that you assign this CSS to the line level of the phone.
Step 14
from Related Links, chooseConfigure Deviceand click Go to get back lo the
Phone Configuration window.
Step 15
Assign CSS Global_css lo the phone.
Note
Make sure that you assign this CSS to the device level of the phone
Step 16
Click Save and then, in the pop-up window, click OK.
Step 17
Repeat these steps to for Phone2-.v.
Step 18
Repeat the above steps for Phone3-.v. butapply CSS RR Phones_css to the fine level
instead of CSS H()_Phoncs_css.
Step 19
Reset all three phones.
Activity Verification
You have completed this task when you attain this result:
•
You ha\e created thepartitions thatarelisted inihe"Partitions" table of the task Job Aids
section.
•
You ha\e created the CSSs that arc listed in the "CSSs" tabic of the Task Job Aids section.
) 2010 Cisco Systems. !nc
Lab Guide
CSSs
CSS Name
Description
Contains Partitions
Assigned to Devices
HQ_Phones_css
HQ phones
Internal
HQ Phone lines
HQ_PSTN
BR_Phones_css
BR phones
Internal
BR Phone lines
BR_PSTN
Global_css
General external
Global
All Phones (device level)
Internal
SIP trunk
access
Trunk_css
Calls received over
SIP trunk
System
System
Translation patterns
XF called, HQ out
xform-cd_HQ-out
HQ gateway
XF calling, HQ out
xform-cg_HQ-out
HQ gateway
xform-cd-BR-out_css
XF called, BR out
xform-cd_BR-out
BR gateway
xform-cg-BR-out_css
XF calling, BR out
xform-cg_BR-out
BR gateway
XF calling, HQ
phones
xform-cg_HQ-phones
HQ phones
XF calling, BR
xform-cg_BR-phones
BR phones
System^css
Route patterns
(globalized)
xform-cd-HQ-
out_css
xform-cg-HQout_css
xform-cgJHQphones_css
xform-cg_BRphones_css
phones
Activity Procedure
Complete these steps:
Configure Partitions
In these steps, you will configure the partitions lhat are listed in the "Partitions" table of the Job
Aids section.
Step 1
In Cisco Unified Communications Manager Administration, navigate to ("all
Routing > Class of Control > Partition, and click Add New.
Step 2
Fnter the partition names and the descriptions as listed in the Job Aids section using
this format:
Internal, Internal IP phones
Global, General patterns
Note
Step 3
Make sure to include all partitions as listed in the Job Aids section.
Click Save.
Configure CSSs
In these steps, you will create the CSSs listed in the "CSSs" table of Ihe Task Job Aids section.
Step 4
Na\ igate to Call Routing > Class of Control > Calling Search Space, and click
Add New.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
) 2010 Cisco Systems, Inc.
Task 1: Configure Partitions and CSSs
In this task. \ou will configure all partitions and CSSs that arc required for the following tasks.
You will assign partitions and CSSs to phones and phone directory numbers.
Task Job Aids
These job aids are available to help \ou complete the lab task.
Partitions
Partition Name
Description
Assigned to Phones or Patterns
Internal
Internal IP phones
Directory numbers of phones
Global
General patterns
Translation patterns that are used
for sile-code dialing and natively
dialed globalized numbers
HQ_PSTN
PSTN as dialed by HQ users
Translation patterns that are used
for HQ PSTN dialing
BR_PSTN
PSTN as dialed by BR users
Translation patterns that are used
for BR PSTN dialing
System
System patterns
Route patterns in globalized format
that are used for PSTN access,
TEHO, and intersite calls
xform-cd_HQ-out
XF called, HQ out
Called party transformation patterns
that are applied to the HQ gateway
for outbound calls
xform-cg_HQ-out
XF calling, HQ out
Calling party transformation patterns
that are applied to the HQ gateway
for outbound calls
xform-cd_BR-out
XF called, BR out
Called party transformation patterns
that are applied to the BR gateway
for outbound calls
xform-cg_BR-out
XF calling, BR out
Calling party transformation patterns
that are applied to the BR gateway
for outbound calls
xform-cg_HQphones
XF calling, HQ phones
Calling party transformation patterns
that are applied to HQ phones for
inbound calls
xform-cg_BR- phones
XF calling, BR phones
Calling party transformation patterns
that are applied to BR phones for
inbound calls
© 20' 0 Cisco Systems. Inc
Lab Guide
•
Cisco IOS MGCP gateway
•
H.323 gateway
•
PSTN with PSTN phone
Job Aids
Thesejob aids are available to help you complete the lab activity.
Phone Numbers
Location
IP Phone
E.164 Number with + Prefix
HQ-x
Phonel-x
+55 51x555 2001
Phone2-x
+55 51x555 2002
BR-x
Phone3-x
+66 52x 555 3001
HQ-y
Phonel -y
+55 51/555 2001
Phone2-y
+55 51/ 555 2002
Phone3-y
+66 52/555 3001
Site
IP Phone
Intersite Dialing
HQ-y
Phonel-y
8 51/2001
Phone2-y
8 51/2002
Phone3-y
8 52/ 3001
BR-y
Intersite Dialing
BR-y
PSTN Access Codes and Emergency Numbers
Location
PSTN Access Code
National Access
International Access
Emergency
HQ
0
0
00
112,0112
BR
9
1
011
911,9 911
Note
For additional information regarding the PSTN, refer to the Dial Plan Information section at
the beginning of this document
16
Implementing Cisco Unrfied Communications Manager, Part 2 (CIPT2) v8.0
>2010 Cisco Systems, Inc.
•to
Lab 1-2: Implementing a Dial Plan for
International Multisite Deployments
Complete this lab activity to practice what you learned in the related module.
Activity Objective
In ibis activity, you will implement a dial plan to support inbound and outbound PSTN calls,
site-code dialing. TFIfO. and PSTN backup. After completing this activity, you will be able to
meet these objectives;
m
•
Configure partitions and CSSs
•
Configure inbound PSTN calls
•
Configure outbound PSTN calls using H.323 and MGCP gateways
•
Configure site codes tor intercluster calls
•
Configure PSTN backup for intercluster calls
•
Configure TEHO with local backup
Visual Objective
The figure illustrates what you will accomplish in this activity.
Lab 1-2: Implementing a Dial Plan fc
International Multisite Deployments
Configure call
routing and digit
manipulation lor a
H 323 PSTN
gateway
Configure intercluster calls
using site codes, configure
PSTN backup for intercluster
calls, implement TEHO
Configure call
routing and digil
manipulation at
CUCMI-x for
MGCP PSTN
gateway
Required Resources
These arc the resources and equipment that are required to complete this activity:
•
Cisco Unified Communications Manager
•
Student PC
•
Cisco IP phones
) 2010 Cisco Systems. Inc.
Lab Guide
Note
Further verification will be done in the next lab exercise.
Task 5: Configure a SIP Trunk in Cisco Unified
Communications Manager
In this task. >ou will add a SIP trunk between the two pods.
Activity Procedure
Complete these steps:
Add a SIP Trunk in Cisco Unified Communications Manager
Step 1
In Cisco Unified Communications Manager Administration, choose Device >
Trunk.
Step 2
Click the Add New button.
Step 3
From the Trunk Type drop-down menu, choose SIP Trunk.
Step4
FortheTrunk Service Typedropdownmenu, leave None(Defaull). Click Next.
Step 5
In the Trunk Configuration window, enter the following parameters:
Step 6
•
Device Name: SIP_Trunk
•
Description: SIP Trunk to POD/
•
Device Pool: Default
In the SIP Information pane, enter the IP address of the other pod's Cisco Unilied
Communications Manager server: \0.y.\A.
Step 7
Step 8
Make sure that the Destination Address is an SRV box is not checked.
From the SIP Trunk Security Profile drop-down menu, choose Non Secure SIP
Trunk Profile.
Step 9
From the SIP Profile drop-down menu, choose Standard SIP Profile.
Step 10
Click Save, and then, in the pop-up window, click OK.
Step 11
Reset the newly added SIP trunk.
Activity Verification
You have completed this task when you attain these results:
•
The SIP trunk appears in the list when you choose Device > Trunk and then click the Find
button in Cisco Unified Communications Manager Administration.
Note
14
Further verification will be done in the next lab exercise.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
© 2010 Cisco Syslems. Inc.
Reduce the Utilized B-Channels from 23 to 8
Step 11
Disable the configuration server byentering the following command in global
configuration mode:
no ccm-manager config
Step 12
In global configuration mode, enter the following commands to shut down the voice
port that is associated with the Tl PRI:
voice-port 0/0/0:23
shutdown
Step 13
Disable PRI backhauling on the HI I) channel:
interface serial 0/0/0:23
no isdn bind-13 ccm-manager
Step 14
Remo\ c the auloconfigured PRI group and add a new one:
controller t l
0/0/0
shutdown
no pri-group timeslots 1-24 service mgcp
pri-group timeslots 1-8 service mgcp
no
Step 15
shutdown
Re-enable PRI backhauling:
interface serial
0/0/0:23
isdn bind-13 ccm-manager
end
Note
Because you deactivated the configuration server feature, the MGCP process at the Cisco
IOS gateway is not automatically reset anymore when you reset the gateway in Cisco
Unified Communications Manager. You have to manually reset the MGCP process at the
Cisco IOS gateway every time after you reset the gateway in Cisco Unified Communications
Manager Enter the no mgcp command, followed by the mgcp command, in order to reset
the MGCP process at the Cisco IOS router.
Activity Verification
You have completed this task when you attain this result:
•
Your MGCP gateway is successfully registered with Cisco Unified Communications
Manager. This successful registration can be verified at the gateway as follows:
LIsc the show ccm-manager hosts command. The status should show Registered.
—
—
Use the show mgcp endpoint command. All controlled ISDN PRI cndpoinl ports
should be up.
Use the show mgcp command. The Admin State and ihe Opcr State should be
active.
Use the show isdn status command. Ihe Layer 2 slate should be
MUI.TIPI.FJ:RAMFS F.STABUISIIF.D.
•
Verify that the MGCP gateway and the MGCP endpoinls are registered in Cisco Unified
Communications Manager:
Step 1
Navigate lo Device > Gateway.
Step 2
Choose the option lo Shaw endpoinls and click Find. The status of the endpoint
should be registered with IO.v.I.I.
©2010 Cisco Systems Inc
LabGuide
13
Activity Verification
You ha\e completed this task when you attain this result:
•
When youchoose Device > Gateway, choose theoption to Show endpoints. and click
Find, ihe MGCP gateway andits endpoints appear on the list.
Note
Further verification will be done in the next task.
Task 4: Configure an MGCP Gateway
In this task, you will configure a Cisco IOS MGCP PSTN gateway thatis located at ihebranch
to register with Cisco Unified Communications Manager.
Activity Procedure
Complete these steps:
Log in to BR-x
Step 1
From PC-.v. connectto your headquarters router(HQ-x) using Telnet to
10_v.250.l02. Log in using the password cisco and switchto enable mode(using the
password cisco again).
Discover the Current Gateway Configuration and Verify IP Connectivity
Step 2
Display the currentrouterconfiguration by enteringshow running-config.
Note
Step 3
The gateway is currently not configured with any MGCP commands. The T1 or E1 controller
is not configured with a PRI group command. There is no ISDN PRI.
Display the network interfaces and their IP configurations by entering Ihe show ip
interface brief command.
Step 4
Display the IP routing table by entering the show ip route command.
Configure the Cisco IOS Gateway for MGCP Using the Configuration Server Method
Step 5
Fnter the terminal monitor command to display the debug output that is generated
by the router.
Step 6
Enter the debug ccm-manager con fig-down load events command to debug the
configuration server feature events.
Step 7
In global configuration mode, enter the following commands:
ccm-manager config server 10.x.1.1
ccm-manager config
Step 8
Monitor the debug output to verify the operation of the configuration server feature.
Turn off all debugging by entering the no debug all command.
Step 9
Fnter the show running-config command. Your gateway should be configured for
MGCP, "fhe configuration that was added by the configuration server feature
includes MGCP, controller, and ISDN PRI settings.
Step 10
Save your configuration changes using ihe copy running-config startup-config
command.
12
Implementing Cisco UnrfiedCommunications Manager, Part 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Note
Further verification will be done in the next lab exercise.
Task 3: Add an MGCP Gateway to Cisco Unified
Communications Manager
Inthistask. >ou will addan MGCP gateway to Cisco Unified Communications Manager.
Activity Procedure
Complete these steps:
Configure an MGCP Gateway in Cisco Unified Communications Manager
Note
Step 1
These steps are platform-dependent. Cross-check with your instructor to find out the actual
hardware that is used for your MGCP gateway. This lab guide is based on the Cisco 2811
Integrated Services Router platform with a VW1C2-1MFT-T1E1-T1 in slot 0/0. You can also
use the show diag command at your router to display its hardware configuration.
In Cisco Unified Communications Manager Administration, choose Device >
Gateway and click the Add \cw button.
Step 2
Step 3
From the Gateway l ype drop-down menu, choose the gateway platform (for
example, a Cisco 2811 Integrated Services Router) that is used for Cisco IOS MGCP
gateway BR-.r. Click Next.
From the Protocol drop-down menu, choose the protocol type MGCP and click
Next
Step 4
[{nterthe following parameters in the Gateway Configuration window, and then
click Save:
•
Domain Name: BR-.v(in which x is your pod number)
•
Description: BR-.v, Branch PSTN Gateway (MGCP)
•
Cisco Unified Communications Manager Group: Default
•
Module in Slot 0: NM-4VWK-MBRD
•
Global ISDN Switch Type: NI2
Add MGCP Endpoints by Selecting Modules and Voice Interface Cards
Step 5
In ihe Configured Slots, VICs and Lndpoints pane, from Subunit 0 in Slot 0. choose
the module VWIC2-1MFT-T1F.I-T1. Click Save.
Step 6
Click the port icon on the right of the displayed endpoint 0/0/0.
Step 7
In the Gateway Configuration window, enter the following parameters:
•
Description: BR-*, Branch PSTN Gateway (MGCP): ISDN PRI
•
Device Poo!: Default
•
PRI Protocol Type: PRI NT2
•
Channel Selection Order: Top Down
Step 8
Click Save and. when the pop-up window appears, click OK.
Step 9
Reset the newh added MGCP endpoint.
<§ 20' 0 Cisco Systems. Inc
Lab Guide
HQ-.t. voice servers network: IOjc.1.101
HQ-x phone netwo k: IOjt.2.101
HQ-x data network 10.x3.I01
HQ-x serial interfaie to BR-x IOjt.6.102
HQ-x. serial interface to HQ-y: 10.z-.jc.101
BR-.r. loopback: Kjc.250.102
BR-x phone netwerk: IOjt.4.102
BR-xserial inter!a ;e to HQ-x 10.x6.102
CUCMl-y: lO.v.l.l
HQ-y. loopback: 10 v.250.101
HQ-y. voice servers network: lO.y.1.101
HQ-y. phone netwo -k: IO.v.2.101
HQ-y. data network: lO.v.3.101
HQ-y. serial interface to BR-y: I0.y.6.102
HQ-v. serial interfa> e to HQ-x- 10.z.y. 101
BR-y. loopback: I0v.250.102
BR-y. phone network: lO.y.4.102
BR-.v. serial interface to HQ-y: 10.y.6.102
Note
As mentioned earlier, x is the number of your pod, and y is the number of your partner pod
(in the same group). Groups are pods 1 and 2, pods 3 and 4, pods 5 and 6, and pods 7 and
8 A z in an IP address stands for the pod number of your pod that is followed by the number
of your partner pod. The z parameter is compounded in ascending order. Examples: for pod
1,x = 1, y = 2, soz = 12; for pod 2, x = 2,y= 1, soz= 12; for pod 3, x= 3. y = 4, soz = 34,
for pod 4, x = 4, y = 3, so z = 34, and so on.
Set the H.323 Interface
Step 7
Set the loopback interface of the router to be the source interface for all H.323
packets. Your configuration should look like the following (in which x is your pod
number):
interface loopback 0
h323-gateway voip interface
h323-gateway voip bind sreaddr 10.x.250.101
Additional configuration will be configured in the next lab activity.
Note
Step 8
Save jour configuration changes using copy running-con tig start up-con fig.
Activity Verification
You have completed this task when you attain these results:
•
Verity that H.323 is enabledon the headquarters router on the loopback interlace:
Enter the show running-config interface loopback 0 command to verify the 11.323
interface configuration df the HQ-.r router.
10
Implementing Cisco Unified Communications Mansger, Part 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Add an H.323 Gateway to Cisco Unified Communications Manager
Step 2
Navigate to Device > Gateway to display the find and List Gateways window.
Step 3
Click the Add New button.
Step 4
from the Gateway '1 ype drop-down menu, choose 11.323 Gateway.
Step 5
Click Next.
Step 6
In the Gateway Configuration window, enter and choose the following parameters:
•
Note
Device Name: 10_v.250.101
An x in device names or IP addresses stands for your pod number. Ask your instructor if you
are not sure which pod number to use.
•
Description: HQ-x IIQ PSTN Gateway (11.323)
•
Device Pool: Default
Step 7
Leave all other parameters at their default settings, and then click Save.
Step 8
In ihe pop-up window, click OK, and then reset the newly added gateway.
Activity Verification
You have completed this task when you attain these results:
•
You ha\e added a new H.323 gateway in Device > Gateway.
Note
Further verification will be done in the next lab exercise.
Task 2: Configure an H.323 Gateway
In this tusk. \ou will configure a Cisco IOS H.323 gateway for connecting to the PSIN at
headquarters.
Activity Procedure
Complete these steps:
Log in to HQ-x
Step 1
From PC-x connect to your headquarters router (HQ-x) using Telnet to
I0_v.250.l01. If prompted, log in using the password cisco and changelo enable
mode (using the password cisco again).
Discover the Current Gateway Configuration and Verify IP Connectivity
Step 2
Display the current router configuration by entering the show running-config
command.
Step 3
Step4
Display the status of your ISDN PRI by entering theshowisdnstatuscommand.
Display the network interfaces andtheir IP configurations by entering the show ip
interface brief command.
Step5
Display the IP routing table by entering Ihe show ip route command.
Step 6
Test IP connectivity to the following IP addresses using the ping or trace command:
)2010 Cisco Systems, inc
•
CUCMI-x 10-t.I.I
•
HQ-x loopback: I0_t.250.10I
Lab Guide
•
H.323 gateway
•
PSTN with PSTN phone
Job Aids
Thesejob aids are available to help you completethe lab activity,
Cisco Unified Communications Manager Information
Cisco Unified
IP Addr (as
Function
Login Administration User and
Password
Communications
Manager Name
CUCM-x
Publisher
10.x. 1.1
cucmadmin, cucmpassl (for Cisco
Unified Communications Manager
Administration)
admin, adpassl (for Cisco Unified
Communications Manager Operating
System administration)
;
Cisco IOS Gateway Information
Cisco IOS Gateway
IP Addrpss
Function
Login Password and Enable Secret
HQ
cisco, cisco
Name
10x.25<).101
HQ-x
gateway
10x.250.102
BR-X
BR
cisco, cisco
gateway
Cisco IP Phones Overview
Cisco IP Phone Name
IP Address
Directory
Location
Number
2001
Headquarters
DHCP subnet 10.X.2.0/24
2002
Headquarters
DHCP subnet 10.X.5.0/24
3001
Branch office
Phonel-x
DHCP subnet 10.X.2.0/24
Phone2-x
Phone3-x
Task 1: Add an H.323 Gatew
iy to Cisco Unified
Communications Manager
In this task, you will add the HQ-.v -1.323 gateway to Cisco Unified Communications Manager
Activity Procedure
Complete these steps:
Access Cisco Unified Communications IV anager
Step 1
Administration
From PC-.v. browse to h tps://10.v.Ll/ccniadmin and log in lo Cisco Unified
Communications Manaf er Administration using the information provided in the
Cisco Unified Commun
cations Manager Information table that is located in Ihe Job
Aids section of this guic
s.
Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0
) 2010 Cisco Systems, Inc.
Lab 1-1: Implementing Basic Multisite
Connections
Complete this lab activity to practice what you learned in the related module.
Activity Objective
In ihis activily. you will implement an H.323 gateway and an MGCP gateway for connecting to
the PSI'N and a SIP trunk for connecting to the other pod. AHercompleting this activity, you
will be able to meet these objectives:
Add an H.323 gateway to Cisco Unified Communications Manager
Configure an H.323 gateway
Add an MGCP gateway to Cisco Unified Communications Manager
Configure an MGCP gateway
Configure a SIP trunk in Cisco Unified Communications Manager
Visual Objective
'] he figure illustrates what you will accomplish in this activity.
Lab 1-1: Implementing Basic tVlultisi
Connections
Required Resources
fhese are the resources and equipment that arc required lo complete this activity:
•
Cisco Unified Communications Manager
•
Student PC
•
Cisco IOS MGCP gateway
© 2010 Cisco Systems, Inc.
Lab Guide
The figure pros idesa summary of (hedial plan.
Dial Plan Summs ry
intiyCode: 55
HQ-I
2 XXX
5552XXX
5552XXX
PSTN Access Code 0
51*S552XXX
Mati
55 5U555 2XXX
51y5552XXX
55 51/5552XXX
al Access Code: 0
iiS5..XXX
U.^.-J55-3XXX
:; »;xx x
b [2^:X< [i."•';>* xtxx
HQ-y
zxxx
Access Coda I
;;-?jxx \:
00-55-51 <-55S-£XX>;
55S-3XXX
G 'nTf' [2 u'XX *Xi*
'i :.ffi p.-j;xx <xx<
0-«i-555.3XXX
00*9 Ml-5 5.1-3 XXX
.Tf P'l
5!. 5 2 XXX
i [2 y)y.t
1 80C [J •
1 i'OO [2-
1 51iE5!)?XXX
(HIMMlBSSiXXX
5553XXX
5SXXXX
1 52* 555 3XXX
<X*XXX
011 SB 521 555 3XXX
5553XXX
y Code: 66
52I5553XXX
j Access Code: 9
66 521 555 3XXX
Ial Access Code: 1
555 3XXX
52/555 3X XX
66 52y 555 3XXX
nal Access Code: 011
Note
The blue text boxes
towards PSTN-Phone-x indicate how thenumber isdialed by HQ-x and
;s) and how the number has to be sent to the PSTN on the PRI
BR-x users (leftblue box£s)
(right blue boxes)
The orange text boxes tctwards the HQ-x and BR-x sites indicate how the number is dialed
by the PSTN user. The fll ure does not show how the numbers are delivered to the HQ-x
and BR-x gateways on It 3 PRI.
Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) vfl.O
12010 Cisco Systems, Inc.
As shown in the table, thecalling number that isused by the PSTN phone depends on the
PSf N phone line that is used to place the call.
Calling Number Presentation of Calls Placed from PSTN-Phone-x
PSTN-Phone-x Line
Calling Number Presentation
Local
556 4444, TON - subscriber
National
606 555 4444, TON = national
Intemtl
77 606 555 4444, TON = international
800
"PSTN"
Premium
none (CLIR)
Emergency
112. TON = unknown (when call is placed to HQ
sites)
911, TON - unknown (when call is placed to BR
sites)
Note
When calls are placed from line 800, the calling number is not provided, the calling name
"PSTN" is presented instead of a calling number.
When calls are placed from line Premium, CLIR is used.
fhe ISDN PRIs have to be configured as shown in the table.
ISDN PRI Configuration
HQ-x Gateways
BR-x Gateways
Controller type
E1
T1
esf
Framing
crc4
Line code
hdb3
b8zs
Clock source
line
line
Timeslot range
1-8, 16
ISDN switch type
primary
©2010 Cisco Systems, Inc
1-8.24
net5
pnmary-ni
Lab Guide
Calls Sent to PSTN-Phone-x
Received from BR-x
Sent to PSTN-
Sites
Phone-x Line
Received from HQ-x Sites
[2-9JXX XXXX
Local calls
TON = subscriber
National calls
international calls
Toll-free calls
[2-9]XXXXXX
Local
[2-9JXX [2-9JXX XXXX,
1 [2-9]XX 2-9]XX
TON = national
XXXX
(E.164 number of any length),
011 {E.164 number of
any length)
Intemtl
TON = international
800 [2-9]XX XXXX,
1 800 [2-9]XX XXXX
800
1 900 [2-9]XX XXXX
Premium
911
Emergency
National
TON = national
Premium calls
900 [2-9JXX XXXX,
TON = national
Emergency calls
112,
TON = Unknown
Note
The PSTN at the HQ sites allows remote calling numbers to be sent (for example in case of
TEHO or device mobility) The calling number that is sent to the PSTN through HQ
gateways should always be in the shortest-possible format. Calls originating at the local HQ
site should use local format, calls originating at the HQ site of the other pod should use
national format, and calls originating at one of the two BR sites should use international
format
The PSTN at the BR sites does not allow remote calling numbers to be sent. The calling
number that is sent to the PSTN through BR gateways should always be in national format.
Calls originating at any remote site (BR site of the other pod, one of the two HQ sites)
should use the number of the local BR attendant (52x-555-3001).
Calls to the HQ and BR sites can be placed from PSTN-Phone-*, as shown in the table:
Calls That Can Be Placed from PSTN-Phone-x
Number Dialed at PSTN-
Number Dialed at PSTN-
Phone-x, Sent to HQ-x Site
Phone-x, Sent to BR-x Site
Local calls
555 2XXX
555 3XXX
National calls
0 51X555 2XXX
1 52x 555 3XXX
International calls
00 55 51x 555 2XXX
011 66 52x555 3XXX
Note
The table shows the valid numbers that can be dialed at PSTN-Phone-x. No PSTN access
code is dialed from the PSTN phone. The presentation of the called number for these calls is
different The called number is always presented to the HQ and BR gateways without
prefixes and the TON is always set.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
) 2010 Cisco Systems, Inc
The table pro\ ides an overview of the dial plan that is in use.
Dial Plan Overview
HQ-x Site
BR-x Site
Internal directory number range
2XXX
3XXX
PSTN range in local (subscriber) format
555 2XXX
555 3XXX
PSTN range in national format
51x555 2XXX
52x 555 3XXX
PSTN range in international format
55 51* 555 2XXX
66 52x 555 3XXX
PSTN access code
0
9
National access code
0
1
International access code
00
011
Note
The x is a variable indicating your pod number.
The PSTN accepts calls from the HQ and BR sites as shown in the table.
Calls Accepted by the PSTN
Received from HQ-x Sites
[2-9]XX XXXX,
Local calls
TON = subscriber
National calls
51x[2-9]XXXXXX.
Received from BR-x Sites
[2-9]XX XXXX
1 52x [2-9]XX XXXX
TON - national
International calls
Toll-free calls
TON = international
(E 164 number of any length),
011 (E.164 number of any
length)
800 [2-9JXX XXXX,
1 800 [2-9]XX XXXX
TON = national
Premium calls
900 [2-9]XX [2-9]XXX,
1 900 [2-9]XX XXXX
TON - national
Emergency calls
Note
112,
TON = Unknown
911
When HQ gateways are sending calls to the PSTN, the gateways are required to set the
ISDN TON for the called number (see the table for details); national and international
prefixes must not be used.
When BR gateways are sending calls to the PSTN, the gateways are required to send
prefixes for national and international calls; the TON must not be set (it must be set to
Unknown).
The used PSTN numbenng plans do not fully represent the actual numbering plans that are
used in North America or Europe They only use some components of these PSTN
numbering plans.
Ifa dialed number matches the range that is used at one of the HQ or BR sites, the PSI'N routes
the call to that sile. All other valid calls are sent lo PSIN-Phone-.*. as shown in the table:
>2010 Cisco Systems. Inc.
Lab Guide
Lab Topology
The figure illustrates the labtopology andIP addresses.
Lab IP Network Overview
Phone 1-*
Phone2-y
Ptione--x
DHCPf*^
^^TJHCP ^^^DHCP
Note _ - x,
Phone 1-y
DHCPP*^
10 y 2 0/24
followed by y (in
ascending order)
101 10y30J2J
, 101 10700/16 ^101
DHCP t
10. 1 0/24
10yHQ1-J
HQ1-y\ "•
10 >_50 101/32 10
|
10y10C4
10 y 250 101/32
BR_y
Ptione3-y
10 y.250.102/3
1011 0/24
DHCP
4_ffiL_
(PSI^? 10jj4 0/24 T*'
iA~-*
*-_*
M02
DHCP
Pody
Ihe x in the figure indicates your pod number. They in the figure indicatesthe numberof the
pod that will work together with you.
Dial Plan Information
The dial plan is based on two pods,each of which has two sites. The headquarters sites of both
pods (HQ-.v and HQ-y) are located within the same country. The country code of the HQ sites is
55. The branch sites of both pods (BR-x and BR-y) are located within the same country (which
is different from the country that the HQ sites are in), 'fhe country code of Ihe BR sites is 66.
All sites use a fixed-length numbering plan that is like the one used in the NANP. The area
code is three digits long, followed by seven digits of subscriber numbers (grouped into three
and four digits).
The PSTN dial rules at the HQ sites are like the ones that are used in most European countries.
The PSTN dial rules in the country where the BR sites are located are like the ones that are
used in the NANP.
Implementing Cisco Unrfied Communications Manager. Part 2 (CIPT2) v8.0
© 2010 Cisco Systems. Inc.
CIPT2
Lab Guide
Overview
1 his guide presents the instructions and other infonnation concerning the lab activities for this
course. You can find the solutions in the lab activity Answer Key.
Outline
This guide includes these activities:
Lab 1-1
Implementing Basic Multisite Connections
Lab 1-2
Implementinga Dial Plan for Internalional Multisite Deployments
lab 2-1
Implementing SRST and MGCP fallback
Lab 2-2
Implementing Cisco Unified Communications Manager Kxpress in SRSI" Mode
Lab 3-1
Implementing Bandwidth Management
Lab 3-2
Implementing CAC
Lab 4-1
Implemenling Device Mobility
Lab 4-2
Implemenling Cisco Fxtension Mobility
Lab 5-1
Implementing Cisco SAP and CCD
Answer Ke\
Lab 3-2: Implementing CAC
ActivityObjective
Visual Objective
67
Required Resources
67
Job Aids
68
67
67
Task 1: Configure Locations
68
Task 2: Configure RSVP-Enabled Locations
69
Task 3: Configure AAR and CFNB to Route Calls over the PSTN If They Are Not Admitted by the
72
Deployed CAC Methods
Task4 (Optional): Configure SIP Pre[x>nditions
73
75
75
Lab 4-1: Implementing Device Mobility
Activity Objective
Visual Objective
Required Resources
Task 1: Configure Device Mobility
75
75
76
Lab 4-2: Implementing CiscoExtension lability
79
79
Activity Objective
Visual Objective
Required Resources
79
79
Task 1: Activate the Cisco Extension
Mobility Service and Configure the Corresponding Service
80
Parameters
Task 2: Create a Device Profile for a User
81
Task 3: Add and Associate an End U ser with the User Device Profile
82
Task 4: Add the Cisco Extension Mollility IP Phone Service and Subscribe to IP Phones and
Device Profiles
83
Lab 5-1: Implementing Cisco SAF and CI ;D
ActivityObjective
85
85
Visual Objective
Required Resources
85
85
Task 1: Configure SAF Forwarder Fu ictionality on the HQ-x and BR-x Router
86
Task 2: Configure Cisco Unified Corr nunications Manager as SAF Client
87
Task 3: Configure Cisco Unified Corr nunications Manager Express SRST on Branch Router to
91
Leam Routes Using CCD
Answer Key
96
Lab 1-1 Answer Key: Implementing Basic Multisite Connections
96
Lab 1-2 Answer Key: Implementing a Dial Plan for International Multisite Deployments
Lab 2-1 Answer Key: Implementing RST and MGCP Fallback
96
96
Lab 2-2Answer Key: Implementing (Jisco Unified Communications Manager Express in SRST
Mode
96
Lab 3-1 Answer Key: Implementing Ejandwidth Management
Lab 3-2
Lab 4-1
Lab 4-2
Lab 5-1
Answer
Answer
Answer
Answer
Key: Implementing
Key: Implementing
Key: Implementing
Key: Implementing
CAC
Cevice Mobility
Cisco Extension Mobility
. AF and CCD
Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0
96
96
96
96
96
) 2010 Cisco Systems, Inc.
Table of Contents
Lab Guide
Overview
Outline
Lab Topology
Dial Plan Information
Lab 1-1: Implementing Basic Multisite Connections
Activity Objective
Visual Objective
Required Resources
Job Aids
Task 1: Add an H.323 Gateway to Cisco Unified Communications Manager
Task 2: Configure an H.323 Gateway
Task 3: Add an MGCP Gateway to Cisco Unified Communications Manager
Task 4: Configure an MGCP Gateway
Task 5: Configure a SIP Trunk in Cisco Unified Communications Manager
Lab 1-2: Implementing a Dial Plan for International Multisite Deployments
Activity Objective
Visual Objective
Required Resources
Job Aids
1
1
2
2
7
7
7
7
8
8
9
11
12
14
15
15
15
15
16
Task 1: Configure Partitions and CSSs
17
Task 2: Configure Outbound PSTN Calls
20
Task 3: Configure Inbound PSTN Calls
27
Task 4: Configure Intersite Calling
31
Task 5: implement TEHO Within Your Pod
35
Task 6: Implement TEHO Between Pods
37
Lab 2-1: Implementing SRST and MGCP Fallback
39
Activity Objective
39
Visual Objective
39
Required Resources
39
Task 1: Configure SRST Gateways in Cisco Unified Communications Manager
40
Task 2: Configure a Cisco IOS Gateway for MGCP Fallback and SRST
40
Task 3: Implement a Dial Plan in Cisco Unified Communications Manager Supporting Outbound
Calls During SRST Mode
42
Task 4: Implement a Dial Plan at the SRST Gateway Supporting Inbound and Outbound Calls
When in MGCP Fallback or in SRST Mode or Both
43
Lab 2-2: Implementing Cisco Unified Communications Manager Express in SRST Mode
ActivityObjective
Visual Objective
47
47
47
Required Resources
Task 1: Configure Cisco Unified Communications Manager Express in SRST Fallback Mode
Task 2: Configure MOHon Cisco Unified Communications Manager Express
Lab 3-1: Implementing Bandwidth Management
Activity Objective
Visual Objective
Required Resources
47
48
49
51
51
51
51
Job Aids
Task 1: Enable Software Media Resources on Cisco Unified Communications Manager
52
53
Task
Task
Task
Task
54
57
58
61
2: Configure Regions
3: Implement Transcoders
4: Implement a Hardware Conference Bridge
5: Implement Multicast MOHfrom Branch Router Flash
5-74
Impiementing Cisco Unified Communications Manaj :f, Part 2(CIPT2) v8.0
>2010 Cisco Systems, Inc.
Module Self-Check Answer Key
OD
D.G
y_)
h.L)
04)
The toDID ruledescribes how to manipulate thelearned UN pattern in order to gelto thenumber h.
should be used for a PSTN backup call if the CCD path is unavailable.
(.)>)
i:
06)
D
©2010 Cisco Systems. Inc
Call Control Discovery
06)
Which CSS isused for CCD PSTN backup calls? (Source: Implementing SAF and
CCD)
A)
B)
5-72
line CSS of the originating phone
device CSS of the originating phone
C)
CSS of the SAF trunk
D)
K)
AAR CSS of the originating phone
CSS of the PSTN gateway
Implementing Cisco Unitied Communications Managt -, Part 2 (CIPT2) vB.O
at*
12010 Cisco Systems, Inc.
Module Self-Check
Use the questions here to re\ iew what \ou learned in this module, fhe correct answers and
solutions are found in the Module Self-Check Answer Key.
Ql)
Which two ofthe following devices do not support CCD? (Choose two.) (Source:
Implementing SAF and CCD)
Q2)
A)
B)
Cisco Unified SRST
Cisco IOS gateway
C)
I))
F.)
Cisco Unified Border Element
Cisco IOS gatekeeper
Cisco Unified Communications Manager
F)
Cisco Unified Communications Manager Express
G)
Cisco IOS Catalyst switches
Which two statements are true about SAF? (Choose two.) (Source: Implementing SAF
and CCD)
Q3)
A)
B)
C)
SAF forwarders interpret the SAF header and SAF service data.
An internal SAF client is allocated with a SAF forwarder.
An internal SAF client resides in Cisco Unified Communications Manager.
D|
SAF clients do not have to be Layer2-adjaccnt.
1.)
SAF requires FIGRP tobeused as the IProuting protocol.
Which two statements arc not true about CCD? (Choose two.) (Source: Implemenling
SAF and CCD)
A)
B)
C)
D)
F)
Call routing information is learned by the CCD requesting service.
Call routing infonnation is advertised by the CCD advertising service.
Load balancing occurs among trunk protocols and learned remote IP addresses.
Learned call routing information can be placed into different partitions that are
based on the remote call control identity.
Learned call routing information can be placed into different partitions that are
based on the remote IP address.
Q4)
What is the purpose of the toDID rule in CCD? (Source: Implementing SAF and CCD)
Q5)
Which of the following is not aconfiguration step when implementing SAF in Cisco
Unified Communications Manager? (Source: Implementing SAF and CCD)
A)
B)
Configure SAF forwarder.
Configure SAF trunk.
C)
D)
Configure CCD advertising and requesting service.
Configure hosted DN group and hosted DN pattern.
F)
F)
Configure DN blockprofile.
Configure blocked learned patterns.
) 2010 Cisco Systems. Inc
Call Control Discovery
5-71
5-70
Implementing Cisco Unrfied Communications Manaj er, Part 2 (CIPT2) v8.0
J2010 Cisco Systems, Inc.
Module Summary
This topic summarizes the key points that were discussed in this module.
Module Summary
CCD allows separate calf routing domains such as Cisco
Unified Communications Manager clusters and Cisco Unified
Communications Manager Express to utilize a SAF-enabled
network for dynamic exchange of call-routing information.
This module started with a description of Call Control Discovery (CCD). which is a feature that
allows callagents to advertise andlearn dial plan information to and from a CiscoService
Advertisement Framework (SAF)-enabled network. It showed how SAF" works, how CCD
utilizes SAF. and how SAF and CCD are implemented in a Cisco Unified Communications
solution.
References
For additional infonnation. refer lo these resources:
•
Cisco S\ stems. Inc. Cisco Unified Communications System 8.x SR\D. April 2010.
•
Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide
hup:. \sww.cisco.eom/cti/US/docs/voice ip_eomm/euem/snid/S*/ue8\.html
Release 8.0/1). February 2010.
lmp:;www.eisco.com.'en.'LS/docs/\'oicc ip comni/cucm/admin/8 OLccnicfg/hccm-801
cm.html
© 2010 Cisco Systems. Inc
Call Control Discovery
5-69
Summary
This topic summarizes the key points that were discussedin Ihis lesson.
Summary
• Dynamic distribution ofcall-routing information simplifies dial plan
implementation in large or very large networks.
• SAF allowsany services to be advertised to and learned from a
SAF-enabled network.
• CCD allows call agents toadvertise theinternal directory numbers that
they serve, along with the appropriate PSTN numbers, using SAF.
1When a learned VoIP route toa directory number becomes invalid, the
call is automatically rerouted over the PSTN.
SAF and CCD implementation includes the configuration of SAF
forwardersand SAF clients. SAFclients can be internal or external to the
Cisco IOS router used as a SAF forwarder.
Special considerations thatrelate to SAF and CCD implementation
include deployments using SRST, TEHO, globalized call routing, and
environments that have a SAF SIP trunk as well as a SAF H.323 trunk
References
For additional infonnation. refer to tlicse resources:
Cisco IOS Service Advertisemen Framework Configuration Guide 15.1 at
http://www .cisco. com/en/US/dt /ios/saf/configura£ion/guide/saf_cg _ps 105')2_TSD_Prod
uctsC'onfiguration GuidcC haft':er.html.
Cisco Systems. Inc. Cisco Unifiid
Uniji Communications System 8.x SRND, April
2010.
htlp:'V\\w\\.cisco.(jom/en/US.'''do s/voicejp eomm/cuem/sntd/8xAic8\.himl
CiscoSystems. Inc. Cisco Unifi
ifiei Communications Manager Administration Guide
Release 8.0(1). February 2010.
http://w ww.cisco.com/en.T ;S/doc s/voice_ip_comin/eucm/admin/8_0
l/ccmcig/hccin-801-
cm.html
Implementing Cisco Unified Communications Manage -. Part 2(CIPT2) v8.0
»2010 Cisco Systems, Inc.
Other SAF and CCD Considerations
This subtopic describes additional issues that you need to consider when implementing CCD in
Cisco Unified Communications Manager.
Other SAF and CCD Considerate
If you do not assign a trunk when you configure the CCD
requesting service, Cisco Unified Communications Manager will
not subscribe to hosted DN service from SAF network; there is no
learned pattern inside the system.
Each hosted DN pattern must be unique.
If a trunk is assigned to a route group or is associated with a route
pattern, you cannot enable SAF onthe trunk, and vice versa.
You cannot enable SAF on SIP trunks that use authenticated or
encrypted security profiles
Ifvou do not assign atrunk when you configure the CCD requesting service. Cisco Unified
Communications Manager uill not subscribe to the SAF forwarder. No routes will be learned.
Each hosted DN pattern must be globally unique.
Ifatrunk is assigned to aroute group or is associated with aroute pattern, you cannot enable
SAF on the trunk, and vice \ ersa.
You cannot enable SAF on SIP trunks that use authenticated or encrypted security profiles.
Call Control Discovery
©2010 Cisco Systems. Inc
Cisco Unified Communications Manager Clusters and CCD
Configuration Modes
This subtopic describes considerations regarding Cisco Unified Communications Manager
CCD configuration modes.
Cisco Unified Communications
Manager CCD Configuration Modes
• Basic configuration mode:
- All Cisco Unified Commurjications Manager servers use the same
primary or secondary SAP forwarders.
• Advanced configuration mode:
- Multiple sets of forwarder^ can beconfigured and individually applied to
Cisco Unified Communications Managerservers.
- This mode is required whe|n clustering overtheWAN deployment model
is used.
Single-Site Cluster c ^
Clustering ovpr the WAN
Cisco Unified
Communications
Manager Cluster
Primary
Secondary
SRF Forwarder SAF Forwarder
Primary
Secondary
SAF Forwarder SAFForwarder
When you are configuring aCisco Unified Communications Manager cluster with one ormore
SAF forwarders, by default, all Cisao Unified Communications Manager nodes that are applied
to the SAF-enabled trunk or trunks Via the device pool will register with the configured SAF
forwarder.
First, you have to make sure that each local node uses adifferent SAF client ID. You can easily
check the nodes by using a SAF client name that ends with @. In this case, each node will
utilize the configured name that is followed by @and aunique node ID. At the SAF forwarder,
you have toadd the basename keyword tothe end ofthe SAF external-client client-ID
command, or you have to manually ^onfigure all names that are used by the nodes in your
cluster.
In some cases,you may not want all nodes that should use SAF register with all configured
SAF forwarders. For example, as shbwn in the figure, when youuse clustering overthe WAN,
you typically want to registernodes only with their local SAF forwarders. For that
configuration, click the Show Advanced link at the SAF forwarder configuration page.
At the advanced configuration mode page, you can associate individual members of the cluster
selectively with the configured SAF forwarders.
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Trunk Considerations When Using Globalized Call Routing
This subtopic describes considerations regarding the different trunk types when using
globalized call routing.
Trunk Considerations When Using
Globalized Call Routing
• OneSAF SIPtrunk andone SAF H323trunk can be configured
for CCD
- If both are configured, load-sharing occurs.
• with globalized call routing, PSTN numbers areto be dialed in E.164
format with + prefix
TEHO calls received via SAF are assumed to have +in called number.
Hosted directory number isin globalized format with +prefix.
- H 323 trunks do not send + (+ is stripped).
- Requires incoming called-party prefix to be configured at H323 trunk
(prefix +).
- Otherwise, received TEHO call cannotbe routed outto PSTN.
- Difficult totroubleshoot when H.323 and SIPtrunks are configured
for SAF.
• Half of the callswork (SIP), and half ofthe callsfail (H.323)
As mentioned earlier, vou can configure one SAF-enabled SIP trunk or one SAf-enabled H.32.^
trunk in Cisco Unified Communications Manager and in Cisco IOS Software. II both types of
trunk are used bv the advertising SAF client and both are used at the requesting SAF client,
then all routes are learned twice—once per protocol. When placing acall to such aroute, load-
sharing will occur, This means that half of the calls are setup using SIP, and half of the calls
are signaled by H.323.
When implementing TEHO and using globalized call routing, TFHO calls are expected to be
received with a- prefix. The reason is that they are advertised that way and the incoming VoIP
call can be routed back out to the PSTN at the TEHO gateway when the called number is in
globalized format.
II 323 trunks however, do not send the +sign. When acall is received (or placed) through an
H.323 trunk and the called number includes a+. the +sign is stripped. This does not happen on
SIP trunks.
When vou rclv on the +to be received through the H.323 trunk, you have to configure
incoming called-partv settings at the 11.323 trunk. Consequently, the +is prefixed before the
received called-party number is matched in the call-routing table without the +.
If vou have aSIP and an 11.323 trunk, and you do not prefix the 4- at the H.323 trunk due to the
load-sharine algorithm, even' second call would fail (H.323) while the other halfofthe calls
would work (SIP). Such apparently inconsistent errors are difficult lo troubleshoot.
Note
When you expect to receive VoIP calls to internal directory numbers as well as to globalized
(PSTN) numbers, make sure that your incoming called-party settings prefix only the +to the
called numbers where it is required. You can either refer to the ISDN type of number or use
global transformations in order to control which called-party numbers you can modify.
Call Control Discovery
©2010 Cisco Systems, Inc
5-65
TEHO Considerations
This subtopic describes how CCD can be used to advertise TEHO routes.
TEHO Considerations
1TEHO destinations are located atthe PSTN and do not exist intemally.
• Advertised hosted DNis PSTNdestinationand not internal DN.
IfSAF-leamed TEHO route becomes unavailable, ToDID number is usedfor
automatic backup call.
- Typically, both numbers are identical (ToDID rule is 0:) anduse E 164format
with + prefix.
If no ToDID is advertised, calljfails until learned TEHO route is completely removed.
Learned route is removed only after expiration ofCCD PSTN Failover Duration
(default: 48 hours).
Cisco IOS Software can advertise +only in global patterns (which do not support
ToDID).
Avoid the use of global patterns in Cisco IOS Software.
• If - isrequired for TEHO implementation andlocal backup isdesired donot
advertise TEHO routes at'sites that use internal SAFclients.
TEHO destinations are located at tfye PSTN only; they do not exist internally at all. The
advertised directors' number is aP^TN number. When globalized call routing is enabled, this
number has to be in E. 164 format vvith a + prefix.
Calls toPSTN destinations will match the learned directory' number and are therefore sent to
the TEHO site over the IP WAN. lit case the IP WAN is down, the local gateway should be
used as a backup. This situation reduires having aToDID rule of0:. With this rule, the CCD
PSTN backup number is identical tp the learned directory number. When the IP path is marked
as unreachable, the same number would be called using the AAR CSS ofthe calling phone.
In Cisco Unified Communications Manager, generate aToDID rule of0: by checking the Use
Hosted DN as PSTN Failover check box. In Cisco IOS Software, you cannot set the ToDID to
Further, ifglobalized call routing is to be used, you are forced to use global patterns, which do
not allow any ToDID rule to be advertised.
When TEHO pattern is advertised without aToDID rule, local TEHO backup does not work
You could only configure static local backup routes by putting similar patterns into partitions
that are listed later in the phone CSS. Ilowever. such patterns are used only after the learned
pattern has been purged completely. By default, this process occurs after the expiration of the
CC DPSTN Failover Duration timer, which is 48 hours by default.
Based on these issues, it is recommended that you not advertise TEHO patterns from Cisco IOS
Software ifthe + is required and local backup is desired.
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
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»2010 Cisco Systems, Inc.
Solution for PSTN Backup Advertised in E.164 Format Without
Leading +
['his subtopic describes the workaround for CCD PS'fN backup calls lo PSTN destinations that
are in E.164 format without a •+ prefix.
Solution for PSTN Backup Advertis*
E.164 Format Without Leading +
Standard AAR Call
Called Number
Composed Irom
Called DN External
Phone Number Mask
2G01. -MC8S5Sv.XaX
Destination Phone
DN and hxlirriUl
• Si..
52001
Route Pattern
Pattern \+'
Partition AAR
Urgent Pnority
AAR CSS
Route List
Partitions AAR
PSTN
CCD PSTN Backup
Call
Called Number
Composed from
Learned DN Pall em
and 1c _'•[} F<"re
~J The +isprefixed for
\
callsplaced with
^ AAR CSS ifthe
called number does
not include +
already
It would be vcrv cumbersome to add all possible E. 164 numbers lo the overall dial plan of
Cisco Unified Communications Manager. However, because CCD PSTN backup calls are
always placed with the use of the AAR CSS of the calling phone, you can add one translation
pattern ' to apartition that is accessible only from the AAR CSS. At that pattern, you can prefix
a+to the called-party number and set the CSS of the translation pattern to aCSS that has
access to the global PS'fN route pattern (\+!).
That process solves the issue with CCD PS'fN backup calls. Ilowcver. ifAAR is enabled, it
will break AAR. assuming that the AAR implementation is based on globalized call routing. IS
the external phone number mask of the destination phone is in E. 164 format with a+prefix,
then AAR calls would not work anvmore. The reason is that they use the same CSS and
therefore would also match the !translation pattern that prefixes a+. In this case AAR calls
would be placed to E. 164 numbers with two +signs. In order to also make AAR CSS cal s
work vou have to add asecond translation pattern into the same partition that is accessible
from the AAR CSS onh -This second translation pattern \+! is not configured with any digit
manipulation but uses the same CSS as the other translation pattern (!). As aconsequence.
AAR calls are passed on to the V+! route pattern without any digit manipulation (by matching
the more specific \+! translation pattern, which docs not prefix a+). CCD PS 1Nbackup calls
do not match the Vi-! translation pattern and are therefore routes, as explained earlier.
Note
The described solution is a workaround only. The implementation of advertised patterns in
Cisco IOS Software may be changed in the future so that +can be configured in the
advertised pattern and in the ToDID rule. If so, you should change from the described
workaround to the solution that allows CCD PSTN backup calls to be placed to globalized
numbers.
.
Call Control Discovery
i_010 Cisco Systems,
5-63
Cisco IOS SAF Client Considerations When Using Globalized
Call Routing
This subtopic describes the limitations ofCisco IOS SAF clients regarding the use of the +sign
in advertised routes.
Cisco IOS SAF Client Considerations
When Using Globalized Call Routing
Cisco IOS internal SAF clients have limited support for +
sign when advertising patterns:
Solution for PSTN backup of internal DN:
- Advertise PSTN backup number in E.164 formatwithout +.
- Add + by translation pattern used forAAR calls to numbers
that do not start with +.
Cisco IOS Configuration
DN Pattern
and ToDID
p-atlla da-tlock 1 i l l u -
P-.-~_ 197.555
p«e-«B l tj-» uiuuios
4XXX
0:1972555
• The+ not supported In-lias-prefix or extension.
• PSTN backup canrroi be advertised with +.
• The + is added lo patternsconfigured with type
profil. da-block 2
P_tt«rn 1 type global
♦ 14087071222
global,
H0470712-S
(noToDID)
•Global patternsalways have ToDID unset.
•There is no PSTN backupforglobal patterns
Cisco IOS internal SAF clients have limited support regarding the +sign in advertised routes.
In fact, the +sign cannot be configured in either the directory number pattern or the ToDID
rule, "fhe only way to advertise apattern with +is to use the pattern tag type global command
instead the pattern tag type extension command. In this case, however, Ihe ToDID is always
unset, regardless ofthe configured alias prefix at the directory number block profile.
When aCCD-enabled Cisco Unified Communications Manager uses globalized call routing for
PST Naccess, the mentioned limitation ofCisco IOS internal SAF clients causes issues because
the backup PSTN number is not in a format that Cisco Unified Communications Manaeer can
route to the PSTN.
The workaround is to make sure that CCD PSTN backup calls can be routed to the PSTN even
if the number that results from the ToDID rule does notstart with the+.
Note
Cisco IOS Software has limitations only in advertising patterns that include the +sign. Cisco
IOS Software can process received patterns that include a +sign without any problems or
limitations.
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Implemenling Cisco Unified Communications Manager, Part 2(CIPT2) V8.0
©2010 Cisco Systems, Inc.
CCD and Static Routing Integration Considerations
This subtopic describes how CCD can be integrated with static routing in Cisco Unified
Communications Manager.
CCD and Static Routing Integration
Considerations
All routes that are learned by CCDare put intothe same partition.
If partition islisted first in CSS. il has pnority tor equally qualified matches
Partition allows learned routes tolake precedence over statically configured
backup routes.
Make surethatbackup routes in later partitions are not more specific than
learned hosted DNs
Routes in later partition areconsidered only after learned entry iscompletely
deleted
• The learned IP path istried until CCD Learned Pattern IPReachable
Durationexpiration (default is 60 seconds)
IftheIPpathdoesnot work during this time, thecall fails.
• ToDID is usedas backup after expiration ofCCD Learned Pattern IP
Reachable Duration until expiration ofCCD PSTN Failover Duration (default
is 48 hours)
IfnoToDID is configured, the callfailsduring this time.
• The learned pattern iscompletely removed only after expiration ofCCD
PSTN Failover Duration.
Static backup patterns are now considered.
All routes that are learned bv CCD are put into the same configurable partition. Ifthis partition
is listed first in the CSS ofthe calling phones, ithas higher priority for equally qualified
matches than partitions that arelisted later.
Such aconfiguration allows learned routes to take precedence over statically configured backup
routes You have lo make sure that backup routes in later partitions are not more specific than
learned routes, because the order ofpartitions is relevant only ifthe matches are equally
qualified.
Be aware that routes in later partitions arc considered only after learned routes are removed
from the call-routing table.
When Cisco Unified Communications Manager loses IP connectivity to its SAF forwarder, it
wails for 60 seconds until it considers the IP path to be unavailable. You can configure this
time by using the CCD learned Pattern 11' Reachable Duration CCD feature parameter. During
that time, and calls to learned patterns fail.
Once the timer has expired. Cisco Unified Communications Manager starts another timer, the
CCD PSTN Failover Duration. The default value for this timer is 48 hours. During this time
Cisco Unified Communications Manager tries to place aCCD PSTN backup call. 11 no loDID
has been advertised. Cisco Unified Communications Manager assumes that there is no ISIN
backup path and that therefore calls will fail.
The learned route is purged only after the expiration of the timer. Then another (statically
configured backup) pattern, which is in apartition that is listed after the CO) partition, can be
matched If vol, want to use locally configured static backup patterns, etther disable t CO
PSTN backup b> setting the CCD PSTN Failover Duration timer to 0. or set the timer to a
lower value than the default {twodays).
Call Control Discovery
© 2010 Cisco Systems, Inc
When auser dials 89491000 while the gateway is in SRST mode, the IP path is marked
unreachable due to the loss of IP connectivity. Therefore, the ToDID rule is applied: it strips
the first four digits and then adds the prefix +1949222 so that the number that is used for the
PSTN backup call is +19492221000.
The only static configuration that is required at the Cisco Unified SRST gateway is an
outbound dial peer that routes calls starting with + toward the PSTN.
Hfc
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12010 Cisco Systems, Inc.
SRST Considerations
This subtopic describes how CCD can be implemented at Cisco Unified SRST gateways.
SRST Considerations
SRST subscribes to the CCD
service but does not publish
any patterns.
During WAN failures, SRST
uses learned patterns to
New Yort( SRST Routing Table
DN Pattern TaDiDRule
.
840BXXXX **14S855&' ;10©V
B416XXXX 4:»14lB777- fC®'1
transparently reroute calls over
Protocol
BlP
SIP
SIP
the PSTN.
ACisco Unified SRST gateway does not need to advertise any internal directory numbers
because the SRST site is reachable only via the PSTN. It is the responsibility ofCisco Unified
Communications Manager to know how to route calls lo cluster-internal directorv' numbers
when they arenot reachable overthe IP WAN.
However the Cisco Unified SRST gateway needs a local dial plan that allows end users to
place calls to other sites b\ dialing the internal directory number ofthe other site. Cisco linified
SRS r then must transform the internally used directory numbers to the corresponding PS1N
numbers so that the call can be rerouted over the PSTN. This local dial plan does not have to be
configured manually when CCD is used. Instead, Cisco Unified SRST can subscribe to SAf
and hence learn all internallv known DN ranges and the corresponding ToDID rules. Cisco
Unified SRST learns these routes while there is no network problem. At this time, the learned
patterns are not utilized because the Cisco Unified SRST gateway does not route any calls:
Cisco Unified Communications Manager is in control ofall IP phones and performs callrouting sen, ices.
Once IP connectivity is broken. IP phones fall back to the Cisco Unified SRST gateway, and
once roistered, the Cisco Unified SRST gateway has to route calls. Because the gateway has
learned all available internallv used directory numbers with the corresponding ToDID rules, it
can now route to the respective PSTN number any calls that are based on the dialed internal
directory number.
In the example, the Cisco Unified SRST gateway learned three patterns while IP connectivity
was working: 8408XXXX with aToDID rule of4:+l408555, 8415XXXX with aToDID rule
of4:+l8415. and 8949XXXX with a ToDID rule of4:+l949222.
Call Control Discovery
)2010 Cisco Systems. Inc.
5-59
CCD PSTN Backup—CSS
This subtopic describes which CSS is used for PSTN backup calls.
CCD PSTN Backup—CSS
When PSTN backup is invoked,
HQ Learned Routes
the number that is built based on
the DN and the ToDID rule is called.
The AAR CSS of the calling phone
is used for PSTN backup call.
Strip 0 digits, prefix +1972555
to directory number pattern
I4XXX) (or PSTN number.
Call Placed to +19725554001 witfi
AAR CSS of Calling Phone
When a learned pattern is marked unreachable anda ToDID has been advertised withthe
pattern,a PSTN backup call is placed. The CSS that is used for this call is the AAR CSS.
Make sure that the AAR CSS is set atall phones, so that PS'fN backup calls for CCD-learned
patterns will work. Also ensure that the number that is composed ofthe directory number
pattern and the ToDID rule isroutable (in other words, that a route pattern that matches the
number exists).
Note
PSTN backup for CCD iscompletely independent from AAR. AAR isused toplace PSTN
backup calls for cluster-internal destinations when the IP path cannot be used because of
insufficient bandwidth as indicated by CAC.
It is only the AAR CSS that is reused for CCD PSTN backup. Otherwise, CCD PSTN backup
does not interact with AAR at all. For example, CCD PSTN backup works even when AAR is
globally disabled by the corresponding Cisco CallManager service parameter.
5-58
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
!> 2010 Cisco Systems, Inc.
Monitoring Learned Routes in Cisco Unified Communications
Manager Express
"Ihis subtopic describes how to monitor learned routes in Cisco linified Communications
Manager Express.
Monitoring Learned Routes in Cisco
Unified Communications Manager Express
sf
BR2#shc
dndb all
Last successful DB update 8
........
Pattern
.010:03:12 16:03:27:838
private Dialplan Partition ********
- 2XXX
Primary Trunk-Bouts(B> ID
: 273 274
Aliaa-Routels) Prefix/Strip-Lan : -.498952121/0
Pattern -
3XXX
Primary Trunk-Route(s) ID
: 270 269
Alias-Route(B) Pretix/Strip-Len : ♦44228822/0
,,,,,... global
Pattern
(E164) Dialplan Partition •*•««*'
- *4420>
Trunk-Route!-} ID
•- 271 272
Pattern - .16505051234
Trunk-Fouta(g)
ID
: 270 269
The figure shows how to display the SAF-leamed routes in Cisco IOS Software. Note that onl_
acall agent can interpret SAF service data; SAF forwarders cannot interpret SAF service data.
Therefore, this command works only on Cisco IOS routers that arc internal SAF clients as well
as SAF forwarder.
The output shows two types ofpatterns: extensions (with aToDID) that were learned from a
de\ ice other than aCisco IOS device, and global patterns, which include a+sign and no
ToDID infonnation (most likely advertised by another Cisco IOS internal client).
) 2010 Cisco Systems, Inc
Call Control Discovery
5-57
Monitoring Learned Routes in Cisco Unified Communications
Manager
This subtopic describes how to monitor learned routes in Cisco Unified Communications
Manager.
Monitoring Learned Routes in Cisco
Unified Communications Manager
«SAF-learned routes are not displayed in route plan reports
and cannot be viewed from Cisco Unified Communications
Manager Administration,
• Cisco Unified RTMT has to be used.
Oump-n
iM-rt-N. ctnuw-i -,
JraSBf„
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a*»
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rium
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nitMi(fflio)
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t>-umm
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01132 tci?.ai
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SAF-leamed routes are not visible by any tool of the Cisco Unified Communications Manager
Administration web page. The only way to view SAF-leamed routes is by using the Cisco
Unified Real-Time Monitoring Tool (RTMT).
The figure shows an example ofSAF-leamed routes that are displayed by Cisco Unified
RTMT. The ToDID rule 0: means that the '-internal" pattern and the pattern that is used for
PSTN backup are the same pattern. This principle usually applies when advertising TEHO
patterns are advertised. The ToDID rules that are empty mean that there is no PSTN backup
path forthe respective learned patterns.
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Implemenling Cisco Unified Communications Manager, Part 2<CIPT2) v8.0
>2010 Cisco Systems, Inc.
CCD Considerations
This topic describes important issues that you need lo consider when implementing CCD and
SAF.
CCD Considerations Overview
•
•
•
•
•
Monitoring SAF-leamed routes
CSS used for PSTN backup calls
SRST implementation with CCD
CCD integration with static routing
Cisco IOS SAF client limitations when advertising +
• TEHO implementation with CCD
• Globalized call routing and trunk types
• SAF in Cisco Unified Communications Manager clusters that
use clustering over the WAN
• Other SAF and CCD considerations
This topic includes these CCD and SAF-related considerations:
•
Monitoring SAF-learncd routes
•
CSS used for PSTN backup calls
•
SRST implementation with CCD
•
CCD integration with static routing
•
Cisco IOS SAF client limitations when advertising +
•
TIJIO implementation with CCD
•
Globalized call routing and trunk types
• SAF in Cisco Unified Communications Manager clusters that use clustering over th. WAN
•
Other SAF and CCD considerations
Call Control Discovery
i2010 Cisco Systems. Inc
5-55
Step 6: Configure VoIP Dial Peer
The figure shows the configuration ofadial peer that refers to SAF-leamed routes in Cisco IOS
Software.
Step 6: Configure VoIP Dial Peer
rout.t «lgrp SAP
• •Evic.-fM.ily lpv»
u tonohoua-ay•c•
. f - i n t . r f . c . F«.tl[
topology £•••
•Kit-at-topology
ult-»rvlca- Cully
Boic. l . t v i g .
,.(
prof 11a trunk-rout*
••••Ion
protc
• lp intarfaca loopbmckl tr.ui.port top pot
iflla da-block 1 allaa-pratlx 1972555
1 typa «t«n»ioQ 4111
P»
icollli
c.llt
da-jarvlca
trunk.tout*
da-block
1
cbumal 1
vrc
•ubacrib*
ci
publish c.ll
Sir ••yjtan
1
ntrol wildc*rdad
roj
1
VoIP dial peer for
outbound calls using SAF.
I
dial-paar ,OK
!IH5 voip
The configuration ofthis dial peer is like the configuration ofadial peer that refers to asession
target ras command when an H.323 gatekeeper isused. The destination pattern .Tstands for
all learned routes. The rest ofthe dial peer configuration is used as atemplate for the outgoing
dial peer that is irsed on outbound SAF calls, and for the incoming dial peer that is used on
inbound SAF calls.
Ifyou have other dial peers that also represent learned routes, the preference command will be
used to detemiine which dial peer should be treated with higher priority.
5-54
Implementing Cisco Unified Communications Manager. Part 2(CIPT2) vB 0
>2010Cisco Systems, Inc.
Step 5: Configure Requesting Service
"fhe figure shows the configuration of the advertising service in Cisco IOS Sotlware.
Step 5: Configure Requesting Service
router eigrp SAF
Bervice-family ipv4 autonomous-system 1
si-interface PastEtherriStO/0
topology base
exit-sf-topology
exit-service-faaily
profile trunk-route 1
Bession protocol sip interface loopbackl transport tcp port 5060
profile dn-block 1 alias-prefix 1972555
pattern 1 type extension 4XXX
I
profile callcontrol 1
trunk-route 1
dn-block
1
SAF client is enabled for
channel 1 vrouter SAF asysten 1
subscribe callcontrol wildcards
publish callcontrol 1
learning (subscribe).
J
You can also configure the requesting service under channel tag vrouter EiGRP-ID asystcm
-IS Use the subscribe callcontrol wildcarded command to enable the learning ofroutes that
are advertised by the SAF process that matches the autonomous system number that is specified
at the channel configuration level.
Call Control Discovery
i2010 Cisco Systems. Inc
5-53
Step 4: Configure Advertising Service
The figure shows the configuration ofthe advertising service in Cisco IOS Software.
Step 4: Configure Advertising Service
router
servi
s£-i
eigrp SAP
-e-family iPv4
autonomous
ayst
terface FaetEthernetO/0
topo ogy base
exit
exit-
of-topology
ervice-family
ervice saf
pro fi e
trunk-route
1
ip interface loopbackl transport tcp port 5060
rofila dn-block 1 olias-prnfin 1972555
pattern 1 type extension 4XXX
rofile callcontrol 1
dn-service
trunk-route 1
dn-block 1
channel 1 vrouter SAF asyste
publish callcontrol 1
Actual SAF client, which you can enable
foradvertising (shown here)and learning
(shownin nextfigure). Advertising service
(publish) refers to call control profile.
Refers to EIGRP process and
autonomous system number.
You configure the advertising and requesting services under channel tag vrouter E/GRP-ID
asystem AS. "fhe EIGRP-ID argument refers to the name that was assigned to the router EIGRP
process (SAF. in the example shown). The AS argument is the autonomous system number that
wasassigned to the EIGRP service family.
To enable the advertising service itself, you use the command publish callcontrol tag The tag
argument refers to the tag that was applied to the previously configured call control profile
Effectively, you configure the call control profile (which determines which directory numbers
should be advertised by which trunk protocol) by the SAF process that is identified by the
autonomous svstem number.
5-52
Implementing Cisco Unifier) Communications Manager, Part 2(CIPT2) vS.O
© 20)0 CiscoSystems, Inc.
Step 3: Configure Call Control Profile
The figure shows the configuration ofthe call control profile in Cisco IPS Software
Step 3: Configure Call Control Profile
router eigrp SAF
service-family ipv4 autonomous-system 1
sf-interface FastEthernet0/0
topology base
exit-sf-topology
exit-service-family
voice service saf
profile trunk-route 1
session protocol sip interface loopbackl transport top port 5060
1
profile dn-block 1 alias-prefix 1972555
pattern 1 type extension 4XXX
profile callcontrol 1
dn-service
The profile refers to
directory number blocks
and trunk.
trunk-route 1
dn-block
1
fhe call control profile refers to one or more directory number blocks and to atrunk. 1he call
control profile will be used in the next step to specify that the listed directory number blocks
should be advertised at the specified trunk or trunks (if two ofthem are used). Another
command that xou can enter under dn-smice is site-code site-code extension-length length. It
allows asite code to be prefixed to all configured extensions referenced by the call control
profile The extension-length argument sets the number ofdigits (starting with the least
significant digit) that should be preserved from the configured extension before the site code is
added.
Call Control Discovery
© ;010 Cisco Systems. Inc.
5-51
Step 2: Configure Directory Number Blocks
The figure shows the configuration ofdirectory number blocks in Cisco IOS Software.
Step 2: Configure Directory Number Blocks
router eigrp SAF
service-family ipv4 autonomous-system 1
1
sf-lnterface FastEthernet0/0
topology base
exit-sf-topology
exit-service-family
1
voice service saf
profile trunk-route 1
session protocol sip interface loopbackl transport top port 5060
profile dn-block 1 alias-prefix 1972555
pattern 1 type extension 4XXX
Internal Directory Number and
Prefix for External PSTN Number
Each directory number block is configured globally with aToDID that is applied to all
extensions that are listed later. The command to configure adirectory number block is profile
dn-block tag alias ToDID-prefix strip ToDID-strip. The subsequent command toadd
extensions is pattern lagtype extension pattern.
Note
The ToDID-strip argument stands for the number of digits to be stripped; the ToDID-prefix
argument stands for the prefix tobeadded tothe internal number after stripping digits.
Neither the ToDID-prefix argument nor the pattern argument support the use of the +sign. If
you want to advertise anumber with a+sign, you have to use the command pattern tag type
global pattern. Again, you cannot enter the +sign in the pattern argument; however, due to the
type global, a+sign is prefixed to the configured pattern. The ToDID ofglobal patterns is
always unset.
5-50
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
i* 2010 Cisco Systems, Inc.
Step 1: Configure Trunk Profile
The figure shows the configuration ofatrunk profile in Cisco IOS Software.
Step 1: Configure Trunk Profile
router eigrp SAF
service-family ipv4 autonomous-system 1
1
sf-interface FastEthernetO/O
topology base
exit-sf-topology
exit-service-family
voice service saf
profile trunk-route 1
session protocol sip interface loopbackl transport tcp
port 5060
Interface will be used for call
setup (signaling) of SAFcalls.
The trunk profile is con figured with the interface that should be used for call signaling. It is
configured also with the protocol type (in this case. SIP) and the transport parameters (I CP
versus UDP. and port number).
You can configure one SIP trunk or one 11.323 trunk.
Call Control Discovery
© 2310 Cisco Systems, Inc.
5-49
Internal SAF Client Configuration Procedure
This subtopicshowsthe configuration procedure of an internal SAF client.
Internal SAF Client Configuration
Procedure
1
Configure trunk profile.
2
Configure directory number blocks to be advertised.
3
Configure call control profile.
4
Configure advertising service.
5
Configure requesting service.
6
Configure VoIP dial peer referring to SAF.
The configuration steps that are listed in the figure are steps thai you can do multiple times, if
multiple SAF forwarder processes are configured in separate autonomous systems. Each SAF
client channel that is configured with the advertising and the requesting service has to refer to
anotherSAF autonomous system.
The CCD advertising service ofasingle SAF client channel can refer to multiple call control
profiles. This capability allows the configuration oftwo trunk profiles (one SIP and one H.323
trunk per call control profile). Only one trunk isrequired.
5-48
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
S2010 Cisco Systems. Inc.
CCD PSTN Failover Duration: This parameter specifies the number ofminutes that calls
that are placed to learned patterns that have been marked unreachable are routed through
PSTN faikner and are then purged from the system. For the duration that isspecified in
this parameter to start counting down, another service parameter, CCD Learned Pattern IP
Reachable Duration must first have expired. Theexpiration of thatparameter indicates that
IP connectivih is down between the SAF forwarder andCisco Unified Communications
Manager, and that all learned patterns are marked unreachable. Then, when the duration in
this parameter. CCD PSI'N Failover Duration, expires, all learned patterns are purged from
the system. Also, calls to purged patterns are rejected (the caller hears a reorder tone or a
"This number is unavailable" announcement). Setting this parameter lo 0 means that PSTN
failover is disabled. If the SAF forwarder cannot be reached for the number of seconds
defined inthe CCD Learned Pattern IP Reachable Duration service parameter, and no
failo\er options are provided through the PSTN, then calls to learned patterns will
immediateK fail. Setting this parameter to525600 means that PSTN failover will never
expire and.'as aresult, learned patterns will never be purged due to loss of communication
with the SAF forwarder, "fhe default is 2880 minutes (48 hours).
Issue Alarm for Duplicate Learned Patterns: This parameter determines whether Cisco
Unified Communications Manager issues an alarm called DuplicateLearnedPattern when it
learns duplicate patterns from different remote call control entities on the SAF network.
The default value is False.
(CD Stop Routing On Unallocated I nassigncd Number: ihis parameter determine*
whether Cisco Unified Communications Manager continues to route calls to Ihe next
learned call control entitv (ifadvertised by multiple call agents) when the current cail
control entitv rejects the call with the cause code for Unallocated/Unassigned Number. An
unallocated number represents a hosted directory number that does not exist in the current
call control entit\ . The default value is True.
© 21)10Cisco Systems, Inc
Call Control Discovery
5-47
Step 9: Configure CCD Feature Parameters
The figure shows the configuration ofCCD feature parameters in Cisco Unified
Communications Manager.
Step 9; Configure CCD Feature Parameters
fli" ^s*«a
® **• ****!
••n-ajva.
l»H.*S£™ '.!< If.LfcUc ,m^ tsffitra"
|FW>
~3
~3
F"k*
T,"«
Here are the configurable CCD feature parameters:
• CCD Maximum Numbers ofLearned Patterns: This parameter specifies the number of
patterns that this Cisco Unified Communications Manager cluster can learn from the SAF
network. The higher the number ofallowed learned patterns, the more memory and CPU
processing power that is required. Balance the need for the number oflearned patterns in
your system with the resources ofyour deployment hardware components to guide you in
setting the value in this parameter. When Cisco Unified Communications Manager attempts
to leam more patterns than are allowed by the value that is set in this parameter, the alarm
CCDLeamedPatternLimitReached is issued. The default value is20000.
• CCD Learned Pattern IP Reachable Duration: This parameter specifies the number of
seconds that learned patterns stay active (IP reachable) before Cisco Unified
Communications Manager marks those patterns as unreachable. PSTN failover occurs
when Cisco Unified Communications Manager cannot communicate with the SAF
forwarder because of IP connectivity issues for the duration that is specified in this
parameter. For example, this parameter is set to20seconds. When Cisco Unified
Communications Manager cannot communicate with the SAF forwarder after more than 20
seconds, all calls to learned patterns will failover to the PSTN according to the learned
ToDID rule. PS TN failover continues until IP connectivity to the SAF forwarder is
restored. Cisco Unified Communications Manager automatically detects the restored
connects ity to the SAF forwarder. Cisco Unified Communications Manager then falls back
to the IP path of routes as soon as the routes have been received with the appropriate
reachability infonnation again. When the time that is specified by this parameter has
elapsed. Cisco Unified Communications Manager marks the learned patterns as
unreachable. Ifenabled, the CCD PSTN Failover Duration service parameter timer starts
which allows patterns that have been marked as unreachable through IP to instead be
reached through PSTN failover. The default value is60seconds
5-46
Implementing Cisco Unified Communications Manager, Part 2{CIPT2) v8.0
©2010 Cisco Systems, Inc.
Step 8: Configure CCD Blocked Learned Patterns
The figure shows the configuration of the CCD blocked learned patterns in Cisco Unified
Communications Manager.
Step 8; Configure CCD Blocked Learned
Patterns
Blocked wanted Pattern configuration
-Purge»""> ttock S*FCCD Learned Route* Intormat
Le»rnesJ Pattern
Lesrres Ssttetn Prefix
Remote Call Control Identity
Remote IP
Patterns can be blocked based
on remote call control ID, remote
IP. and learned pattern or
learned prefix (the last two are
exclusive!
Multiple blockingrules can be
configured. Ifa learned pattern
matches any rule, it is blocked
CCD blocked learned patterns are optional. If CCD blocked learned patterns are configured, all
routes that match any of the configured criteria arc blocked. As aresult, they are not added to
the call-routing table.
You can configure afilter that is applied to received routes in order to deny the learning of
routes, using these criteria:
• [earned pattern: The received pattern is checked in its entire length. If it matches ihe
configured learned pattern, it will not be added to the local call-routing table.
• 1earned pattern prefix: The received patterns arc compared with the configured prefix,
sorting with the left-most digit. By using alearned pattern prefix for blocking received
routes you can filter internally used numbers by their leading digits-tor example, by their
site code.
. Remote call control identity: Each call agent has aso-called SAF client ID By setting the
remote call control identity, you can filter received routes that are based on the ID of the
ad\ertising call agent.
• Remote IP: B> setting this filler, you can block routes that arc based on the advertising IP
address.
Call Control Discovery
© £010 Cisco Systems, Inc
5-45
Step 7: Configure CCD Requesting Service and Partition
The figure shows the configuration ofthe CCD requesting service in Cisco Unified
Communications Manager.
Step 7: Configure CCD Requesting
Service and Partition
Alllearned routes are put into
the specified partition.
Make sure that all devices
thai should be able to reach
SAF-leamed destinations
have this partition included in
their CSS,
You can configure only one CCD requesting service. You have to enter the partition that all
learned routes should be put into. You must first create the partition as shown in the figure.
In addition to creating the partition, you can configure the CCD requesting service with a
learned pattern prefix and aPSTN prefix. These prefixes are applied to all learned DN patterns
and to all learned ToDID rules, respectively.
Finally, the CCD that isrequesting service isreferred tothe SAF-enabled SIP or to the SAFenabled H.323 trunk.
Ifyou associate the CCD requesting service with only one type oftrunk, all received routes that
are reachable by the other (unconfigured) protocol type are ignored. They are not added to the
call-routing table.
5-44
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
;>2010 Cisco Systems, Inc.
Step 6: Configure CCD Advertising Service
The figure shows the configuration of the CCV> advertising service in Cisco Unified
Communications Manager.
Step 6: Configure CCD Advertising Service
^Sr» X1™" O0™ <£}*•• 'V"""*"
The advertising service
> uri.fe-l.i)*) <.™ii« Inf. -
SVHIH*™-..
,
refers to trunks and
one hosted DN group
p
You need to configure one CCD advertising service for each eonligured hosted DN group. ^
bach CCD advertising sen ice can use the SAF-enabled SIP trunk or the SAF-enabled H.323
trunk. One trunk has to be specified. Multiple CCD advertising (and the CCD requesting
sen ice)can refer to the same SAF-enablcd trunks.
Call Control Discovery
© 2010 Cisco Systems, Inc
Step 5: Configure Hosted DN Pattern
The figure shows the configuration ofahosted DN pattern in Cisco Unified Communications
Manager.
Step 5: Configure Hosted DN Pattern
Is-* X**" D0" sQ>*mmn
®'- Hotted ON PMttra rrrfo •
Ho««fl »*t»re '
DeiciDbon
Hosted im Group"
If no ToDID rules are
configured, ToDID rules of
hosted DN group are
If set here or at hosted DN
group, configured hosted
partem is also ToDID
applied.
number.
Hosted DN patterns refer to ahosted DN group. As mentioned earlier, ifthe parameters at the
hosted DN pattern are unset, the parameters ofthe hosted DN group are applied. When the
PSTN Failover Strip Digits field is set to 0and the PSTN Failover Prepend Digits field is
empty, both fields are considered tobe unset. The configuration example that isshown in the
figure does not generate aToDID rule of 0:, but it applies the settings of the configured hosted
DN group.
At the hosted DN group, the same logic applies. Ifthe PSTN Failover Strip Digits field is set to
0and the PSTN Failover Prepend Digits field is empty at the hosted DN pattern and at the
hosted DN group, then the no ToDID rule is advertised. As aresult, there is no PSTN backup
when the IP path is unavailable.
If you want to advertise aToDID rule of 0:, the number that should be used for backup is
identical to the internally used number (for example, when TEHO patterns are advertised)
Therefore, you have to check the Use Hosted DN as PSTN Failover check box.
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Implementing Cisco Unified Communications Manager. Part 2(CIPT2) v8.0
>2010 Cisco Systems. Inc.
Step 4: Configure Hosted DN Group
The figure shows the configuration ofahosted DN group in Cisco Unified Communications
Manager.
Step 4; Configure Hosted
J^s.™ X0** D0"" &***"•
®The ToDID rules used for
PSTN backup are applied to
- ltDflcd DN Sr
hosted DN patterns only if not
Tconfigured at DNpattern
05™ ItJuvtK SOB DIOJIS
0
'.iUil HJiMOO" «! PSTN F*ltft>
if set at hosted DN pattern
configuration,hosted DN
group configurationis
ignored
The hosted DN group will be referenced from hosted DN patterns. Ifall—or at least most—of
the associated hosted DN patterns share the same ToDID rules, you can configure the ToDID
rule at the hosted DN group. The settings of the hosted DN group arc applied to the hosted DN
patterns if the hosted DN pattern parameters areunset.
The Use Hosted DN as PS'fN Failover check box instructs Cisco Unified Communications
Manager to create aToDID rule of 0:. As aresult, the number that is to be used for PSTN
backup is identical to the internally used number. Usually, this result occurs only when tail-end
hop-off (TF. HO) patterns are advertised.
Call Control Discovery
) 2010 Cisco Systems, Inc.
5-41
Step 3: Configure SAF Trunk
The figure shows how to add aSAF trunk in Cisco Unified Communications Manager.
Step 3: Configure SAF Trunk
SAF SIP trunk is shown. SAF
^^^^^^^^^j
H.323 trunk is also supported.
-- -"- - '
in
S!>T.ur*
I
[*™ tow'
You can configure one SAF-enabled SIP trunk (as shown in the figure) and one SAF-enabled
H.323 trunk. With aSAF-enabled H.323 trunk, you have to first add astandard nongatekeepercontrolled ICT and then check the Fnable SAF check box. Once the check box ischecked, the
IP address field is disabled. The reason is that the configured trunk does not refer to aparticular
destination IP address but instead acts as a template for adynamically created trunk once a SAF
call isplaced. The destination IPaddress isthen taken from the learned SAF service data.
The same concept applies to the SAF-enabled SIP trunk. The only difference is that the SAFenabled SIP trunk is a special trunk service type, which isselected before the trunk
configuration page isshown. Therefore, there isno extra check box like there is atthe
nongatekeeper-controlled ICT. The SAF-enabled SIP trunk also does not have adestination IP
address field.
You can have one SAF-enabled H.323 oroneSAF-enabled SIPtrunk.
5-40
Implementing Cisco Unified Communications Manager, Part 2{CIPT2) v8.0
©2010 Cisco Systems, Inc.
Step 2: Configure SAF Forwarder
The figure shows how to add aSAF forwarder to Cisco Unified Communications Manager.
Step 2; Configure SAF Forwarder
Client label must match SAF
client ID configured at SAF
forwarder.
IPAddress of SAF Forwarder.
i
The destination IP address has to match the one ofthe interface that is specified with the sfinterfaee command at the SAF forwarder.
If you want to register with more than one SAF forwarder, click the Show Advanced link. This
link allows you to configure multiple SAF forwarders and to associate individual members oi
the cluster selectively with the configured SAF forwarders.
Note
If you want to allow multiple nodes of aCisco Unified Communications Manager clusier to
act asSAF clients, each of them needs a unique client name. You can either configt re each
of them individually with separate node names or use aSAF client ID in Cisco Unified
Communications Manager, which is ctient-ID@ The @sign instructs Cisco Unified
Communications Manager to add a unique node number sothat the actual client IDs are
client-ID@1, client-ID@2, and so on.
At the SAF forwarder, you can either create individual entries or add the keyword basename
to the external-client client-ID command. Do not specify the @sign at the SAF forwarder:
only add the keyword basename to the external-client command, and the specifieo client
ID will bepermitted with any suffixes of @followed by a number.
Call Control Discovery
©2010 Cisco Systems, Inc
5-39
Step 1: Configure SAF Security Profile
The figure showshow to configure a SAF securityprofile.
Step 1: Configure SAF Security Profile
©«
This must match the usemame
and password configured at the
(trqitiw.
SAF forwarder.
Make sure that the username and the password match the username and password that were
configured at the SAF forwarder.
5-38
implementing Cisco Unified Communications Manager, Pari 2(CIPT2) v8.0
>2010 Cisco Systems, Inc.
External SAF Client Configuration Procedure
This subtopic shows the configuration procedure ofan external SAF client.
External SAF Client Configuration
Procedure
1 Configure SAF security profile.
Configure SAF forwarder.
2
:>
4
5
6
7
Configure SAF trunk.
Configure hosted DN group.
Configure hosted DN pattern.
ConfigureCCD advertising service.
Configure CCD requesting service and partition.
8 Configure CCD blocked learned patterns (optional).
\i Configure CCD feature parameters (optional).
"fhe last two configuration steps are optional. You do not have to configure the CCD
advertising sen ice and the CCD requesting service ifyou want only lo advertise or leam call
routes (exclusively).
Note
All stepsare performed at Cisco Unified Communications Manager.
Call Control Discovery
© 2010 Cisco Systems. Inc
5-37
Step 2: Configure SAF Forwarder to Support External SAF
Clients
The figure shows how lo configure a SAF forwarder to support an external SAF client.
Step 2: Configure SAF Forwarder to
Support External SAF Clients
router aigrp SAF
service-family Lpvi outoac
1
This must match the SAF client
sf-interface FastEthernet
IID configured at theexternal
topology base
e»it-Bf-topology
SAF client.
external-oliont HQSAP
1
service-family external-eli at listen ipv4 5050
external-client HQSAP
UBername SAFUSEK
password SAFPASSWORD
IUsername and password must match
the settings at the external SAF client.
External client ID refers to the
appropriate SAF process and
autonomous system above.
Fach allowed exiemal client has to be listed in the service-family section. In addition, the
username and password that should be used by the external client have to be specified in the
service-family external client section.
Note
If you want to allow multiple nodes of a Cisco Unified Communications Manager cluster to
act asSAF clients, each of them needs aunique client name. You can either configure each
ofthem individually with separate node names oruse a SAF client ID in Cisco Unified
Communications Manager, which isclient-ID@. The @sign instructs Cisco Unified
Communications Manager to add a unique node number sothat the actual client IDs are
client-ID®-}, client-ID@2, and so on.
At the SAF forwarder, you can either create individual entries or add the keyword basename
to the external-client client-ID command. Do not specify the @sign atthe SAF forwarder;
only add the keyword basename to the external-client command, and the specified client
ID wjjl bepermitted with any suffixes of@followed by a number.
5-36
Implementing Cisco Unifiea Communications Manager, Part 2(CIPT2) v8.0
32010 Cisco Systems, Inc.
Step 1: Configure SAF Forwarder
he figure shows the SAF forwarder configuration.
Step 1: Configure SAF Forwarder
EIGRP process \0 has only
local significance.
router eigrp SAF
service-family ipv4 autonomous-systew^^
sf-interface FastEthernetO/0
topology base
exit-sf-topology
exit-service-family
Autonomous system must be
the same on all SAF forwarders
that should exchange data
"~l
IP of specified interface is used
, by SAFforwarder:
In the example, aSAF forwarder is configured with autonomous system I. All SAF forwarders
that should exchange information with each other have to be in the same autonomous system.
You use the sf-interface command to bind the SAF process to the specified interface. Ifthe
router has multiple interfaces, if is recommended that you use aloopback interface.
Call Control Discovery
>2010 Cisco Systems. Inc
5-35
SAF Forwarder Configuration Procedure
This subtopic shows the configuration procedure of the SAF forwarder.
SAF Forwarder Configuration Procedure
1
Configure SAF forwarder,
2 Configure SAF forwarder tosupport external SAF client (if
used).
The configuration ofthe SAF forwarder consists oftwo steps: the SAF forwarder configuration
(mandatory) and the support ofan external SAF client (if used).
5-34
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems, Inc.
Relationships of Internal SAF Client Configuration Elements
This subtopic describes the relationships ofinternal SAF client configuration elements.
Relationships of Internal SAF Client
Configuration Elements
SAF Client
SAF CH«« "Charmer
DN Block
DN Block
Profile
Profile
CCD Recjueslmg
Service
Call Control Profile
CCD Advertising
Service
Trunk Profile
The figure illustrates how internal SAF client configuration elements relate to each other.
Note
The configuration that is shown in the figure is one that you can do multiple times, if multiple
SAF forwarder processes are configured in separate autonomous systems. Each SAF client
"channel" has to refer to another SAF autonomous system
The CCD advertising service of a single SAF client channel can refer to multiple call control
profiles This capability allows the configuration of two trunk profiles (one SIP and one H.323
trunk per call control profile). Only one trunk isrequired.
Call Control Discovery
© 2010 Cisco Systems, Inc
Internal SAF Client Configuration Elements
This subtopic describes the configuration elements of an internal SAF client.
Internal SAF Client Configuration Elements
Configuration
Configuration Element Function
Element Name
Trunk profile
DN block profile
Call control profile
SAF client -channel'
profiletrunk-route: Configured withinterfacewhose IP
addressshould be usedforsignaling when setting upSAF calls.
profile dn-block: Configured withpatterns to be advertised
(internal number and number used for PSTN backup).
profilecallcontrol: Refersto DN block profiles and trunk
profile.
channel: Configuredwith SAF client IDand autonomous
system. Advertising and requesting services areenabled; the
advertising service refersto the callcontrol profile.
0(81 f>e&
dial-peer voice:Configured with destination-pattern .Tand
session target saf. This isthe incoming and outgoing dial peer
for a caHsent to or received from SAF trunks.
The table shows the configuration elements ofan internal SAF client, their functions, and the
ways that they interact with each other.
Note
You can configure the advertising service and the requesting service independently of each
other.
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Implementing Cisco Unified Communicalions Manager, Part 2(CIPT2) v8.0
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Relationship of External SAF Client Configuration Elements
This subtopic describes the relationship ofexternal SAF client configuration elements.
Relationship of External SAF Client
Configuration Elements
The figure illustrates how external SAF client configuration elements relate to each other.
Note
You can configure only a single CCD requesting service in your cluster. You can configure
Itiple blocked learned patterns, SAF forwarders, and SAF security profiles. You a n
nfigure one CCD SIP trunk and one CCD H.323 trunk. Only one trunk is required
mu
co
Call Control Discovery
© 2010 Cisco Systems. Inc
5-31
External SAF Client Configuration Elements
This subtopic describes the confiiuralion elements ofan external SAF client.
External SAF Client Configuration Elements
SAF security profile
SAF forwarder
Catena *musernameBiWpasBV«n).Reienmc«J(tomSAFtOnWidar.
Points to a
AFforwarder Configured with IP addressof SAF forwarder Refers
to SAF sec
nty profile
trunk andone SAF H323 twr* canbeconfigured. Ttiey arenot
One SAF
SAFtrur*s
configure
configured' Hti
ti a destination IPaddress.Therest ofthe confaurafjon isstrraiar
to normals 3andH323r/uni».
Nested DN group
Configured nthi PSTN laitover stripdigrtsand
1 DN patterns
Hosted DN pattern
PSTN
Directory
hosted DN
PSTN failover prepend diaits
ot directory number raneetoBeadvetttned. Configured wth
pdiojaj andPSTN faliover prepaid aigHs; ifnotonofigwad
i •wipeonriguratlonriiH«).Appli8dtDhosted DNB'OUO
CCD advertising service
Refersto hoptedDNgroup, SAFSIPtrunk, and SAFH.323trunk
CCD requesting service
Configured<iitti route partition, teamed pattern prefix, and PSTN prefix. Refers to
Blocked learned patterns
Configurediith remote IP. remote call control identity, and learned pattern or
learned pre! ;
The table showsthe configuration elements of anexternal SAF client, their functions, and the
ways that they interact with each o :her.
Note
You can configure the C<
< ;d advertising service and theCCD requesting service
independently of each ot ler.
Theconfiguration ofbloc ;ed learned patterns is optional.
«*
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High-Level Configuration Overview (Cont)
Implement external SAF client:
• Add external SAF client (Cisco Unified Communications Manager) to SAF
forwarder:
Specify SAF ID, username, and password of external client.
Map external SAF client toSAF autonomous system.
• Add SAF forwarder (Cisco IOS router) toSAF client (Cisco Unified
Communications Manager).
- Specify SAF ID. username, and password asconfigured at SAF
forwarder.
• Configure CCD at external SAF client.
Configure SAF trunks
- Configure hosted DN patterns andhosted DN groups.
ConfigureCCD advertisingservice.
Configure CCD requesting service and partition for learned patterns.
When implementing external SAF clients, you must perfonn these high-level configuration
tasks:
Stepl
At the SAF forwarder (Cisco IOS router), add the external SAF client (Cisco
Unified Communications Manager):
• Specif\ the SAF ID. username. and password ofthe external client.
•
Step 2
Map the external SAF client to the SAF autonomous system.
At the external SAF client (Cisco Unified Communications Manager), add the SAF
forwarder (Cisco IOS router):
• Specify the SAF ID. username. and password as configured at the SAI
forwarder.
Step 3
Contigurc CCD atthe external SAF client:
• Configure a SAF SIP ora SAF H.323 trunk.
• Configure the hosted DN patterns and hosted DN groups.
•
Contigurc the CCD advertising service.
• Configure the CCD requesting service and the partition to be used for learned
patterns.
Call Control Discovery
i 2010 Cisco Systems, Inc
5-29
SAF and CCD Implementation
This topic describes how to impl ment SAF in Cisco IOS routers. It alsodescribes howto
implement CCD in Cisco Unifiel Communications Manager and
as Cisco Unified Communicatior s Manager Express.
in Cisco IOS call agents such
High-Level Configuration Overview
Configure SAF networf
• Configure SAF forwarde s on Cisco IOS routers.
- Specify same autono nous system on all SAF forwarders,
Implement internal SAI client:
• Configure trunk profilew:h IP to be used for call setup,
• Configuredirectorynuml er blocks to be advertised,
Configure call control profile referring to directorynumber blocks
and trunk to be used.
Configureactual CCDrpn »cess (channel) referring to SAF forwarder
by autonomous system n jmber.
- Enable advertising servicebyreferring to callcontrol profile,
- Enable requesting se
*»
ConfigureVoIP dial peer eferring to SAF.
The first main configuration task : ;to enable SAF inthe network. You have toconfigure SAF
forwarder functionality on a Cisoc IOS router. All SAF forwarders must share the same SAF
autonomous system number. You
:an specify the interface thatshould beused bytheSAF
forwarder.
Whenusing internal SAF clients, •ou mustperform these main configuration tasks:
Step 1
Configure a trunk prof e and specify the interface and theprotocol to beused for
call signaling.
Note
Step 2
Step 3
The IP address (interfaci) that is used for call signaling can be different from the IP address
that is used by theSAF forwarder.
Configure the directoryjnumber blocks to be advertised.
Configure acall controljprofile that refers to the directory number blocks and the
trunk profile to be used.
Step 4
Configure the actual cqD process ("channel") that refers to the SAF forwarder by
its autonomous system dumber. Then perfonn these tasks:
• Fnable the CCD advertising service by referring to the call control profile.
•
Step 5
5-28
Fnable the CCD requesting service.
Configure a VoIP dial pier that refers to the SAF.
Implementing Cisco Unified Communications Mans per, Part 2(CIPT2) v8.0
12010 Cisco Systems, Inc.
Note
if the leamed pattern was removed when the IP path became unavailable, the originating
site would not know what PSTN number to use for the backup call. By default, a route is
completely removed only if it has not advertised for 48hours.
Call Control Discovery
© 2010 Cisco Syslems. Inc
5-27
CCD—Call from HQ to BR During Link Failure
This subtopic describes the call $ow for acall from the HQ site to the BR site during a link
failure at the BR site.
CCD—Call frorri HQ to BR during Link
Failure
BR Learned Routes
XKtX
-
PSTtt
•
"
Q>.«a953tS1
101.5.10
Cal1 Placed lo +19725554001
UsingPSTN
When a user at the HQ site dials I00I during the link failure at the BRsite, the called numberis
still found in the call-routing tabl of the HQ Cisco Unified Communications Manager cluster.
However, the IP path is marked i ireachable, and therefore the call cannot besetup over the IP
network with the use of a SIP tru ik
Cisco Unified Communications M anager now checks whether there is a ToDID rule that is
associated with the leamed patte i. In this case, a ToDID rule of 0:+1972555 has been learned.
Cisco Unified Communications
Manager does not strip any digits {because ofthe 0 in front of
the : [column]), but adds the pref +1972555 to the dialeddirectorynumber 4001. The
resulting number +19725554001 s now matched in the call-routing table of Cisco Unified
Communications Manager, when a match is found in a PSTN routepattern.
Note
The CSS that is used fo • the PSTN
backup call istheAAR CSSofthecalling phone. The
oute group, and gateway for PSTN access has to be in place in
order for PSTN backup o work. Further, the PSTN route pattern has to be in a partition that
route pattern, route list,
is reachable from the AAR
CSS of the calling phone.
In the example, the ToDID rule results in globalized PSTN numbers (E.164 format with a +
prefix) Therefore, a PSTN route pattern that matches this format (for example, \+.!) has to
bein place. If all sites share the same PSTN dial rules-for example, all sites arewithin the
North American Numbering Plan (NANP)—then you could also configure ToDID rules that
result in PSTN patterns with a PSTN accesscode, followed by a national accesscode
followed by the 10-digit PSTN number. In this case, your PSTN route pattern would have to
be91[2-9]XX[2-9]XXXXXX.
The call is now set up over the PSTN.
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© 2010 Cisco Systems, Inc.
CCD—Link Failure at BR
This subtopic describes how CCD manages a link failure between the SAF client and its SAF
forwarder at the BR site.
CCD—Link Failure at
BR learned Routes
HQ Learned Routes
«M
B0«
0«1WI559
r);M0R6il!1
' WI.S.10
SIP
Marked Unreachable
•-
- PSTN-
•49S953i2ixxxx>s:;:::;;
HQ
101510
SAF-Enabled
IP Network
When the connection between the SAF client and the SAF forwarder at the BR site isbroken,
the SAF forwarder atthe BR site detects this problem that is based on the missing keepalives of
the registered SAF client.
The BR SAF forwarder sends an update throughout the SAF-enabled network so that all SAF
forwarders are aware that the IP path to4XXX is currently unavailable. Other than in IP
routing, the learned route is not removed, but only the IP path is marked unreachable.
All SAF forwarders that have registered SAF clients now pass this update on totheir SAF
clients, so that all SAF clients in the network can mark the IP path lo4XXX as unreachable.
As shown in the figure, the call-routing table at the HQ site also gets updated accordingly.
© 2010 Cisco Systems. Inc
Call Control Discovery
CCD—Call from HQ to BR
This subtopic describes the call flow for a call from the HQ site tothe BR site.
CCD—Call from HQ to BR
BR Learned Routes
WWt
»*48WS3«1
W13.10
Call placed to 4001 using SIP trunk to 10.1.7.10.
PSTN;
The HQ site dials 4001. The called number is found in the call-routing table ofIhe IIQ Cist
Unified Communications Managercluster.
Note
All learned routes are put into the same configurable partition The CSS of the calling phone
must have access to this partition in order for the call towork. If the calling phone does not
have access to the partition that includes all learned patterns and there isno match in any
other partition, the call fails.
Cisco Unified Communications Manager identifies the matched pattern as aCCD-learned
pattern. According to the learned route, the call has to besetup through a SAF-enabled SIP
trunk, which is dynamically created tothe destination IP address as leamed by CCD (in this
case. 10.1.7.10).
The call is now set up over the IP network.
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CCD—Propagation of BR Routes
This subtopic describes how BR routes are propagated to the IIQ site.
CCD—Propagation of BR Routes
BR Learned Routes
HQ Learned Routes
W&nmmmWmMmW mmnV^&mmmmmn^R Mf
0-*!B7M65
10.1 T.10
ao«
SIP
«»«0M»iai,
10.iS.1tt
BIP
Learn hosled DN range and ToDID
rule and store in memory
PSTN
Advertise hosted DN range
(4XXX) and ToDID rule
(0+1972555).
M972555XXXX
SAF-Enabled
IP Network
The Cisco Unified Communications Manager cluster atthe BR site advertises its direeto >
number range 4XXX with aToDID rule of0:+1972555 to its SAF forwarder, fhe SAF
network propagates this new route throughout the network, and the SAF forwarder at the HQ
site sends the information to the HQ Cisco Unified Communications Manager cluster. The callrouting table of the HQ cluster is populated with the directory number pattern 4XXX and a
ToDID rule of 0:^1972555.
Again, onh aSIP trunk has been associated with the CCD advertising service at the originating
site. Therefore, the IIQ cluster learns the route only forSll\
The network is in a converged state. All sites know about the routes ofall other sites.
Call Control Discovery
) 2010 Cisco Systems, Inc
5-23
CCD—Propagation of HQ Routes
This subtopic describes how HQ routes are propagated tothe BR site.
CCD—Propagation of HQ Routes
HQ Learned Routes
BR Learned Routes
WW
&MW85S1S1
W.J5.10" " ~W
Learn hosted DNrange
and ToDID rule and store
in memory.
97255SXXXX
Advertise hosted DN
range (2XXX) and ToDID
rule (0:^-498953121
The Cisco Unified Communications Manager cluster at the HQ site advertises its directory
number range 2XXX with a ToDID rule of0:+49895312I to itsSAF forwarder. The SAF
network propagates this new route throughout the network, and the SAF forwarder atthe BR
site sends the information to the BR Cisco Unified Communications Manager cluster. The callrouting table ofthe BR cluster ispopulated with the directory number pattern 2XXX and a
ToDID rule of 0:+498953121.
At the advertising site, only aSIP trunk has been associated with the CCD advertising service.
Therefore, the BR cluster learns the route only for SIP.
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CCD Operation
This topic describes how CCD works for on-net calls and how CCD reroutes calls to the PSTN
if the IPpath is notavailable.
CCD—Base Configuration
HQ Learned Routes
BR Learned Routes
The figure shows the base configuration. There are two sites, each with aCisco Unified
Communications Manager cluster. One site (••HQ" in the figure) is located in Germany, and it
has -i DID ran°e of +4989^3121 XXXX. Intemally, range 2XXX is used. The other site ("BR )
is in the United States: it has a DID range of+ I972555XXXX. Internally, the directory number
range 4XXX is used.
m
Call Control Discovery
12010 Cisco Systems. Inc
All learned routes are put into one configurable partition. All devices that should have access to
leamed routes need that partition to be included in their calling search space (CSS). Sometimes
the IP path tor alearned route may not be available, and aToDID rule may have been
advertised with the hosted directory number. In that situation, acall to the transformed number
(a ToDID rule that is applied to the advertised pattern) is placed with the automated alternate
routing(AAR) CSS of the callingdevice.
Note
PSTN backup for CCD is completely independent from AAR. AAR is used to place PSTN
backup calls for cluster-internal destinations when the IP path cannot beused because of
insufficient bandwidth as indicated by Call Admission Control (CAC).
It is only the AAR CSS that is reused for CCD PSTN backup. Otherwise, CCD PSTN backup
does not interact with AAR atall. For example, CCD PSTN backup works even when AAR is
globally disabled by the corresponding Cisco CallManager service parameter.
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Processing Received Routes in Cisco Unified Communications
Manager
This subtopic describes how received routes are processed in Cisco Unified Communications
Manager,
Processing Received Routes in
Unified Communications Manager
Administrator can block received routes based on;
• Learned pattern prefix
• Learned pattern
• Remote call control identity
* Remote IP
Load balancing occurs for learned routes:
• Round robin between protocols, among local trunks (SIP and
H 323), and learned remote IP addresses
Partitions and CSS:
• All learned patterns are put into oneconfigurable partition.
• All devices that should have access to learned routes need
access to that partition from their CSS
• AAR CSS is used for PSTN backup calls.
You can configure afilter that is applied to received routes in order to deny the learning of
routes, using these criteria:
• Iearned pattern prefix: The received patterns are compared with the configured prefix,
starting with the left-most digit. By using alearned pattern prefix for blocking receiv ed
routes, you can filter intemally used numbers by their leading digits—lor example, by their
site code.
• 1earned pattern: The received pattern is checked in its entire length. If it matches the
configured learned pattern, it will not be added to the local call-routing table.
• Remote call control identity: Each call agent has aso-called SAI-' client ID. By setting the
remote call control identity, you can filter received routes that are based on the ID o. the
advertising call agent,
. Remote IP: By setting this filler, you can block routes that are based on the advertising IP
address.
You can configure one or more criteria when setting up afilter. However, as soon as one
criterion ismatched, the learned route is filtered.
The same destination number can be learned multiple times. It may be advertised by different
call agents. It mav allow SIP and 11.323 to be used for setting up the call (both signa ing
protocol capabilities are advertised separately). It also may be reachable at multiple IP
addresses (of the same call agent, in the case of aCisco Unified Communications Manager
cluster). Ifa route is learned multiple times, Cisco Unified Communications Manager™! loadshare the outbound calls to the corresponding destination among all possible paths (that is. t>>
protocol and remote IP addresses).
Call Control Discovery
S 2010 Cisco Systems. Inc
Note
Like thetrunks that areassociated with CCD advertising services, thetrunks thatare
associated with theCCD thatis requesting services are not used to learn patterns via SIP or
H323. They determine theoutbound capabilities for calls thatare placed tolearned
destinations. If the CCD that isrequesting service isassociated only with an H.323 trunk,
learned routes that areto bereached via SIP are not added to the call-routing table of the
receiving Cisco Unified Communications Manager.
The Cisco Unified Communications Manager nodes of the cluster that ispermitted to place
outbound calls to learned routes are determined by the device pool that isapplied to the
trunk that is associated with the CCD requesting service.
mm
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CCD Services in Cisco Unified Communications Manager
This subtopic describes the two CCD services that exist in acall agent such as Cisco Unified
Communications Manager.
CCD Services in Cisco Unified
Communications Manager
CCD advertising service
* Responsible for advertising:
-- Hosted DN ranges
- PSTN failover information (ToDID rule)
- Trunk information
• Configured with one or two trunks (one SAF CCD SIP trunk and
one SAF-enabled H.323 ICT supported)
CCD requesting service
• Responsible for learning DN ranges from the SAF network
* Exists only once per Cisco Unified Communications Manager
cluster
• Configured with one or two trunks (one SAF CCD SIP trunk and
one SAF-enabled H.323 ICT)
fhe CCD advertising sen ice is configured with the directory' numbers that are to be advertised.
In Cisco Unified Communications Manager, they are configured by so-called hosted DN
(directorv number) ranges. Fach hosted DN range is configured with its PSTN failover
information (the ToDID rule for the hosted DN range). In addition, the signaling protocol and
the IP addresses ofthe call agents have to be advertised. They are configured by a trunk. The
trunk can be a SAF-enabled H.323 intercluster trunk (ICT) ora SAF CCD SIP trunk. CCD
advertises call routes with one ormore call agent IP addresses. 1he IP addresses tobe
advertised are determined bv the device pool that isapplied tothe SAF-enabled trunk.
Note
The trunk is not used to advertise call routes Call routes are advertised by CCD and SAF
and not via H323 orSIP. The trunk is used todetermine the IP addresses ofthe call agents
and the supported signaling protocols, in case another call agent wants to establish a call to
a learned call route
The CCD that is requesting service is responsible for subscribing to call-routing infonnation
from its SAF forwarder. It allows Cisco Unified Communications Manager to leam routes from
the SAF-enabled network. Onlv one CCD that is requesting service exists per Cisco Unified
Communications Manager cluster. However, like the advertising service, it can be configured
to accept patterns that are reachable via SIP or H.323. depending on the associated trunk or
trunks.
Call Control Discovery
)2310 Cisco Systems. Inc
• Number ofdigits to be stripped: The first part ofaToDID rule is the number of digits to
be stripped from the intemally used number. For example, ifsite code dialing is used and
the internally used number to reach ablock ofdirectory numbers is 8-408-2XXX, you may
want to strip the leading 8408 before prefixing the necessary digits to directory number
range 2XXX. In this case, your ToDID rule would start with 4: (because the four leading
digits should bestripped).
• Prefix to be added to the (deflated) internal number: The second part ofaToDID rule is
the prefix that should be added to the intemally used number after digit stripping has been
performed. In the previous example, ifthe PSTN direct inward dialing (DID) range of the
intemally used directory number range 2XXX (dialed as 8-408-2XXX from other sites) is
408 555-2XXX, the prefix would be 408555. Usually E.164 format with a+prefix is used
to represent the PSTN number, so the configured prefix would be+140855.
In the given example, the complete ToDID rule would be 4:+1408555, because the numbers to
be stripped and the prefix are separated by a column.
By advertising only the locally present internal numbers and the corresponding ToDID rule at
each call agent, the dial plan implementation oflarge networks is extremely simplified. Ifthere
are any changes at acall agent, you have to change only the advertised number (range) and its
ToDID mle at the affected call agent. All other call agents will dynamically leam the changes
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CCD Characteristics
Ihis subtopic explains the main characteristics ofCCD.
Call Control Discovery Characteristics
• CCD enables call agents to exchange call-routing information:
•- Dial plan information
Reachability information (IP address ofcall agents)
• Allows dynamic routing based on information learned through SAF
- No need for full-meshedcall-routing configuration
No need for centralized gatekeepers
Only local number ranges that should beadvertised must be
configured
• Propagated dial plan information has two components:
Enterprise-owned, internally used numbers
External (PSTN) numbersfor PSTN backup
• Simplifies dial plan implementation in large networks
CCD enables call agents to exchange call-routing information. The infonnation that is relevant
consists of these components:
• Dial plan information: Dial-plan infonnation includes internally used directory numbers
(potentially with internal prefixes such as site codes), the IP addresses ofthe respective call
agents and the signaling protocol that will be used by the call agents. Ali ofthis
information is advertised by call agents that is propagated throughout the network by SAP.
and then learned by other call agents.
• Reachability information: This dynamic routing for call reachability infonnation
drastically simplifies dial plan implementations in large networks. There is no need lor a
static full-mesh configuration and no need even for the configuration of acentralized callrouting sen-ice (such as an H.323 gatekeeper or aSIP network service). You have tc
configure onlv the internal number range that should be advertised per call agent at the
respective call agent. CCD and SAF then ensure that the locally known numbers are
distributed among all call agents.
When rerouting over the PSTN is desired, call agents are configured not only to ^crtise their
internallv used number ranges, but also with the corresponding PSTN numbers, Hie PS 1N
number is not advertised as adistinct number, but it is advertised by aPMNtailovcr digit
transformation rule that is known as aToDID rule. AToDID rule describes how he mtemally
used number has to be manipulated to get to the associated PS TN number. AIoDID rul.
consists of two components:
Call Control Discovery
© 2010 Cisco Systems, Inc.
5-15
CCD Characteristics
This topic describes the characteristics ofCCD and how CCD forwarding and requesting that
services are used in Cisco Unified Communications Manager.
CCD Overview
SAF clients generate call-routinginformationand send it as SAF
Cisco Unified
Co mmunica lions
service data to their SAF forwarders.
Manager
SAF forwarders propagate information to olher SAF forwarders
only considering SAF header.
SAF forwarders send information to SAF clients, which process
SAF service data (call-routing information).
SAF Service Oata (CCD
information) as Sent lo
and from SAF Client
SAF External Clianl
Cisco IOS
SAF Advertisement as Exchanged
Between SAF Forwarders
CCD is a function ofcall agents. Itallows call agents to advertise locally known internal
directory numbers and the corresponding PSTN numbers to other CCD-enabled call agents.
CCD utilizes SAF for distributing call-routing infonnation over the SAF-enabled network.
ACCD-enabled call agent is configured to send its locally configured directory number range
as a SAF service to a SAF forwarder. The CCD SAF client generates SAF service data (call
reachability information) and passes iton to the SAF forwarder that will propagate the
information within the SAF network. All SAF forwarders that have SAF subscribing clients
that are attached send the SAF service data to their clients. From aCCD perspective, all SAF
clients exchange SAF service data (call-routing information).
You can compare the SAF service data with the TCP or the User Datagram Protocol (UDP).
which establishes an end-to-end communication between IP endpoints. Likewise, CCD-enabled
call agents exchange call-routing infomiation via the end-to-end service data exchange.
The SAF header can be compared to the IP header. It is also interpreted at intermediate nodes
(SAF forwarders), while these intermediate network nodes do not process the payload (that is.
the SAF service data).
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© 2010 Cisco Systems, Inc.
SAF Client and SAF Forwarder Functions
This subtopic describes SAF client and SAF forwarder functions.
SAF Client and SAF Forwarder Functions
SAF client functions:
• Register to the network
• Publish services
• Subscribe to services
• Send keepalives
SAF forwarder functions:
* Propagate updates received from SAF clients to other SAF
forwarders
• Send hellos to other SAF forwarders
• Propagate updates to SAF clients
SAF clients register to the network, more precisely lo aSAF forwarder. They can publish
sen ices (that is. advertise infonnation) to the SAF network or subscribe to services (that is.
request infonnation) from the SAF network. In order to allow the SAF client and the SAF
forwarder to quickly detect dead peers (for example, ifthe device was powered off), they
exchange keepalives.
SAF forwarders propagate updates that are received from SAF clients that publish sen ices to
other SAF forwarders. Thev send updates to SAF clients, which subscribe to services. Ir.
addition. SAF forwarders exchange hellos with other SAF forwarders in order to detect dead
peers.
Call Control Discovery
© 2010 Cisco Systems. Inc.
SAF Neighbor Relationships
This subtopic describes how SAF forwarders can be connected to each otht
SAF Neighbor Relationships
Two options for neighbor relationships:
• Layer 2 adjacent
- " k k- ?(with dynamic discovery)
- Unicast (with static configuration)
• Non Layer 2 adjacent
- Static configuration
SAF forwarders can be. but do not have to be, directly neighboring devices. If they are Layer 2
adjacent, there are two configuration options:
• Multicast: When multiple SAF forwarders are connected via abroadcast-capable medium
like a LAN using Fthernet, they can communicate toeach other via multicasts. This
communication allows adynamic neighbor discovery because there is no need to statically
configure the Layer 2 adjacent neighbors.
• Unicast: When it is not desired that all SAF devices on abroadcast-capable medium
automatically discover each other. SAF forwarders can be configured to send updates only
to statically configured neighbors via unicast messages. In the figure, the lower-left
example shows three routers that are connected to an Fthemct. However, other than in the
upper-right example, they should not build adjacencies among each other in a full-mesh
fashion. Instead, they should communicate only in ahub-and-spoke fashion (one router
communicates with both ofthe others, and the other two routers do not communicate
directly with each other).
When SAF forwarders are not Layer 2adjacent—that is, when there are one or more IP hops
between them—these nonadjacent neighbors have to be statically configured. No discoverv is
possible. See the Hlustration in the lower-right comer of the figure for an example of non-L aver
2 adjacent SAF forwarders.
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SAF Routing Characteristics
This subtopic describes the main characteristics ofSAF routing.
SAF Routing Characteristics
• SAF-FP uses features and functions of EIGRP, including:
- Bandwidth percent
Hello interval
Hold time
- Split horizon
Authenticated updates
- Incremental updates (only when changes occur)
• Independent of IP routing protocol
Static
• Dynamic (EIGRP. OSPF, BGP, and so on)
The SAF-FP uses features and functions of the Enhanced Interior Gateway Routing Protocol
(F1GRP) for SAF routing. Features and mechanisms that are utilized and known from EIGRP
include the DifTusing Update Algorithm (DUAL) to prevent loops, reliable transport over IP (IP
protocol 88). support for authenticated updates, and incremental, event-triggered updates tor
fast convergence and low -bandwidth consumption. Configurable parameters that relate to these
FIGRP-derived features include bandwidth percent, hello interval, holdtime. split horizon,
maximum hops, andmetric weights.
Although SAF routing is verv like EIGRP, it is independent of the used IP routing protocol,
SAF works over static routing, as well as in networks that use dynamic routing protocols such
as FIGRP Open Shortest Path First (OSPF). and Border Gateway Protocol (BGP).
Call Control Discovery
i 2010 Cisco Systems. Inc
SAF Message Components
This subtopic shows the two components of a SAF message.
SAF Message Components
Relevant to SAF forwarders
Relevant to SAF clients
Identifies service type and
Service-specific
unique instance
information
Used by forwarders to
Transparent to forwarders
propagate advertisements
Metrics used to avoid loops
Client data depends on
service type
A SAF message consists of two components:
•
SAF header: The SAF header is relevant mainly to SAF forwarders. It identifies the
service type (for example, CCD) and includes information that is relevant for the dynamic
distribution of SAF services, such as metrics and loop detection information.
•
SAF service data: SAF service data is relevant only to the SAF client. A SAF forwarder
cannot interpret the SAF service data. SAF service data includes the IP address and port of
the advertising SAF client and detailed client data that describes the advertised service.
With CCD. client data includes call-routing information such as directory numbers, the IP
address of the call control device, the signaling protocol to be used to communicate with
the call control device. PSTN prefixes, and so on.
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SAF Client Types
This subtopic describes the two available CCD SAF client types.
SAF Client Types
SAF forwarder is always Cisco IOS router.
Two types of SAF clients:
Cisco Unified
Communications
Manager
External SAF clients
*
SAF client and SAF forwarder are
different devices.
« SAF client is Cisco Unified
Communications Manager.
Internal
*
SAF-CPisused.
API
• Internal SAF clients
•
SAF client and SAF forwarder are
colocated functions within the
same device.
*
SAF clients are Cisco Unified
Communications Manager
Express, Cisco Unified SRST,
Cisco Unified Border Element,
and Cisco IOS gateways.
•
Internal API is used
A SAF forwarder is alwavs a Cisco IOS router. Remember that SAF forwarders do not process
the propagated service infomiation. Their function is to propagate the information within the
SAF network and lo pass it on to SAF clients.
SAF clients then interpret the service infomiation. With CCD. the SAF client is a call control
device, which sends and receives call-routing infomiation. Depending on the type of call
control device, the CCD device can be an internal or external SAF client:
*•
•
External SAF client: The SAF client and the SAF forwarder are two different devices.
They use the SAK-C1' for communication. An exampleof a CCD externalSAF client is
Cisco Unified Communications Manager.
•
Internal SAF client: The SAF client and the SAF forwarder arc two different functions
within the same device—a Cisco IOS router. They use an internal application programming
interface (API) for communication. Examples of CCD internal SAF clients are Cisco
Unified Border Element. Cisco Unified SRST. Cisco Unified Communications Manager
Express, and Cisco IOS gateways.
©2010 Cisco Systems, Inc
Call Control Discovery
SAF Characteristics
This topic describes the characteristics of SAF.
SAF Components
SAF supports any service
lo be advertised
CCD is the first Cisco application
Cisco Unified
Cisco Unified
Communications
Communications
Manager
Manager
using SAF to advertise services
(call routing)
SAF network components.
•
Exchange service
information among
each other
•
Use the SAF Forwarding
Protocol (SA^-^PI
Advertise services lo and
leam services from SAF
forwarders
Use SAF Client Protocol
(.
."I to interact with
SAF forwarders
(With CCD) serve as call
agents
SAF is a network-based, scalable, bandwidth-efficient, real-time approach to service
advertisement and discovery.
SAF can be used to advertise and learn any service to and from the SAF-enabled network. CCD
is the first Cisco application that utilizes SAF. As mentioned earlier, the devices within the SAF
network are SAF forwarders. They have the responsibility to propagate services within the
network. SAF forwarders do not interpret the service information itself; they only guarantee
fast and reliable exchange of the information. SAF forwarders use the SAF Forwarding
Protocol (SAF-FP) between each other.
SAF forwarders can interact with SAF clients. A SAF client is an entity thatprocesses SAF
service data.A SAF client can independently advertise (generate) SAF service information to
be propagated in the network, or subscribe to (receive) SAF service information. A SAF client
communicates with oneor more SAF forwarders by the SAF Client Protocol (SAF-CP). With
CCD. the SAF client is a call agent such as Cisco Unified BorderElement, Cisco Unified
SRST. Cisco Unified Communications Manager, Cisco Unified Communications Manager
Express, and Cisco IOS gateways.
5-8
Implementing Cisco Unified Communications Manager, Part2 (CIPT2) vfl.O
) 2010 Cisco Systems, Inc.
Call Control Discovery Overview
This subtopic provides an overview of CCD.
CCD Overview
CCD-enabled call agents
advertise to and leam from
"the network"
SAF is used to distribute
information within the network.
SAF forwarders interact with
CCD-enabled call agents
(SAF clients)
- SAF forwarder learns
information from SAF client
•- SAF forwarders exchange
information with each other.
SAF forwarder advertises
all learned information to
SAF client.
CallAgent Catl/
With CCD.each CCD-enabled call agent advertises locally found directory numbers or
director.' number ranges and their corresponding PS'FN numbers or prefixes to theSAFenabled network. In addition, each CCD-enabled call agent learnscall-routing infonnation from
the network.
SAF isused to propagate infomiation within the SAF-enabled network. SAF forwarders
interact with CCD-enabled call agents (thatis. SAF clients). A SAF forwarder learns
infonnation from a SAF client. SAF forwarders exchange learned call-routing information with
each otherso that the SAF-enabled network is aware of all learned call routes. SAF forwarders
do not onh leam from SAF clients, but they alsoadvertise all learned information lo SAF
clients, fhat way. all SAF clients are aware of all available call-routing information—internal
director, numbers and their corresponding PSTN numbers.
©2010 Cisco Systems. Inc
Call Control Discovery
Scalable Dial Plan Solution for Large Networks
This subtopic describes a scalable dial plansolution for large networks.
Scalable Dial Plan Solution for Large Networks
Solutions for dynamic exchange of routing information exist.
-
Dynamic IP routing protocols.
•
Routers have local networks attached.
•
Routers advertise local networks to other routers.
• All routers leam all available networks and how lo gel there.
Same concept can be used for call-routing information.
- Call-routing domains advertise telephone numbers or number ranges.
•
Internal numbers and IP address for VoIP
•
External numbers for PSTN backup
Call Control Discovery has been introduced with Cisco Unified Communications
Manager Version 8.
- Call agents can advertise and leam call-routing information using
•
Cisco Unified Communications Manager
• Cisco Unified Communications Manager Express
•
Cisco Unified SRST
•
Cisco Unrfied Border Element
• Cisco IOS gateway
The problem of dynamically distributing reachability information is known also in areas other
than call routing. In IP networks, for example, routing has changed from simple static routing
to large, fully dynamic clouds, such as the Internet.
The solution for scalable IP routing is provided by dynamic routing protocols. IP routers have
local networks that are attached. They advertise these locally known networks to other routers
so that all routers can leam about all available networks and the path to get to those networks.
The same concept can be used to distribute call-routing infonnation. Each call-routing domain
advertises locally known telephone numbers or number ranges. Because local numbers are
typically used by internal patterns (using VoIP) as well as via the PSTN, each call-routing
domain advertises both the internally used numbers and the corresponding external PSTN
numbers.
Cisco CCD. a new feature that was introduced with Cisco Unified Communications Manager
Version 8. provides exactly such a service. It allows Cisco Unified Border Element. Cisco
Unified SRST. Cisco Unified Communications Manager, Cisco Unified Communications
Manager Express, and Cisco IOS gateways to advertise and leam call-routing information in
the form of internal directory numbers and PSTN numbers or prefixes.
5-6
Implementing Cisco Unrfied Communications Manager, Part 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Dial Plan Scalability Issues in Large Networks
The main scalability issues in targe networks are caused by the fact that call-routing
infonnation has to be configured separately at each call-routing domain.
Dial Plan Scalability Issues m Large Networks
• Call-routing information between separate call-routing domains
has to be manually configured:
Full-mesh configuration
• Extremely complex, suitable only for small networks
Hub-and-spoke configuration when centralized call-routing
entities (SIP network services or H.323 gatekeepers) are used
• Scales better than full-mesh topologies
• Requires redundant deployment of central services
• Changes have to be manually configured.
• PSTN backup has to be implemented independently at each callrouting domain.
• No dynamic exchange of call-routing information and no
automatic PSTN backup.
Without centralized services (such as H.323 gatekeepers or SIP network services), a full-mesh
configuration is required. In other words, each call control domain has to be configured with
call-routing information toward all other call-routing domains. This implementation model does
not scale at all and therefore is suitable only for smaller deployments.
In a hub-and-spoke deplovment model, call-routing information for each call-routing domain is
eonligured onk once at the centralized call-routing entity. This eentrali/ed call-routing entity
can be a SIP network service or an H.323 gatekeeper. Such a solution scales better than fullmesh topologies: however, it introduces a single point of failure and therefore requires
redundant deployment of the centralized service. In addition, the centralized call routing still
has to be manually configured. For example, if telephone number ranges or prefixes are
changed at one of the call-routing domains, these changes also have to be manually performed
at the centralized call-routing service. Further, PSTN backup has to be implemented
independently at each call-routing domain.
In summary, there is no dynamic exchange of call-routing infomiation between call-routing
domains, and there is no automatic PSTN backup.
mm
i 2013 Cisco Systems, Inc.
Call Control Discovery
SAF and CCD Overview
This topic prov ides an overviewabout SAFand CCD.
Dial Plans in Large Networks
Dial plans in large networks are difficult to implement and maintain.
Centralized call-routing intelligence improves scalability but still
does not scale well in very large networks.
Cal Asent
Call tgtrl
Call AOent_sia_Call Agent
Call Agent
Call Agem
Call Agent
Call Agent
Call Agent Call Agent
CaHAgent Call Agent
In large networks with several call agents—such as Cisco Unified Communications Manager
Express. Cisco Unified Communications Manager, Cisco Unified Border Element, Cisco
Unified SRST. and Cisco IOS gateways—the implementation and maintenance of dial plans
can be very complex.
The use of 11.323 gatekeepers or Session Initiation Protocol (SIP) network services reduces the
complexity. However, dial plan implementation still does not scale well in very large
deployments.
5^
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
J2010 Cisco Systems, Inc.
Lesson 1
Implementing SAF and CCD
Overview
With the increasing deployment of Cisco Unified Communications solutions, dial planshave
become more complex to implement, especially in largeenterprises that consistof numerous
call control devices. Some examples of these call control devices arc Cisco Unified Border
Element. Cisco Unified Survivable Remote Site Telephony (SRST), Cisco Unified
Communications Manager. Cisco Unitied Communications Manager Express, and Cisco IOS
gateways.
To simplifv dial plan implementation in large deployments, it is desirable thatcall control
devices dynamically exchange call-routing infonnation so that no any-to-any static
configuration is required. Cisco Service Advertisement Framework (SAF) allows services to be
propagated through SAF-enabled network devices. Cisco Call Control Discovery (CCD) is the
first application that utilizes SAF to advertise services: the reachability of internal director.'
numbers and publicswitched telephone network (PS'fN) backupnumbers.
This lesson explains how SAF works, describes its components, andshows how to configure it.
The lesson also describes how CCD utilizes SAF to dynamically exchange call-routing
infonnation and how you can implement CCD in Cisco Unified Communications Managerand
in Cisco IOS Software.
Objectives
Upon completing ihis lesson, you will beable to describe and implement SAF clients and
forwarders in an environment with CCD. This ability includes being able to meet these
objectives:
Describe what SAF is. what CCD is. and how CCD utilizes SAF
Describe the characteristics of SAF
Describe the characteristics of CCD
Describe how CCD works
Describe how to implement SAF and CCD
Describe special considerations for using SAF and CCD
Implementing Cisco Unified Communications Manager, Part 2 (C1PT2) v8.0
© 2010 Cisco Systems, Inc
Module 5
Call Control Discovery
Overview
In large deployments with many call agents, dial plan implementation can be very'complex.
Cisco Service Advertisement Framework (SAF) and Call Control Discovery (CCD) allow call
agents to propagate call-routing infonnation to the network and to learn routes from the
network. Thus. SAF and CCD facilitate the deployment of very large Cisco Unified
Communications solutions by greatly simplifying dial plan implementation.
This module explains how SAF and CCD work and how you can implement SAF and CCD to
allow the duiamic discover) of call-routing information.
Module Objectives
Upon completing this module, you will be able lo describe and implement CCD deployments.
I his ability includes being able to meet this objective:
•
Describe and implement the SAF client and forwarder in an environment with CCD
Module Self-Check Answer Key
4-74
QU
B
Q2)
C. D.G
03)
A.C
Q4)
C
Q5)
B
06)
B
Q7)
A
Q8)
CD
09)
A. D. E
Q10)
A, D
Oil)
A, B
Implementing Cisco Unitied Communications Manager. Part 2 (CIPT2) v8.0
© 20t0 Cisco Systems, Inc.
Oil)
Which two statements describe the result when theuser logs into a device but is still
logged in to another device? (Choose two.) (Source: Implementing Cisco Extension
Mobility)
A)
If the multiple login behaviorserviceparameter is set to disallowed, the login
fails.
B)
If the multiple login behaviorservice parameter is set lo auto-logout, the user is
automatically logged out of the other device.
C)
If the multiple login behaviorenterprise parameter is set to allowed, the login
succeeds and the user also remains logged in at the other device.
2010 C sco Systems, inc.
D)
If the multiple login behaviorenterprise parameter is set lo prompt, the user is
E)
given the option to log out of the other device first.
fhe login fails.
Implementation of Features and Applications for Multisite Deployments
Q6)
Q7)
If no physical location is configured at the device pool, the physical location that is
configured at the phone is used. (Source: Implementing Device Mobility)
A)
true
B)
false
Which of the following is not a problem when users roam between sites? (Source:
Implementing Cisco Extension Mobility)
A)
B)
C)
D)
Q8)
Which two settings cannot be updated when Cisco Extension Mobility is used?
(Choose two.) (Source: Implementing Cisco Extension Mobility)
A)
B)
C)
D)
E)
F)
Q9)
The phones that they use have the wrong location and region settings.
Users get the wrong extensions on their phones.
Users get the wrong calling privileges.
Users do not have speed dials available.
phone button template
softkey template
device CSS
network locale
phone service subscriptions
phone lines and speed dials
Which three configuration elements are not relevant for Cisco Extension Mobility
configuration? (Choose three.) (Source: Implementing Cisco Extension Mobility)
010)
A)
location
B)
C)
D)
F)
F)
Ci)
phone
end user
device security profile
device pool
device profile
phone service
Which two of the following are recommended approaches to implementation of calling
privileges when Fxtension Mobility is used? (Choose two.) (Source: Implementing
Cisco Extension Mobility)
A)
Configure the line or lines of the device profile of the user with a CSS that
includes blocked route patterns for the destinations that the user should not be
allowed to dial.
B)
C)
D)
E)
Do not configure a device CSS.
Do not configure a line CSS.
Configure the device with a CSS that includes all PSTN route patterns pointing
to die local gateway.
Configure the line or lines of the physical phone with a CSS that includes
blocked route patterns for the destinations that the user should not be allowed
to diai.
4-72
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
© 2010 Cisco Systems, Inc.
Module Self-Check
Use the questions here to review what you learned in this module. The correct answers and
solutions are found in the Module Self-Check Answer Key.
Which setting is not modified when a user is roaming between sites with a device1!
Qi)
(Source: Implementing Device Mobility)
021
A)
region
B)
directorv number
O
location
D)
SRST reference
Which three of the following are device mobility-related settings in Device Mobility?
(Choose three.) (Source: Implementing Device Mobility)
A)
Region
B)
SRST Reference
C)
D)
F.)
AAR Calling Search Space
Device Mobility Calling Search Space
Media Resource Group List
F)
Location
Ci)
AAR Group
Which two statements are correct about (he relationship between Device Mobility
03)
configuration elements? (Choose two.) (Source: Implementing Device Mobility'.
A)
B)
C)
D)
E|
Device Mobility Infos refer to one or more device pools.
Device pools refer to one or more physical locations.
Device pools can refer to one device mobility group,
Device pools can refer to one Device Mobility Info.
Physical locations refer to device mobility groups.
Which statement is not correct about Device Mobility operation? (Source:
Q4)
Implementing Device Mobility)
A)
B)
C)
A dev ice pool is selected basedon the IP address of the device.
If the selected device pool is the home device pool, no changes are made.
If the selected device pool is in a different device mobility group than the home
dev ice pool, the device-mobility-related settings of the roaming device poolare
applied.
D)
If the selected device pool is in a different physical location than the home
dev ice pool, the roaming-sensitive settingsof the roaming device pool are
applied.
05)
Which statement is not correct about the interaction of Device Mobility and globalized
call routing? (Source: Implementing Device Mobility)
A)
B)
C)
D)
©20'0
ico Systems. Inc
The user of a roaming phone can use the home dial rules.
Fhe user of a roamingphone can use the homedial rules, but then the home
gateway is used all of the time.
The user of a roaming phone can use the roaming gateway.
The same device mobility group can be used at all devicepools.
Implementation of Features and Applications for MultisiteDeployments
4-70
Implementing Cisco Unified Communicalions Manager, Part 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Mo lule Summary
This lopic summarizes the kev points that were discussed in this module.
Module Summary
• Device Mobility allows users to roam between sites with their
phones.
• Cisco Extension Mobility allows users to log in to any phone
in a Cisco Unified Communications Manager cluster and
have a personal profile applied to the phone.
fhismodulebegins with a description of Device Mobility. Itdescribed how Device Mobility
can be implemented in environments with and without globalized call routing, fhen the -nodule
described the operation and implementation ofCisco Extension Mobility that allows Cisco
!
j
Unified Communications Manager users to log into an IP phone and have their personal profile
applied, regardless ofthe device and physical location that they are using.
Rel frences
For additional infonnation. refer to these resources:
•
•
Cisco Systems. Inc. Cisco Unified Communications System 8.x SRND, April 2010.
hup: www.cisco.conVen/I IS/docvYoiee ip comm/cucm/srnd/Kx/iicSx.htmi
Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0(1). February 2010.
!itlp:.:\\\\w.ei^co.conv'cn/US''docs;voiceJp_comm/cucm/admin/K_0 I.ecmcfg'bccn-SOlem.html
• Cisco Systems. Inc. Cisco Unified Communications Manager Features and Services Guide.
Release'8.0{i). March 2010.
imp: www.cisL'o.coin't'iiT^.'docs/voice ip comm/eucm/admin/8 0 l/ccmfeat/fsg^-801cm.html
;isco Systems. Inc
Implementing Features and Applications for Multisite Deployments
4-69
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
The Device Mobility and Cisco Extension Mobility features of
Cisco Unified Communications Manager allow users to roam
between sites,
Cisco Extension Mobility enables users to log into IP phones
and apply their profiles, including extension number, speed
dials, services, MWIstatus, and calling privileges.
The device profile of the user is used to generate the phone
configuration in the login state.
Seven steps are needed to configure Cisco Extension
Mobility.
References
For additional information, refer to these resources:
•
Cisco Systems. Inc. Cisco Unified Communications Svstem Release 8.x SRND. San Jose,
California. April 2010.
http:/.'www. cisco.com/en/lJ S/docs/voiee_ip_comm/eucm/srnd/8x/ue8.\srnd.pdf.
•
Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0(2). San Jose, California, March 2010.
http:.'/v\ww.cisco.com''en/US/docs/voice ip comni/cuem/admin/8 0 2/ccmcfg/bccm.pdf.
•
Cisco Systems. Inc Cisco Unified Communications Manager Features andServices Guide
Release 8.0(2). San Jose, California, March 2010.
http://vvw\\xisc().com/en/US/docs/voice__ip_comm/cucni/admin/8J)_2/cciiifeal/fsgd.pdf'.
4-68
Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0
) 2010 Cisco Systems, Inc
Step 7b: Subscribe Phone to Cisco Extension Mobility Phone
Senvice
The last step of Cisco Extension Mobility configuration is to subscribe the IP phoneto the
Cisco Extension Mobilitv phone service.
Step 7b: Subscribe Phone to Cisc<
Extension Mobility Phone Service
In the Phone Configuration window, choose
Subscribe/Unsubscribe Services from Related Links to open
the Subscribed Cisco IP Phone Services window.
Subscribed Qgn IP Phone Services tor SeptMH4C**551S6
2
rScnrtte lnfonnaMon
name as rt should
Enter the service
appear at the phone.
1 Logon 1 LcbjI'
FM Lngfm ' Loaon'
FM logon / '-v^fflt
-Sulrairlbed Scrvic
-| rvcxl I
Close
1 Choose the Cisco
Exlension Mobility
phone service by
using the name
assigned in Step 3
3. Click Subscribe.
Then, click Next
Then, click Save
The process ofsubscribing the IP phone tothe Cisco Extension Mobility service isthe same as
the process that was explained in Step 5. in which the device profile was subscribed tothe
Cisco Extension Mobility service. Inthe Phone Configuration window, use the related link
Subscribe/Unsubscribe Services to open the Subscribed Cisco IP Phone Services window and
subscribe to the sen ice.
) 2010 Cisco Systems, Inc
Implementation ofFeatures and Applications for Multisite Deployments
4-67
Step 7a: Configure Phones for Cisco Extension Mobility
Finally, the phone must be enabled for Cisco Extension Mobility and subscribed to the Cisco
Extension Mobility phone service. The figure shows the first part—enabling Cisco Extension
Mobilitv on a phone.
Step 7a: Configure Phones for Cisco
Extension Mobility
Cisco Unified Communication ManagerAdministration: Device >
kJ Oct* rt I.
Enable or disable Cisco
Extension Mobility;
choose specific device
profile or use current
device settings for logout
state.
In the PhoneConfiguration window(which you can access from Cisco Unified
Communications Manager Administration by choosing Device > Phone), check the Enable
Extension Mobility check box to enable Cisco Extension Mobility. Then choose a specific
device profile or the currently configured device settings to use during the logout state. The
recommendation is to use the currentdevice settings.
B*.
4-66
Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0
>2010 Cisco Systems, Inc.
Step 6: Associate Users with Device Profiles
This step describes how to associate users withdevice profiles.
Step 6: Associate Users with Device
Profiles
Cisco Unified Communications Manager
Administration: User Management > End User
l™"
~<
—1
=====
1
Usei ID and PINaie
1
]
used for Cisco
J Extension Mobility login
1
-fr.--.o-* «j-t*f
Associate user
-
/
device profiles,
optionally, set
default profile
VA
(if moielhan
A
one controlled
profile exists)
Inthe End User Configuration window (which you can access from Cisco linified
Communications Manager Administration by choosing User Management > End User), choose
the device profile orprofiles that you want to associate with the user in the list of Available
Profiles. Click the down arrow to add them lo the list of Controlled Profiles.
©2010 Cisco Systems. Inc
Implementation ofFeatures and Applications for Multisite Deployments
4-6S
Step 5b: Subscribe Device Profile to Cisco Extension Mobility
Phone Service
The next step is to subscribe the configured device profileto the Cisco Extension Mobility
phone service.
Step 5b: Subscribe Device Profile to
Cisco Extension Mobility Phone Service
In the Device Profile Configuration window, choose
Subscribe/Unsubscribe Services from Related Links to open the
Subscribed Cisco IP Phone Services window.
ESmmmmmm^LmmmW
2. EnlerBie service
name as it should
appear at the phone.
Sew* a £erv-c*'
5ei-Lic« Des^ripfic
EH Lo^on ; Lagotf
Not Selected
-
Sen.-*
0* mo5e<v«
otion: EM Logon / Logoff
EM Logon .' Logoff
EM Logon / Logoff
1. Choose the Cisco
Extension Mobility
phone service by
using the name
- Subscribed Services
assigned in Step 3.
3. Click Subscribe.
Then, cick Next.
Then, click Save,
In the Device Profile Configuration window, choose Subscribe/Unsubscribe Services from
the Related Links field and click Go. Then choose the phone service that you added in Step 3.
Click Next, and enter the name with which the phone service should be displayed in the list of
phone services on the IP phone after the Services button is pressed. Click Subscribe, and then
click Save. The device profile is now subscribed to the Cisco Extension Mobility service.
Caution
If the device profile is not subscribed to the Cisco Extension Mobility service, users do not
have access to Cisco Extension Mobility phone service after they log in and their device
profile has been applied. As a result, users can no longer log out of Cisco Extension Mobility
at the phone. Therefore, make sure that you do not forget to subscribe the phones (see Step
7b)—and the device profiles—that you use for Cisco Extension Mobility to the Cisco
Extension Mobility phone service.
Since Cisco Unified Communications Manager version 7, an enterprise subscription can be
enabled at each phone service. If an enterprise subscription is enabled, the corresponding
phone service applies to all phones and device profiles.
mm
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ImplementingCisco Unified Communications Manager, Part 2 (CIPT2)vfl.O
>2010 Cisco Systems, Inc.
Step 5a: Create Device Profiles
The next step is the configuration of device profiles.
Step 5a: Create Device Profiles
Cisco Unified Communications Manager Administration: Device
> Device Settings > Device Profile
5
PKor n.*-™ TrPijtn.E*
Setter Tcmcla-rB
Configure phone
lines and buttons.
[g-.=rr
1 (Default) Profile l«lfOrM*lkirl-
modeiand
protoccrf.
IK^
Enter device
profile name and
description.
Configure userspecific phone
parameters.
To configure device profiles, in Cisco Unified Communications ManagerAdministration,
choose Device > Device Settings > Device Profile. After choosing the phone model and
protocol, vou can configure user-specific deviceconfiguration parameters. Afteryou configure
the phone button template, you can configure the appropriate buttons.
m
© 2010 Cisco Systems. Inc
Implementation of Features and Applications for Multisite Deployments
4-63
Step 4: Create Default Device Profiles
If multiple phone models are used for Cisco Extension Mobility, default device profiles should
be enabled.
Step 4: Create Default Device Profiles
Cisco Unified Communications Manager Administration: Device
> Device Settings > Default Device Profile
1
J
Select phone
modeland
™"
Set default phone
configuration for
the selected phone
type.
—
To configure a default device profile, in Cisco Unified Communications Manager
Administration, choose Device > Device Settings > Default Device Profile. Choose the
product type (phone model) and device protocol first. You can then configure the default device
profile for the chosen phone model and protocol.
Note
The available configurationoptions depend on the chosen phone model and protocol. The
default device profiledoes not include phone button configuration (for example, lines or
features buttons) but does include the phone button template.
«M
4-62
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
52010 Cisco Systems, Inc.
Step 3: Add the Cisco Extension Mobility Phone Service
fhe next step is to add the Cisco I'xlension Mobility phone service.
Step 3: Add the Cisco Extension
Mobility Phone Service
Cisco Unified Communications Manager
Administration: Device > Device Settings > Phone
Services
IP Ptwne Servke*, Configuration
Enter service name and
S»rvi£P Name'
EH Logon ,' LogcC
A5CTI Service f*a'n
EH Logon ,J LOQOt*
S=rv>cc Descfipt.oi
Extension Mcbilil. ..ckju
Semce UHL
htrp ''101 i.i.eoeoien OPC'EWAnpSt vlet'devic c-tDI
service description;
enter Cisco Extension
Uicif 5trv.c
<l
Mobility service URL.
^?rMrj Catega
Enable the Cisco
Extension Mobility IP
phone service.
Setvice URL
htlp /:Server_IP_Aa<t-ess S0SO'emBpp/EMAppServlePdevice=#OEVICENAMEB
Cisco fxtension Mobility is implemented as a phone service, fherefore,you must add this
service to the available phoneservices in Cisco Unified Communications Manager. To add the
Cisco Extension Mobility phone service, in Cisco Unified Communications Manager
Administration, choose Device> DeviceSettings > Phone Services. Configure the Cisco
Fxtension Mobility service with a service name anddescription, andthenenterthe service
URI.:http:'.V/'rt'r IP ,-W(//-csj:8080.'cinapn/l:MAppScrv!cl?dc\icc=rfI)l'.VICFNAMI;.S.
Note
The service URL is case-sensitive and can be found in the Cisco Unified Communications
Manager Help pages.
© 2010 Cisco Systems, Inc
Implementation ofFeatures andApplications for Multisite Deployments
4-61
Step 2: Set Cisco Extension Mobility Service Parameters
The Cisco Extension Mobility service has several configurable service parameters.
Step 2: Set Cisco Extension
Service Parameters
Cisco Unified Communications Manager
Administration: System > Service Parameters
Cisco Fxtension Mobility can be configured with the following service parameters.
If the Enforce Intra-cluster Maximum Login Time parameter is set to True, the user is
automatically logged out after the Intra-cluster Maximum Login Time expires. The Intra-cluster
Multiple Login Behavior parameter specifies how to process users who log into a device but are
still logged in at another device. There are three options: Login can be denied, login can be
allowed, or the user can be logged out automatically from a phone on which the user logged in
earlier and did not log out.
Alphanumeric User ID can be enabled or disabled, and the last logged in username can be
remembered (and presented as a default on the next login) by setting the Remember the Last
User Logged In parameter to True. Call lists can be preserved or cleared at logout, depending
on the setting of the Clear Call Logs on Intra-Cluster EM service parameter.
Note
All of these parameters are clusterwide service parameters of the Cisco Extension Mobility
service and can be accessed from Cisco Unified Communications Manager Administration
mm
by choosing System > Service Parameters.
mm.
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Step 1: Activate the Cisco Extension Mobility Service
fhe first step is to activate the Cisco Extension Mobility feature service.
Step 1: Activate the Cisco Extensh
Mobility Service
Cisco Unified Communications Manager
Serviceability: Tools > Service Activation
I Control Center - feature Services
-
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I>e actuated
lo enable Cisco Extension Mobility, you must activate the Cisco Extension Mobility feature
service from Cisco Unified Serviceability. To do so, click Tools > Service Activation.
Note
Starting with Cisco Unified Communications Manager Version 6.0, Cisco Extension Mobility
is considered a User Facing Feature and can be activated on any server in a Cisco Unified
Communications Manager cluster, to provide a redundant Cisco Extension Mobility
environment
*>
© 2010 Cisco Systems. Inc.
Implementation of Features and Applications for Multisite Deployments
4-59
Cisco Extension Mobility Configuration
This topic describes how to configure Cisco Extension Mobility.
Cisco Extension Mobility Configuration
Steps
1
Activate the Cisco Extension Mobility service.
2
Set Cisco Extension Mobility service parameters.
3
Add the Cisco Extension Mobility phone service.
A
Create default device profiles for all phone models used
(optional).
5.
Create device profiles and subscribe them to the Cisco
Extension Mobility phone service.
6
Create end users and associate them with device profiles.
7.
Enable Cisco Extension Mobilityfor phones and subscribe
phones to the Cisco Extension Mobilityservice.
The figure lists the steps that are required to configure Cisco Extension Mobility in Cisco
Unified Communications Manager. The following topics explain these steps in detail.
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Alternatives for Mismatching Phone Models and CSS
Implementations
To avoid issues with mismatching IP phone models or with calling privileges when the
traditional approach for implementing partitions andCSSs is used, multiple device profiles can
be configured per user.
Alternatives for Mismatching Phone
Models and CSS Implementations
Alternative to using the default device profile and the (applicable)
settings of the user device profile.
Create multiple device profiles for the user (one per phone
model family).
Alternative ts applicable only ifusers use Cisco Extension
Mobility across phone model families that do not support feature
safe
Alternative to line/device CSS approach in multisite environments.
Create multiple device prof les for the user (one per physical
location), giving each profile the appropriate CSS to enforce CoS
and to choose the conect gateway.
Neither alternative scales well.
Using Local Route Groups is recommended ifthe traditional CSS
approach (CSS applied at the line) is followed.
When different phone model series are used, issues can arise when the settings of the default
device protlle are applied. DilTerenl users might require different settings. This problem can be
solved by creating multiple device profiles peruser. When you configure and associate one
device profile (per phone model) with a username, Cisco Unified Communications Manager
displavs this list ofprofiles after successful login. The user can choose a device profile that
matches the phone model ofthe login device. Ilowever. ifmany users need touse Cisco
E\tension Mobility and many different phone models arcused, this solution does notscale
well.
The same concept can be used as an alternative tothe line/device approach for implementing
CSSs. Aseparate device profile can be created per site and is configured with the appropriate
CSS to allow local gatewa> s to be used for external calls. Again, the user chooses the
corresponding device profile after logging in. and the correct CoS and gateway choice are
applied without depending on a separate line and device CSS. The recommendation, however,
isto use the line/device approach in a multisite environment, because that approach simplifies
the dial plan and scales belter.
Note
When using thetraditional CSS approach with only oneCSS applied at theline, use Local
Route Groups to prevent gateway-selectionproblems.
© 2010 Cisco Systems. Inc
implementation ofFeatures and Applications for Multisite Deployments
4-57
Cisco Extension Mobility and CSSs
This subtopicdescribes how Cisco F.xiension Mobility interacts with CSSs.
Cisco Extension Mobility and CSSs
• Cisco Extension Mobility does not modify the device CSS.
• Cisco Extension Mobility modifies the line CSS:
- When using the line/device CSS approach (recommended for CoS
implementation).
• Line CSS of user device profile is applied; CoS settings of the user
are enforced
• Device CSS is not modified; local gateway selection is allowed,
depending on used device (at any location).
• When using the traditional CSS approach (only one CSS at phone), the
same CSS is used all the time, causing problems in multisite
environments with different classes of sen/ice for users.
- For proper gateway selection, use Local Route Groups if the CSS is
applied at the phone line.
• AAR CSS is configurable only at the device and is never updated by Cisco
Extension Mobility, so local gateway can be used for AAR calls.
Cisco Extension Mobility does not modify the device CSS or the automated alternate routing
(AAR) CSS (both of which are configured at the device level). Cisco Extension Mobility does
replace the line CSS or CSSs that are configured at the phone with the line CSS or CSSs that
are configured at the device profile of the logged-in user.
Thus, in an implementation that uses the line/device approach, the following applies:
•
The line CSS of the login device is updated with the line CSS of the user. This update is
used to enforce the same class of service (CoS) settings for the user, independent of the
physical device to which the user is logged in.
•
The device CSS of the login device is not updated, and the same gateways (those gateways
that were initially configured at the phone before the user logged in) are used for external
route patterns. Because the phone did not physically move, the same local gateways should
be used for PSTN calls, even when a different user is currently logged into the device.
If the traditional approach is used to implement partitions and CSS, the following applies:
•
If only device CSSs are used, the CSS is not updated, and no user-specific privileges can be
applied. The user inherits the privileges that are configured at the device that is used for
logging in.
•
If only line CSSs are used, the line CSS that is configured at the device profile of the user
replaces the line CSS of the login device. In a multisiteenvironment, this configuration can
cause problems in terms of gateway choice because the same gateway is always used for
external calls. To avoid gateway selection problems in such an environment, you should
use Local Route Groups.
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2. The phone button template and the softkey template of the default device profile are
applied to the Cisco Unified IP Phone 7905.
3. The user has access to the phone services that are configured in the Cisco Unified IP Phone
7905 default de\ ice profile.
mm
) 2010 Cisco Systems, Inc
Implementation ofFeaturesand Applications forMullisite Deployments
4-55
How Cisco Extension Mobility Handles Phone Model
Differences
This subtopic describes the phone configuration process when Cisco Extension Mobility isused
with different phone models.
How Cisco Extension Mobility Handles
Phone Model Differences
When a phone model mismatch is identified, Cisco
Extension Mobility works as follows:
• Device profile:
- Copy all dev ice-independent parameters from the device profile
of the u ser
• Default device profile:
- Apply devtce-dependentparameters, such as phone button and
softkey template, from the default device profile.
• Device profile
- Copy device-dependent parameters that can be applied (such as
phone lines and feature buttons, ifsupported by the applied
phone button template).
- Apply phone service subscriptions from the device profile of the
user (if phone services are supported at the phone that is used).
After successful authentication, if the phone model series of the device protlle does not match
the phone model series of the used phone, the following happens:
1. Device-dependent parameters, such as phone button template and softkey template, from
the default device profile are applied lo the phone.
2. Then the system copies all device-independent configuration settings (user hold audio
source, user locale, speed dials, and line configuration, except for the parameters that are
specified under I inc Setting for This Device) from the device profile to the login device.
3. Next, the applicable device-dependent parameters of the device profile of the user are
applied. These parameters include buttons(such as line and feature buttons) that are based
on the phone button template that has been applied from the default device profile.
4.
Finally, if supported on the login device, phone service subscriptions from the device
profile of the user are applied to the phone.
5. If the device profile of the user does not have phone services that arc configured, the
system uses the phone services that are configured in the default device profile of the login
device.
For example, the following events occur when a user who has a device profile for a Cisco
Unified IP Phone 7960 logs into a Cisco Unified IP Phone 7905:
I.
The personal user hold audio source, user locale, speed dials (if supported by the phone
button template that is configured in the Cisco Unified IP Phone 7905 default device
profile), and directory number configuration of the user are applied to the Cisco Unified IP
Phone 7905.
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Default Device Profile and Feature Safe
This subtopic describes the feature safe functionality of Cisco Extension Mobility.
Default Device Profile and Feature Safe
The default deviceprofile is applied only if
the phone model series is different
Cisco Unified IP
(feature safe)
Phone 7945
Phones can use any phone button
template that has the number of line
buttons that the phone model supports
Log In
Series 794x models are equivalent
and can share a Cisco Extension
Mobiity profile.
Series 796x models are equivalent
and can share a Cisco Extension
Mobiity profile.
Series 797x models are equivalent
and can share a Cisco Extension
Mobiity profile
Phone model
User Device
is similar
Prose tor
User dewce
CISCO
profile is
Unified fP
Prone 7940
used
No administration tasks are required to
activate a Cisco Extension Mobility feature
safe
Cisco Extension Mobility feature safe is
protocol independent.
Ihe default deuce profile is applied only ifa users device profile and the phone on which the
user tries to log in are of a different phone model scries (for example, Cisco Unified IP Phone
Scries 794x. 796x. or 797x).
When the phone model scries of the physical phone and the user device profile are the same,
the feature safe function allows different phone models lo be used for user device profiles and
physical phone models.
for example, a user with an associated device profile for a Cisco Unified IP Phone 7940 phone
can log into a Cisco Unified IP Phone 7945 phone without having the default device profile
applied.
No administrative tasks are required to enable feature safe. Feature safe is independent of the
used signaling protocol (SIP or SCCP).
© 2010 Cisco Systems. Inc
Implementation of Features and Applications for Multisite Deployments
Issues in Environments with Different Phone Models
This subtopic describes issues with mixed IP phone environments.
Issues in Environments with Different
Phone Models
Device profile configu ration ineludes phone
mobel inform at on.
Cisco unified IP
Different conigu ration parameters are available,
depending on the phone model lhal is selected
Phone 7945
What ifa user logs into a different phone model
than Ihe model tiat s conlgured in the device
profile of the user'
Log In
Default device profiles can be configured
-
Default device profiles contain default
phone-con figuration parameters for the
Phone model
large! phone:
•
Phone button template
•
Softkey template
•
0etaiilt values for parameters that might
not be available at the device profile of
the user (if fhe phone model specified
series is different
User Dovfcs
PmBftfor
Cisco
Unified IP Phone
Unified IP
7945 is used.
Default device
profile for Cisco
Phone 7965
there supports fewer features)
•
Ooes not include lines or feature buttons
When different IP phone models are used in a Cisco Unified Communications Manager cluster
for which Cisco Extension Mobility is enabled, an end user may log into an IP phone that is of
a different model series than the one that is configured in the device profile of the user.
Different phones support different features. Therefore, when a user logs into a phone that
supports more features than are supported by the model that is associated with the user, the
default device profile is used to apply the parameters that the target phone supports but that are
not included in the device profile of the user. The default device profile includes phone
configuration parameters such as phone button templates, softkey templates, phone services,
and other phone configuration settings. However, the profile does not include button
configuration (including line buttons).
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5. The IP phone is reset and loads the updated configuration.
Now. the phone can be used as it would be used in the home location. Directory' numbers,
speed dials. MWI. and so on. are all correct, regardless of the location and the IP phone that is
used.
Users can log out of Cisco Extension Mobility by pressing the Services button and choosing
Logout in the Cisco Extension Mobility service. If users do not logout themselves, the system
automatically logs them out after the expiration of the maximum login lime (if the appropriate
sen ice parameter has been configured accordingly).
The user is also automatically logged out of a phone when the user logs into another phone and
when Cisco Unified Communications Manager is configured for auto-logout on multiple
logins. Another option is that the next user of the phone logs out a previous user so that the new
user can log in and have the phone that is updated with the settings of that new user. After
logout. Cisco Unified Communications Manager reconfigures the phone either with the
standard configuration of the IP phone or by using another device profile (as specified in the
Phone Configuration window).
© 2010 Cisco Systems. Inc.
Implementation ofFeatures andApplications for Multisite Deployments
4-51
Cisco Extension Mot ility Operation
This topic describes how Cisco I xtension Mobility works, how phone model mismatches are
processed, and how calling scare l spaces (CSSs) and partitions areupdated when Cisco
Extension Mobilitv is used.
Cisco Extension Mobility Login Process
User presses the Sen ices button
on an IP phone and selects the
Cisco Extension Mobi ity service.
The user is authenticated by user
ID and PIN
After successful authentication,
Cisco Extension Mobility selects the
deviceprofile associated with the
user (prompts user to choose, if
multiple associations e'xist).
The IP phone configurjation is
updated with the configuration
parameters from the device profile.
The phone is reset and loads the
updated configuration.
Cisco Unified
Communications
Manager Database
When a userwants to log into a phone, the following sequence of events occurs:
1. The userpresses the Services button on the phone andchooses the CiscoExtension
Mobility service from the list of phone services that are available onthe phone.
2. The Cisco Extension Mobility service requires the user to log inby using a user ID and
PIN. The user enters the required data.
3. Ifthe entered user ID and PIN are correct, Cisco Extension Mobility chooses the device
profile that is associated with the user.
m
Note
If a user isassociated with more than onedevice profile, all associated profiles are
displayed andthe user must choose thedesired profile. Assigning multiple profiles toa user
meansthatthe useris provided a separate device profile per site. This approach is common
when thetraditional approach is used toimplement CSSs. Cisco Extension Mobility updates
onlythe lineconfiguration (including the line CSS), not the device CSS. To allowthe choice
ofa local gateway, a different (line) CSS must be applied persite. In such a scenario, the
userchoosesa site-specific device profile thatdiffers from the device profile thatis usedat
other sites in theline CSS. The line CSS ofsuch site-specific profiles gives accessto route
patterns that route public switched telephone network (PSTN) calls tothe appropriate (local)
gateway
4. Cisco Unified Communications Manager updates the phone configuration with the settings
ofthe chosen device profile. User-specific device-level parameters, lines, and other phone
buttons areupdated with user-specific settings.
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Implementing Cisco Unified Communications Manager, Part 2 (C1PT2) v8.0
) 2010 Cisco Systems, Inc
m
mm
Relationship of Cisco Extension Mobility Configuration
Elements
The figure shows how the Cisco Extension Mobility configuration elements relate toeacl
other.
Relationship of Cisco Extension
Mobility Configuration Elements
'••'tm&k
Device
Profile: A
—
Cisco Extension
MobilityPhone
Service
Stilt N—
\
\
Oevice
Profile: B
Device
Profile: C
Cisco Unified IP
Phone 7940
SCCPDef&ft
Device Profile
Cisco Unified IP
Phone 7940 SIP
Cisco Unified !P
OefeuS Device
SCCP Default
Device Proffle
Cisco Unified IP
Phone 7065 SIP
Default Device
Profits
As the figure shows, an end user isassociated with one ormore device profiles. For each
possible IP phone model and protocol (SCCP and SIP), adefault device profile can be
configured. Because Cisco Extension Mobility is implemented as aCisco IP Phone Seniee. all
phones that should support Cisco Extension Mobility must be subscribed to the Cisco
Extension Mobility phone service, loallow a user to log into the phone. In addition, each
mm
device profile must be subscribed to the Cisco Extension Mobility phone service; this
subscription isrequired to allow a user to logoutof a phone.
©2010 Cisco Systems, Inc.
Implementation ofFeatures andApplications for Multisite Deployments
4-49
Note
The default device profileis not applied ifa device profileof a user and the phone on which
the user tries to log in are of the same phone model series; for example, Cisco Unified IP
Phone 7960, 7961, or 7965.
Note
Cisco Unified Communications Manager automatically creates a default device profile for a
specific phone model and protocol, as soon as Cisco Extension Mobility is enabled on any
phone configuration page for this phone model.
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Cisco Extension Mobility Configuration Elements
Thistopic describes the configuration elements thatCisco Extension Mobility uses.
Cisco Extension Mobility Configuration
Elements
Configuration
Configuration Element Function
Element Name
SB res configuration of physicalphones. Conlguration parameters include device-
specificphone parameters (such as deviceCSS,location,or MRGL). user-specific
phoneparameters(such as userMOHaudiosource, DND, or soflkBy template),
and (oser-specifc) buttonconfiguration (suchas linesor speed dais).
The end user is associated with one or more device profiles The user IDand the
prj are used to log into a phone withCisco Enlension Mobility
Device profile
Stores user-specific phoneeoniguraflon in logical proSles. ContDuration
parametersinclude user-specific phone and button {lines, speed dials,etc.)
parameters.The parametersof the deviceprofle art appliedtoa physicalphone
ater a user logs intothe phone using Cisco Extension Mobility.
Phone service
Cisco ExtensionMobility is implementedas a phone service. Hardwarephones
and deuce prof les need to be subscribed to the serwce
Stores tie default device-configuration parameters thai should be appliedwhen
Default device profile
the phonemode* offie deviceprofileofa user isUrfferert than the phonemodelof
[he phone ^ wMW) fte usw |(jgs |(| ^ ^^ ^^ ^ ^ js^Av!f,ai ,s;ty
ci^ired lor«vetvphone mwlBlihaS hasCiKoExiensie'R MosilMy activated
The figure lists the configuration elements that are related toCisco Extension Mobility and
describes theirfunction. Theconfiguration elements that are introduced with Cisco Extension
Mobility are the device profile and the default device profile.
The device profile isconfigured with all the user-specific settings that are found althe device
level ofan IP phone (user MOH audio source, phone button templates, softkey templates, user
locales. DND and privacy settings, and phone service subscriptions) and all phone bulto.s
(lines, speed dials, and so on). One ormore device profiles are applied toan end user. ir> the
End User Configuration window.
The default dev ice profile stores default device configuration parameters thatCisco Extension
Mobilitv applies when there isa mismatch ofthe phone model series on which the user logs in
and the phone model series that is eonligured in the device profile ofthe user. The default
device profile exists once per phone model type and per protocol (Session Initiation Protocol
[SIP] and Skinny Client Control Protocol [SCCP]). All ofthe parameters that cannot be applied
from the device profile of the user are taken from the default device profile.
Eor example, a user is associated with adevice profile for aCisco Unified IP Phone 7945 that
runs SCCP. Ifthis user logs in toa Cisco Unified IP Phone 7965 that runs SIP. some features
(configuration parameters) that exist on the target phone are not configurable on the Cisco
Unified IP Phone 7945 dev ice profile. Inthis case, theconfiguration parameters that arc
unavailable onthe device profile of the user are taken from the default device profile ofthe
Cisco Unified IP Phone 7945 SCCP.
Ifadevice profile includes more parameters than the target phone supports, the additional
settings are ignored when the target phone with the user-specific settings is reconfigured.
© 2010 Cisco Systems. Inc.
Implementation ofFeatures and Applications for Multisite Deployments
4-47
Cisco Extension Mobility: Dynamic Phone Configuration by
Device Profiles
The figure illustrates how user-specific settings roam with the user when the user logs out of
one phone (for example, at the home location) and then logs into any other phone at any other
location.
Cisco Extension Mobility: Dynamic
Phone Configuration by Device Profiles
Cisco Unified
Communications
Remote
Manager
Gateway
Logout
Device Profile of
User Andy:
UMTUKda
Iter yOHAufe Sara
Ftcru tUscr, Timvlm
Login
Roaming User
Andy
H[
Device Profile of
Login User
and PIN
Unicss
NJ
User Andy:
UwrMOH AMID South
ftm* fcltwt'r«iwl*»
SetmTtmme
\limeas
As shown in the figure, the user-specific parameters (that is, some device-level parameters and
all phone button settings, including line configuration) are configured in device profiles. Based
on the user ID that is enteredduring login,Cisco Unilied Communications Manager can apply
the personaldevice profileof the user and can reconfigure the phone with the configuration
profile of the user who logs in.
With Cisco Extension Mobility, CiscoUnified Communications Manager is aware of the end
userof a device andapplies the appropriate user-specific configuration, according lo a device
profile that is associated with the logged-in user.
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Cisco Extension Mobility: Dynamic Phone Configuration
Parameters
Ihis subtopic describes the parameters that arc updated when a user logs in to a phone wnen
using Cisco Kxtension Mobility,
Cisco Extension Mobility: Dynamic
Phone Configuration Parameters
Cisco Extension Mobility can apply two types of phone
configuration parameters:
• User-specific, device-level parametersUser MOH audio source
• Phone button template
Softkey template
User locale
-- DND and privacy settings
Phone service subscriptions
• Other
• Complete configuration of all available phone buttons:
Includes lines (directory numbers) and feature buttons such
as speed dials, service URLs, Call Park, and others
(depending on phone model)
There arc two types of configuration parameters that aredynamically configured when Cisco
Extension Mobility is used:
•
User-specific de\ice-le\el parameters:
—
•
fhese user-specific phone configuration parameters include user Music on Hold
(MOH) audio source, phone button templates, softkey templates, user locales. Do
Not Disturb (DND) andprivacy settings, and phone service subscriptions. All these
parameters are configured at the device level of an II' phone.
Configuration of phone buttons(including lines):
—
©2010 Cisco Systems Inc
Cisco Extension Mobility updates all phone buttons -not only the button types that
are specified inthe phone button template but also the complete configuration of the
phone buttons. This update includes all configured lines, with all the line
configuration settings, speed dials, service UKUs. Call Park buttons, and any other
buttons that are configured in the device profile lhal is to be applied.
Implementation ofFeatures and Applications for Mullisite Deployments
4-45
Cisco Extension Mobility
Characteristics (Cont.)
At login, the phone configuration is updated with parameters
that are stored in the device profile of the user:
- Ifthe user ID is associated with multiple device profiles,
the user must choose the device profile that should be
used at login time.
- For duplicate logins, one of the following can be
configured:
• Allow multiple logins
• Deny login
• Auto-logout
At logout, another device profile (configured logout profile) is
applied, or the standard phone configuration is restored:
- Logout is performed manually by the user or enforced
after expiration of a configurable maximum login time.
If a user logs in with a user ID that is still logged in at another device, one of the following
options can be configured:
•
Allow multiple logins: When this method is configured, the user profile is applied to the
phone on which the user is logging in. The same configuration remains active at the device
on which the user logged in before. The line number or numbers become shared lines
because they are active on multiple devices.
•
Deny login: Whenthis optionis configured, the user receivesan error message. Login is
successful only after the user logs out of the other device on which the user logged in
before.
•
Auto-logout: Likethe preceding option,this optionensuresthat a user can be logged in at
only one device at a time. However, this option allowsthe new login by automatically
logging out the user of the other device.
On a phone that is configured for Cisco Extension Mobility, anotherdevice profile (a logout
device profile) canbe applied, or theparameters that are configured on the phone are applied.
The logoutcan be triggered by the user or enforced by the system after expiration of a
maximum login time.
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Cisco Extension Mobility Overview
This topic describes the keycharacteristics and features of Cisco Extension Mobility.
Cisco Extension Mobility Characteristics
- Allows users to log into any phone and have their user-specific
phone configurations applied to the phone
• Makes users reachable at their personal extensions,
regardless of their locations and the physical phones they use
« Is implemented as a phone service; works within a Cisco
Unified Communications Manager cluster
- Stores user-specific phone configuration in device profiles
Cisco Extension Mobility allows users to log in to any phone and have theirindividual, user-
specific phone configuration that isapplied tothat phone. Thus, users can be reached at their
personal directory number, regardless of their location or the physical phone that they are
using, Cisco Extension Mobility is implemented asa phone service and works within a Cisco
Unilied Communications Manager cluster.
The user-specific configuration is stored in device profiles. After successful login, the phone is
reconfigured with user-specific parameters; other (device-specific) parameters remain the same.
Ifa user is associated with multiple device profiles, the user must choose which device profile
to use.
© 2010 Cisco Systems, Inc.
Implementation ofFeaturesand Applications forMultisite Deployments
Cisco Extension Mobility Solves Issues of Roaming Users
Cisco Extension Mobility offers functionality that is designed to enhance the mobility of users
within a Cisco Unified Communications Manager cluster.
Cisco Extension Mobility Solves Issues
of Roaming Users
issue Without Cisco Extension
Cisco Extension Mobility Feaiure that-
Mobiiily
Solves the Issue
Extensions are bound to physical devices.
Extension* am bound b device profile*.
Speed dials are assigned lo physical devices. Spaed dials are assigned to device profiles.
Services are assigned tophysicai devices.
Services are assigned to device prolles.
MWIslaluss defined ror physical devrces.
MWI status is updated during Cisco
Extension
^ ^
..,.,... .
.,.=
.,.
•_
-
,j
-
casng privileges are delned for phytic*!
devices andlocations
oevicej
analocaons.
^
»*'*«••'**>* "™ mar9a *lln9
tttttoa.* Wevfce-based) Bfldptiyscal device
umg.s(locatiCKKef*«d).
The figure shows which issues Cisco Extension Mobility solves.
Although the device is not the homedevice of the user, it is reconfigured with user-specific
settings that are stored in profiles. This action allows the separation of user-specific parameters
(which are stored in profiles)from the device-specific parameters that are still stored in the
phone configuration (along with default values for user-specific settings). The phone willadapt
someof its behavior, according to the individual user who is using the phone.
A userlogin, in which the useris identified by user IDand PIN, triggers the configuration
changes. When the userstops using the phone, the userlogsout andthe default configuration is
reapplied. Thus, the phone configuration adapts to the user.
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Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) vB.O
) 2010 Cisco Systems, Inc.
Issues of Roaming Users
Using guest phones at remote sites leads to several issues.
Issues of Roaming Users
Ifa user uses a different phone in a different site:
* Extensions are traditionally bound to constant devices.
» User gets the wrong extension on that phone.
* User gets the wrong calling privileges.
• User has no speed dials available.
* User has wrong services assigned.
• MWI status does not work with different extension.
The figure lists the most common issues that arise when users use any available guest phone at
sitesto which they have traveled. These issues include wrong extension numbers andcalling
privileges, other speed-dial configuration and phone-sen.'ice assignments, and no MWI status
for the actual number of the user.
For correct settings, the user requires Cisco Unified Communications Manager to reconfigure
the used phone with user-specific configuration instead ofhaving device-specific settings that
are applied to the phone.
© 2010 Cisco Systems, Inc
Implementation otFeatures and Applications for Multisite Deployments
4-41
Issues with Users Roaming Between Sites
This topic describes the issues that can occur when users temporarily change their workplaces
and roam between sites.
Roaming Users
Roaming users do not travel with a device
(softphone) but use any available phone at the
current location.
Cisco Unified
Communications
Remote
Manager
Gateway
Roaming User
- - - - . : - _ . u—
Whenusers roam between sites and do not have their phone with them (for example, via Cisco
IP Communicator), they might want to use any available phone at the site to which they have
traveled.
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Lesson 2
Implementing Cisco Extension
Mobility
Overview
Some users roam between office desks or sites on a regular basis. Such users, who use phones
that are provided at the sites that they visit, would like to (but cannot) use their personal
settings, such as directory number, speed dials, calling privileges, and Message Waiting
Indicator (MWI). A professional Cisco Unified Communications solution needs to solve this
problem.
This lesson describes Cisco Extension Mobility, a feature of Cisco Unified Communicat oris
Manager. Cisco Extension Mobility allows Cisco Unified Communications Manager us..rs to
log into an IP phone and have their personal profile applied, regardless of the device and
physical location that they are using.
Objectives
Upon completing this lesson, you will be able to describe how Cisco Extension Mobility works
and how it is implemented. This ability includes being able to meet these objectives:
Identity the issues when users roam between sites
Describe the Cisco Extension Mobility feature
Describe the Cisco Extension Mobility configuration elements and their interaction
Describe Cisco Extension Mobility operation
Implement Cisco Extension Mobility
m
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Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
© 2010 Cisco Systems, Inc.
References
I'or additional infonnation. refer to these resources:
•
Cisco Systems. Inc. Cisco Unified Communications System 8.xSRND. April 2010.
hup: '.www.cisco.coni en US/doesAoice_ip comm/aicm/srn<l/8,\/uc8x.html
•
Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0(1). February 2010.
hup: 'www.cisco.com en US/docs/\oice ipcoin nv'cu em/admin/8 0 l/ccmcfg/beem-XOI
cm.html
)2010 Cisco Systems. Inc
Implementation of Features and Applications for Multisite Deployments
Summary
'ITiis topic summarizes the key points that were discussed in this lesson.
Summary
• Issues with roaming devices include inappropriate region,
location, time zone, and SRST reference configuration. PSTN
calls are using the home gateway instead of the local gateway
at the roaming site.
• Device Mobility allows roamingdevices to be identified by their
IP addresses, and configuration settings to be applied that are
suitable for the current physical location of the device.
• Device Mobility configuration elements are device mobility
groups, physical locations,device pools, and device mobility
infos.
Summary (Cont.)
• You apply roaming-sensitive settings to devices that roam
between physical locations. In addition, you apply Device
Mobility-related settings to devices that roam within the same
device mobility group.
1Implementation and operation of Device Mobility are optimized
when globalized call routing and local route groups are used.
Afterconfiguring device mobility groups, physical locations,
device pools, and device mobility infos, you must enable Device
Mobility eitherdusterwide as a service parameter or individually
per phone.
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Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0
52010 Cisco Systems, Inc.
Step 5b; Set the Device Mobility Mode for Individual Phones
The fig jrc shows how to set the Device Mobility mode for each phone.
Step 5b: Set the Device Mobility Mode
for Individual Phones
Cisco Unified Communications Manager Administration:
Device > Phone
EH£-
This is the home device pool of the
phone
^=^
Dev :e p™r
SL
n(j .|.|>
-
There are overlapping settings at the
device pool and at the phone; they
have higher prioritythan settings at
" " =•"
- s"'" s"»
-
the home device pool but are
replaced by settings from the roamng
device pool
Set the phone device mobilitymode:
the default applies the service
parameter setting
r-
H
.
In the I'honc Configuration window, you enable ordisable Device Mobility for each phoie by
cither setting Device Mobility Mode to On orOff or leaving the default value as Default. If
Device Mobilitv Mode is setto Default, the Device Mobility mode that is setat theCisco
CallManager serviceparameteris used.
The figure also shows the configuration ofthe overlapping parameters (these arc parameters
that can be configured at the phone and at the device pool). The overlapping parameters for
roaming-sensitive settings are Media Resource Group List, Location, and Network Locale. The
overlapping parameters for the Device Mobility-related settings are Calling Search Space
(called Device Mobility Calling Search Space atthe device pool), AAR Group, and AAR
Calling Search Space. Overlapping parameters that are configured at the phone have higner
prioritv than settings at the home device pool, and lower priority than settings at the roaming
device pool.
© 2010 Cisco Systems. Inc
Implementation ofFeatures and Applications for Multisite Deployments
4-35
Step 5a: Set the Device Mobility Mode Cisco CallManager
Service Parameter
The figure shows how to set the system default for the Device Mobility mode.
Step 5a: Set the Device Mobility Mode
Cisco CallManager Service Parameter
Cisco Unified Communications Manager Administration:
System >Service Parameter >Cisco CallManager
£( '.'.;-.i,;i[i;: *
off
u c«
Qt- -s!•;!f.. 1SJ* "
n„
,«|
^u^^,
A
y\
Set the default Device Mobility
mode for all phones.
Device Mobility is turned off by default and is configurable for eachphone. To set the default
for the Device Mobility mode (ifit isnot setdifferently atthe phone), choose System >
Service Parameter. ChooseCisco CallManager, and in the Clusterwide Parameters
(Device—Phone) section, set Device Mobility Modeto On or Off (OtTis the default).
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Step 4: Configure Device Mobility Infos
The figure shows how to configure device mobility infos.
Step 4; Configure Device
Cisco Unified Communications Manager Administration:
System >Device Mobility >Device Mobility Info
.
I"".. Mobil II, IMd li.f.rm.1".*
Enter name, subnet, and subnet |
mask.
..I. lnrH.ll w u l u Mobility Into-
Assign device mobilityinfo to
one or more device pools
w
Io configure device mobility infos, choose System > Device Mobility >Device MobilityInfo. They are configured with aname, a subnet, and a subnet mask. Then they are associated
with one or more device pools.
m
© 20'0 Cisco Systems. Inc.
Implementation of Features and Applications for Multisite Deployments
4-33
Step 3: Configure Device Pools
The figure shows how to configure adevice pool when using Device Mobility.
Step 3: Configure Device Pools
Cisco Unified
Communications
Manager Ad ministration:
System > Device Pool
ConGguteroamingsensitive sellings mat <
be applied lo Ihe ptione
configuration The Local
Route QOLp ib also a
roaming -sens itiveset! ing'
Choose physical location
ana device mobilitygroupto
determine vi»>ettier tc apply
roa m mg-sens ilive s etlings
and Device Mobility-related
settings
Configure Dev.ce Mobilityrelated sett ngs The Called
Party Transtormaton CSS
is no Device Mobility-related
selling1
You configure a device pool with aname and a Cisco Unified Communications Manager
group. The configuration includes roaming-sensitive settings and Device Mobility-related
settings. (You configure the Device Mobility-related settings in the Device Mobility-Related
Information section.) You configure both the physical location and the device mobility group in
the Roaming-Sensitive Settings section. You use both of those configurations todecide which
settings to apply to a phone: nosettings, the roaming-sensitive settings only, orthe roamingsensitive settings and the Device Mobility-related settings. Thephysical location and the device
mobility group themselves arenotapplied to the configuration of a phone, butareused to
control which settings to apply.
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Steps 1 and 2: Configure Physical Locations and Device
Mobility Groups
fhe figure shows the configuration of physical locations and device mobility groups.
Steps 1 and 2: Configure Physical
Locations and Device Mobility Groups
Cisco Unified Communications Manager Administration:
System > Physical Location
••-
•
•
Entername arid description. L^_ "'""' "'>-"'
Cisco Unified Communications Manager Administration:
System > Device Mobility > Device Mobility Group
I1»IIU« HiUUh. I:—..,. 1-h.Pm.M—
Entername and description <~
To configure physical locations, choose System > Physical Location, for each physical
location, you configure a name and a description. To configure device mobility groups, choose
System > Device Mobility > Device Mobility Group. For each device mobility group, you
configure a name and a description.
Note
Device mobility groups are not necessary when there is no need to change the device level
CSS, AAR CSS, and AARgroup This principle applies also when local route groups are
used in an environment where all sites share the same dial rules or in an environment where
globalized call routing is implemented
© 2010 Cisco Systems, Inc
Implementation of Features and Applications for Multisite Deployments
Device Mobility Configuration
This topic describes how to configure Device Mobility.
Device Mobility Configuration Steps
1
Configure physical locations.
2
Configure device mobility group.
3
Configure device pools.
4
Configure device mobility infos (IP subnets).
5
Set the Device Mobility mode by using:
a.
A Cisco CallManager service parameter to set the
default for all phones
b
The Phone Configuration window for individual
configuration per phone
"Die figure lists the required steps for implementing Device Mobility. As discussed in the
previous topics, device mobility groups are not required when you implement Device Mobility
in an environment where globalized call routing is used.
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Example: Globalized Call Routing
The figure shows an example of Device Mobility in an environment where globalized call
routing is implemented. Also, gateway selection is performed by the local route group feature.
Example: Globalized Cail Routing
HQ Translation Patterns
(Partition HQ, CSS. System:
000 ' -> DDI PreDot, Prefix +
00 i -> DDI PreDot. Prefix +49
DDI PreDot, Prefix +4940
Device Pool
HQ
Physical
Location HQ
Local Route
Group >~'j
Route Pattern \+'
Partition System
Single Route List.
Default Local Roule Group
Device Pool. E
Physical
BR Translation Patterns
(Partition BR. CSS. System|
91 [2-9)XX[2-9JXXXXXX -> DDI
PreDct, Prefix+1
Location BR
CSS BR]
Local Route
Group. BR
9 [2-9]XXXXXX -> DDI PreDot,
Prefix+1408
9011 !-> DDI PreDot. Prefix +
Normalization of localized call ingress is in place
BR user roaming to HQ uses NANPdial rules and uses HQ gateway
Theexample in the figure is based on the previous scenario: IIt) is in Europe. BRis in the
United States. A BR user will roam to Europe.
However, in this example, globalized call routing has been implemented. Therefore, the (line)
m
CSS of BR phones provides access to translation patterns thatconvert localized call ingress at
the phone (NANP formal) to global E.I64 formal. EU phones have access to translation
patterns that cotwert HI] input to global K.164 fonnat.
A single PSTN route pattern (\(!) is configured: it is in a partition that is accessible by all
translation patterns.
When a BR user roamsto the IIQ. the lineCSS is not modified; no device CSS is configured at
the phone or at the device pool. Thedevice mobility groups arc alsonotset (or areset
differently).
As a result, there is effectively no change in matching the translation patterns: The BRuserstill
uses NANPdial rules (like at home), fhe numberis converted to international format by-
translation patterns and matches the (only) PSTN roule pattern. The route pattern refers to a
route listthat is configured to use thedefault local route group. The default local route group is
taken from the roaming device pool. Therefore, if the phone is physically located in the BR
office, the local route group is BR; if the phone is roaming to the HQ site, the local route group
is HQ. As a result, the local gateway is always used for a PSTN call.
IfTEHO was configured, there would be aTEHO route pattern in E.I64 format with a leading
+sign. The TEHO pattern would refer toa site-specific route list in order toselect the correct
gateway for PSTN egress. The backup gateway would then again be selected by the local route
group feature.
) 2010 Cisco Systems. Inc
Implementation of Featuresand Applications for Multisite Deployments
4-29
Example: No Globalized Call Routing—Same Device Mobility
Group
The figure shows an example of Device Mobility with identical device mobilitv groups in an
environment where globalized call routing is not implemented. Also, gateway selection is
performed by the device CSS ofthe IP phone.
Example: No Globalized Call RoutingSame Device Mobility Group
Device Pool HQ
Physical
Location. HQ
Device Mobility
Group; World
Device Mobility
CSS' HQ
| BR user roaming to HQ has lo use EU dial rules and uses HQ gateway [
This example is identical to the previous example with one exception: This time the device
mobility group ofthe home and the roaming device pool are the same.
When a BR user roams to the HQ, the device CSS ofthe phone is updated with the device CSS
of the roaming device pool. In the example, CSS BR is changed to HQ. As aconsequence the
phone has access to the HQ partition that includes PSTN route patterns in EU dialing format
Therefore, the roaming user has to follow EU dial rules. Calls to 9.@ are not possible anymore
However, this configuration allows the BR user to use the HQ gateway when roaming to the
HQ.
•fe
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^2010 Cisco Systems, Inc.
Example: No Globalized Call Routing—Different Device
Mobility Group
The figure shows an example of Device Mobility with dilTerenl device mobility groups in an
em ironment where globalized call routing is not implemented. Also, gateway selection is
performed b\ the device CSS ofthe IP phone.
Example: No Globalized Call RoutingDifferent Device Mobility Group
Device Pool HQ
Route Pattern
«49 40 552XXX
Physical
01
Location i C
Device Mobility
Group HQ
Route List
(Device Mobility
Route Group HQ-GW
CSS. HQ)
Device Pool BR
Route Pattern
Physical
9®
Location1 Lit*
Partition Branch
Device Mobility
Group. BR
(Device Mobility
CSS. BR)
Route List
Roule Group. BR-GW
HQ: EU Numbering
Plan
UR. NANP
| BR user roaming lo HQ uses NANP dial rules and uses BR gateway |
In the example, there are two sites. The main site ("I IQ" in the figure) is in Europe, the branch
site ("BR"' in the figure) is in the United States. Separate route patterns (representing the
different dial rules) are configured in different partitions. The CSS of HQ phones provides
access to the HQ gateway, the CSS ofBR phones provides access to the RR gateway.
Device Mobilitv is configured with different device mobility groups. This configuration allows
BR users who arc roaming with their phones to the HQ to use the home d.al rules. Ihe device
CSS is not updated by Device Mobility, and therefore, the CSS still provides access to the BR
route pattern (9,ff). Howler, as aconsequence, the BR gateway ,s used tor all PS fN calls.
© 2010 Cisco Systems. Inc.
Implementation of Features and Applications for Multisite Deployments
Advantages of Using Local Route Groups and Globalized Call
Routing
Device Mobility benefits from globalized call routing and local route groups, especially when
implemented in international environments.
Advantages of Local Route Groups and
Globalized Call Routing
Cisco Unified Device Mobility benefits from globalized call
routing and local route groups:
• Updates ofroaming sensitive settings still apply {no changes)
- Local route group is a roaming sensitive setting
• Globalized call routing and local route groups allowsthe
combination of two features, which are exlusive otherwise:
~ Roaming gateway can be used
- Home dial rules can be used
• No updatesofdevice mobility-related settings (device CSS, AAR
CSS, and AAR group) required
• Functionality ofdevice level CSS(device/line approach) is
replaced by local route group in route list
• Different AAR groups andAAR CSS not required
• Eliminates need fordevice mobility groups
When Device Mobility with globalized call routing is used, there are no changes in the
roaming-sensitive settings. Their application always makes sense when roaming between sites
Iliey have no influence on the gateway selection and the dial rules that auser has to follow.
The dial plan-related part of Device Mobility, however, changes substantially with globalized
call routing. It allows aroaming user to follow the home dial rules for external calls and
nevertheless utilize the local gateway ofthe roaming site.
This situation is possible because globalization of localized call ingress at the phone occurs
1his function is provided by the line CSS of the phone. It provides access to phone-specific
translation patterns that normalize the localized input ofthe user to global format. The device
CSS that was used for gateway selection is obsolete, because gateway selection is now
performed by the local route group feature.
The AAR CSS and AAR group that are configured at the device level can be the same for all
phones as long as the AAR number is always in global format. (You can ensure that it is always
in global format by configuring either the external phone number mask orthe AAR
transformation mask to E. 164 format.) In this case, no different AAR groups arc required
because there is no need for different prefixes that are based on the location of the two phones
Further, there is no need for different AAR CSS, because the gateway selection is not based on
d.fferent route lists (referenced from different route patterns in different partitions) Instead it
is based on the local route group that was configured at the device pool of the calling phone.
In summary, when using globalized call routing is used, Device Mobility allows users lo use
local gateways at roaming sites for PSTN access (or for backup when TEHO is configured)
while utilizing their home dial rules. There is no need to apply different device CSS AAR CSS
and AAR groups, and hence, device mobility groups are no longer required
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
32010 Cisco Systems, Inc.
Note
If TEHO isused, there areno suboptimal paths when using Device Mobility with different
device mobility groups. However, when local route groups and globalized call routing arenot
being used, TEHO implementation can be very complex, especially when local PSTN
backup is desired and when TEHO is implemented in international deployments.
In summarv. unless TEHO is used, the implementation ofDevice Mobility without globalized
call routing leads to this situation: Either the home gateway has to be used (when allowing the
user touse the home dial rules) orthe user is forced touse the dial rules ofthe roaming site (in
orderto use the local gateway of theroaming site).
m*
>2010 Cisco Systems, Inc
implementation otFeatures and Applications for Multisite Deployments
4-25
Device Mobility Interaction with Globalized Call
Routing
This topic describes the interactions of globalized call routing and local route groups with
Cisco Unified Device Mobilitv.
Device Mobility Review Without Local
Route Groups and Globalized Call Routing
As discussed earlier, there are two types of device pool settings
for Device Mobility:
• Roaming-sensitive settings (location, region, SRST reference, MRGL, etc.)
are always updated when device roams to different physical location.
• Device Motility-related settings (device CSS, AAR CSS, and AAR group) are
updated based on device mobility group:
-
Different device mobility group:
• Noupdateof devceCSS.AARCSS.artd AARgroup
• No changes to dial odes, but use of home gateway
- Same device mobility group:
• UpdateofdeviceCSS.AARCSS.andAARgroup
• Use of roaming gateway but changes to dial rules
Required only when gateway selection is based on CSS (that is, local
mute group and globalzed cal routing are not used)
Local route groups have been introduced with Cisco Unified Communications Manager
Version 7. When local roule groups—and globalized call routing, which utilizes local route
groups—is not used or supported. Device Mobility is typically implemented:
•
Roaming-sensitive settings are always updated when the device roams between different
physical locations. These settings are location, region, SRST reference, MRGL, and other
parameters that do not affect the selection of the PSI'N gateway or the local rules.
•
Device Mobility-related settings can be appliedin addition to the roaming-sensitive
settings (which meansthat a phone has to roam between differentphysical locations). The
Device Mobility-related settings are device CSS, AAR CSS, and AAR group. The
configuration of the device mobility group shoulddetermine your decision about whether
to apph the Device Mobility-related settings.
—
If the device roams between different device mobility groups, the Device Mobilityrelated settings arenot updated with the values that were configured at the roaming
de\ ice pool. This configuration hasthe advantage thatusers do not have to adaptto
different dial rulesbetween homeand roaminglocation (if they exist).The
disadvantage is that all PS'fN calls will use the homegateway that can leadto
suboptima! routing.
—
Ifdie device roams within the same device mobility group, the Device Mobilityrelated settings areupdated with the values of the roaming device pool. This
configuration has the advantage that all PSTN calls will use the local (roaming)
gateway, which is typically desired for roamingusers. However, the users will have
to use to the local dial rules.
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For the next three scenarios, the assumption is that the home device pool and the roaming
device pool are assigned to different DMGs. As a result, the Device Mobility-related settings
are not applied. Therefore, calls that are placed from the roaming device are routed in the same
way as when the device is in its home location.
•
If the user places a call to a UK PS'fN destination, the call will use the IP WAN to the
London site and break out to the PSTN at the London gateway with a local or national call.
This solution is the optimal one from a toll perspective.
•
If the user places a call to a PSTN destination that is close to the roaming location (for
example, a U.S. PSTN number) and TEHO is not configured, the call will use the IP WAN
from the U.S. office to the I .ondon site. It wilt break out to the PSTN at the London
gateway to place an international call back to the United States. From a toll perspeebve,
this is the worst possible solution, because the call first goes from the United States to
London over the IP WAN (consuming bandwidth) and then goes back from London to the
United States via a costly international call.
•
If the user places a call to a PSTN destination that is close lo the roaming location (for
example, a U.S. PSTN number) and TEHO is configured, the U.S. gateway is used for a
local or national call. This event is optimal from a loll perspective.
Note
In these three examples, the user has to dial PSTN destinations by following the dial rules of
the home location (Great Britain).
In summary, when you allow the Device Mobility-related settings to be applied (by using the
same device mobility group), calls to the home location will use a local PS'fN gateway to place
a long-distance or international call when not implementing TEI10.All othercalls areoptimal.
When the Device Mobility-related settings arc not applied (by using different device mobility
groups) and TEHO is not used, calls to the roaming location will first usethe IP WAN to go
from the roaming location to the home location, fhe calls then use the home gateway to place a
long-distance or international call back to the roaming location. All othercalls areoptimal.
The discussed scenarios assume that globalized call routing and local routegroupsare not used,
fhe impact of globalized call routing and local route groups is discussed inthe next topic.
©2010 Cisco Systems, Inc.
Implementation ofFeatures and Applications for Multisite Deployments
4-23
Examples of Call-Routing Paths Based on Device Mobility
Groups and TEHO
The table shows how calls are routed in different Device Mobility scenarios.
Examples of Different Call-Routing Paths
Based on Device Mobility Groups and TEHO
Same device mobity group, call lo PSTN
Cal use* local PSTN gateway at roaming
destination close id dome location, no TEHO.
location for a long-distance PSTN cal.
Same device mobility group, call lo PSTN
Cal uses IP WAN to gateway at home location
destination close D home location, TEHO.
for a local PSTN call.
Same device mobity group, call to PSTN
Cal uses local PSTN gateway at roaming
destinalIon close b roamhg location.
location for a local PSTN call.
Different device motility group, call to PSTN
Cal uses IP WAN to gateway at dome local on
destination close lo home location.
for a local PSTN call.
Different device mobility group, call lo PSTN
destination close Id roaming location, no TEHO.
Cal uses IP WAN to gateway at home location
Different 0evice mobility group, call lo PSTN
CaB uses local PSTN gateway at roaming
destination close to roammg location, TEHO.
location for a local PSTN call.
tor a king-a stance PSTN caK.
Calls are routed differently depending on the configuration of DMGs (whether Device
Mobility-relatedsettings are applied or not), the dialed destination, and the use of tail-end hopoff (TEHO). In some scenarios, calls might take suboptimal paths.
The example in this discussion assumes that a user from London roams to the U.S. office (for
simplicity, it is assumed that there is only one U.S. office). This user uses Cisco IP
Communicator.
Forthe first three scenarios, the assumption is that the home devicepool and the roaming
devicepool are assigned to the same device mobility group.Therefore, Device Mobility applies
Device Mobility-related settings. As a result, PSTN calls that are placed from the roaming
device are treated like PSTN calls of standard U.S. phones.
•
If the user places a call to a PSTN destination that is close to the home location (for
example, a UK PSTN number) and TEHO is not configured, the call will use the local
(U.S.) PSTN gateway for an international PSTN call, from a toll perspective, this is a
suboptimal solution, because the IP WAN is nol used as much as it could be used when
implementing TEHO. This factor applies not only to the roaming user, but also to U.S.
users who place calls to PSTN destinations in Great Britain.
•
If the user placesthe same call (to a UK PSTN number) and TEHO is configured, the call
will use the IP WAN to the London site and breakout to the PSTN at the London gateway
with a local call. This solution is the optimal one from a toll perspective.
•
If the user places a call to a U.S. destination number, the U.S. gateway is used fora local or
national call. This event is optimal from a toll perspective.
Note
In all ofthe examples that are shown inthe table, the user has to dial PSTN destinations by
following the U.S. dial rules (North American Numbering Plan [NANP]).
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Implementing Cisco Unified Communicalions Manager, Part 2 (CIPT2) v8.0
© 2010 Cisco Systems, Inc.
The line/device approach is recommended when you are implementing CoS in a multisite
environment. Here is a description of the operation of the line/device approach:
•
The line CSS implements CoS configuration by permitting internal destinations (other
phone director; numbers, and access to features such as call part and Meet-Me
conferences). The line CSS also blocks PSTN destinations. Because the line CSS is not
changed by Device Mobility. CoS settingsof the device are kept when the device is
roaming.
•
I he de\ice CSS is modified when the device is roaming within the same device mobility
group. In this case, the device CSS that is used at the home location is replaced by a de\ ice
CSS that is applicable for the roaming location. This device CSS will refer to the local
gateway of the roaming site instead of to the gateway that is used at the home location.
If the traditional approach is used (only one CSS. combining CoS and gateway choice), the
device CSS must be used. The reason is that Device Mobility cannot modify the line CSS. and
the line CSS has priority over the device CSS (which can be modified by Device Mobility).
The AAR CSS is configurable only at the device level and therefore is always correctly
replaced when the device roams between physical locations within the same device mobility
group.
Note
When using globalizedcall routing and local route groups, there is no need for site-specific
device-level CSS. More information about the interaction of globalized call routing and
Device Mobility is provided in a later topic of this lesson.
>2010 Cisco Systems. Inc
Implementation of Features and Applications for Multisite Deployments
4-21
Device Mobility and Calling Search Spaces
This subtopic describes how CSSs are processed when Device Mobility is used.
Device Mobility and Calling Search
Spaces
• Line CSS is nevermodified by Device Mobility.
• Device CSS is modified only when device roams between
physical locations and within the samedevice mobility group:
- Operation when using the line/device CSS approach
(recommended forCoS implementation);
• Line CSS is not modified; CoS settings are kept.
• Device CSS is modified; itallows local gateway
selection by applying CSS ofroaming device pool.
- When using the traditional CSS approach (only one CSS
at phone), use the device CSS, instead ofthe line CSS, to
allow the CSSto be modified by Device Mobility.
• AAR CSS is configurableonly at the device and therefore is
always correctly modified when the device roams between
physical locations and within thesamedevice mobility group.
Review of Line and Device CSS
An IP phone can be configured with aline CSS and adevice CSS. Ifboth CSSs exist, the
partitions ofthe line CSS are considered before the partitions ofthe device CSS when a call is
being routed.
fhese two CSSs allow the use of the line/device approach for implementing calling privileges
and the choice ofalocal gateway for PSTN calls. With the line/device approach, all possible
PS INroute patterns exist once per location and are configured with asite-specific partition
This partition is included in the device CSS ofthe phones, and therefore itenables the use ofa
local gateway for PSTN calls. To implement class ofservice (CoS), PSTN route patterns that
should not be available to all users (for example, international calls, long-distance calls or all
toll calls) are configured as blocked route patterns. These blocked route patterns arc then
assigned to separate partitions. The line CSS ofaphone now will include Ihe partitions of those
route patterns that should be blocked for this phone. Because the line CSS has priority over the
device CSS. the blocked pattern will take precedence over the routed pattern that is found in a
partition that is listed at the device CSS.
Device Mobility and CSSs
Device Mobility never modifies the line CSS ofaphone. It does change the device CSS and
AAR CSS of aphone when the phone is roaming between different physical locations within
the same device mobility group.
4-20
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) u8.0
©2010 Cisco Systems, Inc.
Gemian users who roam with their softphoncs tothe United Stales might beconfused when
they obtain their home extensions, but they have to use U.S. dialing rules (access code 9instead
of6. or011 instead of00 for international numbers, and so on). Ifyou want to avoid such
behavior, vou need to suppress the application of Device Mobility-related settings. You
suppress the settings by assigning device pools that are to be used at sites with different dialing
rules into different device mobility groups (and in different physical locations). When a user
now roams with a device from Germany tothe United States, all the roaming-sensitive settings
are applied to use local media resources and Cisco Unified SRST gateways. Also, codecs and
CAC settings are applied correctly, but the Device Mobility-related settings arc not applied.
The result is that the phone will use the PSTN gateway and dial rules ofits home location even
though the user has moved to another site. The user does not have to adapt to the dial rules of
the local site to which the phone was moved.
Note
The preceding statements regarding call routing and dial behavior that are based on Device
Mobility-related settings do not apply when globalized call routing is used. Alater topic in
this lesson presents more information about the interaction of globalized call routing and
Device Mobility
Ho10 Cisco Systems. Inc.
Implementation of Features and Applications for Multisite Deployments
4-19
Device Mobility Considerations
This subtopic describes key facts that you need to consider when implementing Device
Mobility.
Device Mobility Considerations
• Roaming-sensitive settings:
- Ensure the use of local media resources and SRST references.
-
Ensure correct use of codecs and CAC between sites.
- Shouldalways be applied to roaming devices
• Settings relatedto Device Mobility affect call routing:
- WhatgatewaytouseforPSTNaccessandAAR PSTN calls(device CSS
andAAR CSS),and how to composethe AAR number(AAR group)?
- Changes may result indifferent dialing behavior (forexample, different
PSTN accesscodes, different PSTN numbering plans, and so on).
- Users might getconfused byhaving their home extensions and yet being
required to follow dial rules of roaming site.
- tfthisis notdesired, suppress application of settings related to Device
Mobility by assigningdifferent device mobility groups.
- Not applying settings related to Device Mobility might lead to suboptimal
call-routng paths.
Roaming-sensitive settings ensure that local media resources and SRST references are used by
the roaming device. Inaddition, they ensure the correct use ofcodecs and CAC between sites.
Typically, this is always desired when adevice roams between different sites. It is not required
when the device moves only between IPsubnets within the same site. Therefore, the
recommendation isto assign all device pools that are associated with IPsubnets (device
mobility info) that are used atthe same site tothe same physical location. This action results in
phone configuration changes only when the phone roams between sites (physical locations) and
notina situation where a phone is moved only between different networks of thesame site.
Device Mobility-related settings affect call routing. By the application ofthe device CSS. AAR
group, and AAR. CSS calls can be routed differently depending on the site where the phone has
roamed to. The settings at the roaming device pool determine which gateway will be used for
public switched telephone network (PSTN) access and AAR PSTN calls (based on the device
CSS and AAR CSS) and how the number to be used for AAR calls is composed (based on the
AAR group).
Such changes can result in different dialing behavior. For instance, when roaming between
different countries, the PSTN access code might be different and PSTN numbering plans (for
example, how to dial international calls) might apply. As an example, in order to dial the
Austrian destination +43 699 18900009 from Germany, users dial 0.0043 699 18900009. while
users in the United States have to dial 9.01143 699 18900009.
4-18
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
© 2010 Cisco Systems, Inc.
7. The roaming-sensitive settings ofthe chosen device pool (that is. the roaming device pool)
arcused to update the configuration of thephone.
Note
In this case, overlapping settings (that is,settings thatexist at the phone as well as at the
device pool, namely, MRGL, Location, and Network Locale) ofthe roaming device pool have
priority over the corresponding settings atthe phone. This behavior isdifferent from the
default behavior (see Step 10).
8. Ifthe dev ice mobility groups ofthe chosen device pool and ihe home device pool are the
same, the phone configuration isupdated by applying the Device Mobility-related settings:
otherwise continue.
Note
In this case, all settings are overlapping settings (that is, all Device Mobility-related settings
exist atthe phone as well as at the device pool), and the parameters of the roaming device
pool have priority over the corresponding settings atthe phone. This behavior isdifferent
from the default behavior (see Step 10).
9. Where the phone configuration has been updated (either with the roaming-sensitive settings
onh orwith the roaming-sensitive settings and the Device Mobility-relalcd settings), the
phone is reset in order for the updated configuration to be applied to the phone.
Caution
This is the end of the process: do not continue to Step 10. Step 10 was directly referenced
from previous steps in certain conditions and does not apply after Step 9.
10. Here is adescription ofthe default behavior. First, the settings ofthe home device pool
(that is. the device pool that is configured at the phone) are applied. Some configuration
parameters of the device pool can also be set individually at the phone. These overlapping
phone configuration parameters are MRGL Location. Network Locale. Device Mobility
CSS (which is called simply CSS at the phone). AAR CSS, and AAR Group. Ifthese
parameters are configured at the phone (that is. are not set lo [None]), the phone
configuration settings have priority o\er the corresponding setting at the device pool.
m*
m
^D10 Cisco Systems. Inc
implementation of Features and Applications for Mulfsite Deployments
4-17
Device Mobility Operation: Flowchart
The figure illustrates Device Mobility operation in a flowchart.
Device Mobility Operation: Flowchart
I No
hi
^m
I
Apply roamtngssnsttiua sellings
from selected device
pool.
Slop- Apply DeviceMobiifty-related
settings from
| DP - device pool. DWG =device mobility group, DMI - flevice mobility info |
Here is how a phone registers with Device Mobility:
1. Adevice attempts to register with Cisco Unified Communications Manager.
2. Cisco Unified Communications Manager checks whether Device Mobility is enabled for
the device.
—
Ifit is not enabled for the device, the default behavior applies (go to Step 10);
otherwise continue.
3. Cisco Unified Communications Manager checks whether the IP address ofthe IP phone is
found inone of the device mobility groups.
—
Ifit is not found, the default behavior applies (go toStep 10); otherwise continue.
4. The device pool to be used is chosen.
—
Ifthe home device pool isassociated with the device mobility info in which the IP
address of the phone was found, the home device pool ischosen.
—
Ifthe home device poo! is not associated wilh the device mobility info in which the
IP address ofthe phone was found, the device pool ischosen based on a loadsharing algorithm (ifmore than one device pool is associated with the device
mobility info).
5. Ifthe chosen device pool is the home device pool, the default behavior applies (go to
Step 10); otherwise continue.
6. Ifthe physical locations of the chosen device pool and the home device pool are the same
the default behavior applies (goto Step 10); otherwise continue.
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
)2010 Cisco Systems, Inc.
In summary, the roaming-sensitive parameters are applied when the physical location ofthe
current device pool is different from the physical location ofthe home device pool (thai .s.
when roaming between physical locations). The Device Mobility-related settings are applied in
addition to the roaming-sensitive parameters when the physical locations are different and the
dev ice mobility groups are the same (that is, when roaming between physical locations within
the same device mobility group).
As a result, physical locations and device mobility groups should be used in two ways:
• Physical locations: Configure physical locations in such away that codec choice and CAC
truh reflect thecurrent location of the device. Hnsurc thatlocal SRST references and local
media resources atthe roaming site are used instead ofSRST references and media sources
that are located atthe (currently remote) home network. Depending upon the network
structure andallocation of services, vou may define physical locations that arebased upon
a cit\. an enterprise campus, or a building.
• Device mobility groups: Adevice mobility group should define agroup ofsites with
similar dialing patterns or dialing behavior. Device mobility groups represent the highestlev el geographic entities in your network. Depending upon ihe network size and scops,
your device mobilitv groups could represent countries, regions, states or provinces, c.ties,
or other entities. Because Device Mobility-related settings (which areapplied only v hen
roaming within the same device mobility group) affect call routing, different device
mobilitv groups should be set up whenever roaming users should not be forced to adapt
their dialing behavior. In this case, as in roaming between different device mobility groups,
the phone configuration parameters that affect call routing (that is. the Device Mobilityrelated settings) arc not modified.
Note
When using globalized call routing and local route groups, device mobility groups aru
irrelevant Thereason isthatthere is no need tochange thedevice-level CSS, theAAR
CSS, and the device-level AAR group. More information about the interaction of globalized
call routing and Device Mobility isprovided in a later topic of this lesson.
**
~ 10 C|sco Systems, mc
implementation of Features and Applications for Muttisite Deployments
4-15
Device Mobility Operation
This topic describes how Device Mobility works.
When and How Is the Phone
Configuration Modified?
1 Each phone is configured with a device pool (that is, the home
device pool).
IP subnets are associated with device pools,
Ifthe IP address of the phone matches a configured IP subnet
one ofthe associated device pools is selected (load-shared).
Ifthe selected device pool is different from the home device pool
of the device, these settings of the two device pools are
checked:
- If the physicallocationsare not different the phone
configuration is not modified.
- If the physical locationsare different, the roaming-sensitive
settings of the roaming device pool are applied.
- Additionally, ifthe device mobility groups are not different
the Device Mobility-related settings of the roaming device
pool are also applied.
In all other cases, the home device pool configuration is applied.
As discussed earlier, each phone isconfigured with a device pool. This device pool isthe home
device pool of the phone.
IP subnets are associated with device pools (by configuring device mobility infos).
Ifa phone for which Device Mobility isenabled registers with Cisco Unified Communications
Manager and has an IP address that matches an IP subnet that is configured in a device mobility
info, these actions occur:
•
The currentdevice pool is chosen.
—
Ifthe device mobility info is associated with the home device pool ofthe phone, the
phone is considered to be in its home location; Device Mobility will not reconfigure
the phone.
—
Ifthe device mobility info isassociated with one ormore device pools other than the
home device pool ofthe phone, one ofthe associated device pools ischosen based
on a load-sharing algorithm (round robin).
• Ifthe current device pool is different from the home device pool, these checks arc
performed:
—
Ifthe physical locations are not different, the configuration ofthe phone is not
modified.
—
Ifthe physical locations are different, the roaming-sensilive parameters ofthe
current (that is. the roaming) device pool are applied.
—
If(in addition to different physical locations) the device mobility groups are the
same, the Device Mobility-related settings are also applied (in addition to the
roaming-sensitive parameters).
Implementing Cisco Unified Communications Manager. Part 2(CIPT2) v8.0
>2010 Cisco Systems. Inc.
In summarv. the U.S. device mobility group consists oftwo physical locations: San Jose and
New York.'ln San Jose. IP subnets 10.1.1.0/24. 10.1.2.0/24. and 10.1.3.0/24 arc used: New
York uses IP subnet 10.3.1.0/24. and London is configured with IP subnet 10.10.1.0/24.
Basedon the IP address of an IP phone. Cisco Unified Communications Managercan
determine oneor more associated device pool or pools and the physical location anddevice
mobility group ofthe device pool orpools. Ifan IP phone uses an IP address ofIP subnet
10.1.3.6/24. there are two candidates for the device pool. However, inthis example, the
physical location and the device mobility group are the same for these two device pools.
>201D Cisco Systems, Inc.
Implementation of Features and Applications for Multisite Deployments
4-13
Relationship of Device Mobility Configuration Elements
The figure illustrates how the Device Mobility configuration elements relate to each other.
Relationship of Device Mobility
Configuration Elements
1
SJ1 dmi
SJ_A_ dp
(10 1.1 0/24)
(Building A)
c--.^
i
SJ2_dmi
SJ_B1_ dp
!
(10.1.2.0/24)
_.*!
^ ^
/
i
i
i
SJ3 dmi
(10 1.3.0/24)
s
sU
(Buiding B)
\
SJ_pl
j
i (SJ-campus) I
Jr
"
•
N
,4\ US dmg !
~i s,,\
1
SJ_B2_ dp
(Building B)
* **
i
*
NY dmi
i
(10.3 1.0/24)
mv
NY_dp
r
'
j (NY-campus) j
i
W.
LON dmi
(10.10.1.0/24)
.
.>
_
•.
_
_
LON_pl
—•
LON_ dp
_
j
i GB_dmg , i
(LON-campus) j
—
Theexample in the figure shows five device mobility infos. They are configured as follows:
•
SJl_dmi: The IP subnet of this device mobility info is 10.1.1.0/24. This device mobility
info is used at Building A of the San Jose campus and is associated with device pool
SJ_A_dp.
•
SJ2_dmi: The IPsubnet of this device mobility info is 10.1.2.0/24. This device mobilityinfo is used at Building Bof the San Jose campus and is associated with device pool
SJ_Bl^dp.
•
SJ3_dmi: The IPsubnet of this device mobility info is 10.1.3.0/24. Like SJ2_dmi, this
device mobility info isused at Building Band isassociated with device pool SJ_Bl_dp. It
is also associated with devicepool SJ_B2_dp.
•
•
NY_dmi: The IPsubnet ofthis device mobility info is 10.3.1.0/24. This device mobility
info is used at the New York campus and is associated with device pool NY _dp.
LON_dmi: The IPsubnet of this device mobility info is 10.10.1.0/24. This device mobility
info is used atthe London campus and is associated with device pool LON_dp.
Device pools SJ_A_dp. SJ_B1 jjp. and SJ_B2_dp are configured with the same physical
location (SJ_pl) because they are used for devices that are located at the San Jose campus.
Device pool NY_dp. serving the New York campus, is configured with physical location
NY_pl. and device pool LON_dp, serving the London campus, isconfigured with physical
location LON_pI.
All device pools that are assigned with a U.S. physical location (that is, SJ_A_dp, SJ_Bl_dp,
SJ_B2_dp, and NYjJp) are configured with device mobility group US_dmg. This setting
means that all U.S. device pools are in the same device mobility group. The London campus is
in a different device mobility group: GB_dmg.
4-12
Implementing Cisco Unrfied Communications Manager, Part 2 (CIPT2) v8.0
»2010 Cisco Systems, Inc.
Device Mobility Configuration Elements
This topic describes the configuration elements that arc used by Device Mobility.
Configuration Element Functions
Configuration
Configuration Element Functon
Element Name
Defines a set of common characteristics for devices. The device
poolcontans onlydevice-and location-related information. One
device pool has to be assigned to each device.
Deuce MoNiy Info
Specifies an IP subnet and associates it withone or more device
pools. Multiple device mobility inlos can be associated with one
Physical Location
Tliephysical location Is a teg assignedto one or moredevice
pools.It is used io identify whethera deviceIs roaming wiSiin a
device pool
physicallocation or between physicallocations.
The device mobility group is a tag assigned to one or more
device pools. 11 is used lo identify whether a device is roaming
Device MoDdKy Group
willnna device mobility group or between device mobility
groups
The table lists the Dev ice Mobility-related configuration elements and describes their function,
fhe newlv introduced elements arc device mobility infos, the physical location, and the device
mobility group.
The dev ice mobilitv info isconfigured with a name and an IP subnet and isassociated with one
ormore device pools. Multiple device mobility infos can be associated with the same device
pool.
The physical location and the device mobility group are just lags: they are configured with a
name onlv and do not include any other configuration settings. Both arc nonmandatory device
pool configuration parameters: tti3t is. at the device pool, no physical location or one physical
location and one (or no) device mobility group can be chosen. They are used todetermine
whether two device pools are at the same physical location and in the same device mobility
group.
& 20 10 Cisco Systems, Inc
Implementation ofFeatures and Applications for Multisite Deployments
Devi :e Mobility—Dynamic Configuration by LocationDependent Device Pools
This subtopic describes where location-dependent configuration is stored in the Cisco Unified
Communications Manager database and how the applicable set of parameters is chosen.
Device Mobility—Dynamic Configuration by
Location-Dependent Device Pools
Main Site
Remote Site
Device Pool Remote
Local'on
-
Region
•
Location
SRST Reference
SRST Reference
•
Device Motility CSS
•
AAR CSS
•
AAR CSS
MR Group
•
AAR Group
Deuice Mobility CSS
As shown in the figure, the location-dependent parameters (that is, roaming-sensitive settings
and Device Mobility-related settings)are configured at device pools. Basedon the IP subnet
that is used by the phone (which is associated with a device pool), Cisco Unified
Communications Manager canchoose the appropriate device pooland apply the location-
dependent parameters. With Device Mobility, Cisco Unified Communications Manager is
aware of the physical location of a device andapplies the appropriate location-specific
configuration by selectingthe corresponding device pool.
4-10
Implementing Cisco Unitied Communications Manager, Part2 (CIPT2) v8.0
) 2010 Cisco Systems, Inc.
Note
The physical location and device mobility group parameters are used to determine which
settings should be applied to a roaming phone (none, the roaming-sensitive settings only, or
the roaming-sensitive settings and the settings that are related to DeviceMobility). They are
not phone configuration parameters themselves, so therefore, they are not applied to the
phone configuration like the other listed roaming-sensitive settingsare. Theyare used only
at the decision to change the phone configuration and how to change it. Consequently, they
cannot be overlapping and are configurable only at device pools.
Local Route Group
Device Mobilitv -related settings
Device Mobilitv CSS
AAR CSS
—
AAR Group
—
Calling Party Iransformation CSS
Note
All listed DeviceMobility-related settings are overlapping parameters, that is, they are
configurable at phones and at device pools. However, the Device Mobility CSSis called
CSS only inthe Phone Configuration window. Itis notoverlapping with the CSS that is
configured at the line level.
Roaming-sensitive settings are settings that do not have an impact oncall routing. Device
Mobilitv -related settings, however, may have animpact on call routing because they modify the
dev ice CSS. AAR group, and AAR CSS. Depending onthe implementation of Device
Mobilitv. roaming-sensitive settings only—or bolh roaming-sensitive settings and Device
Mobilitv -related settings—can be appliedto a roaming phone.
The GUI does not show the local route group in the roaming-sensitive settings pane.
Nevertheless, the local roule group is a roaming-sensitive setting and is updated when the
physical locations ofthe home device pool and the roaming device pool are different. The
called party transfoniiation CSS is shown in the Device Mobility-related settings pane ofthe
GUI. but this setting does not apply toIP phones and hence isno Device Mobility-related
m
setting, although shown as such in the GUI.
m
©2010 Cisco Systems. Inc
Implementation otFeatures and Applications for Multisite Deployments
Device Mobility—Dynamic Phone Configuration Parameters
This subtopic shows the two types ofphone configuration parameters that can be dynamically
assigned by Device Mobility.
Device Mobility—Dynamic Phone
Configuration Parameters
Two types of phone settings can be applied by Device Mobility.
• Roaming-sensitive settings:
•
Local Roule Group
- Date/Time Group
Region
SRST Reference
Media Resowce Group I iat
•
Locat-oi!
-
Netw:);k Locale
-
Physical Locations
Device MobilityGroup
• Device mobility-related settings:
• Dew.e Mobihiy CSS
•V.RCRS
-
A/.RGi^ir.
- L?.-ii':n,) Party Ti30»fi.imsai!O!i CSS
Device Mobilitv can reconfigure site-specific phone configuration parameters that are based on
the physical location ofthe phone. Device Mobility does not modify any user-specific phone
parameters or any IPphone button settings such as directory numbers.
The phone configuration parameters that can be dynamically applied todie device
configuration are grouped into two categories:
•
Roaming-sensitive settings:
—
Date/Time Group
—
Region
—
Location
Note
The Date/Time Group, Region, and Location are configured at device pools only.
m
—
Network Locale
—
SRST Reference
—
MRGL
Note
The Network Locate, SRST Reference, and MRGL are overlapping parameters. That is, they
areconfigurable at phones and atdevice pools.
—
Physical Location
—
DeviceMobility Group
Implementing Cisco Unified Communications Manager, Pan 2(CIPT2) v8.0
©2010CiscoSystems. Inc.
Device Mobility Overview
This topic describes the key characteristics and Icalures of Device Mobility.
Device Mobility Characteristics
• Device Mobility can be used in multisite environments with
centralized call processing,
• Device Mobility allows users to roam between sites with
their Cisco IP phones (typically, Cisco IP Communicator or
Cisco Unified Wireless phones).
• IP phones are assigned with a location-specific IPaddress
by DHCP.
• Cisco Unified Communications Manager determines the
physical location of theIPphone based on the IP address
used by the IP phone.
• Based on the physical location ofthe IP phone, the
appropriate device configuration is applied.
Dev ice Mobilitv can beused in multisite environments with centralized call processing. It
supports Skinny Client Control Protocol (SCCP) or Session Initiation Protocol (SIP) Cisco IP
phones andCisco IP Communicator.
Dev ice Mobilitv allows users to roam between sites with their IP phones. Typically, these are
Cisco Unified Wireless IP phones or Cisco IP Communicator phones.
When the device is added to the network ofroaming sites, itis first assigned an IP address.
Because the IP networks are difTerent for each site. Cisco Unified Communications Manager
can detemiine the physical location of(he IP phone that is based on its IP address.
Based on the physical location ofthe IP phone. Cisco Unified Communications Manage:
reconfigures the IPphone with site-specific settings.
© 2Q10 Cisco Systems, Inc
Implementation ofFeatures and Applications for Multisite Deployments
Using Device Mobility to Solve Roaming Device Issues
Device Mobility offers functionality thatis designed to enhance themobility of devices within
an IP network.
Device Mobility Solves Issues of
Roaming Devices
Issue WithoutDevice Mobility
When mottle user moves to liferent location,
Cal Admission Control selings are not
Device Mobility Feature to Solve Hieissue
Location settiigs are dynamical? assigned.
adjusted.
PSTN gateways tone used are fixed.
SRST reference is feed.
When mobile user moves to efferent region,
Dynamic phone CSS allows for site-Independent
local gateway access.
SRST inference is flynamicaBy assigned.
codec settings are not adjusted.
Region settings are dynamicallyassigned.
AAR does not work for mobile users.
AAReating search space andAARflroup of
directorynumbers are dynamically assigned.
Media rescurces are assigned independent of
location.
Meda resource listis dynamically assigned.
AAR causes Issues wih Cisco Extension
Cisco Extension Mobtity also benefits from
Mobitty.
dynamic assignment.
ITie device stil! registers with the same Cisco Unified Communications Manager cluster, but it
adapts some ofits behavior that is based on the actual site where itis located. Those changes
aretriggered by the IP subnet inwhich thephone is located. Thetable shows which issues are
solvedby Device Mobility.
Basically, all location-dependent parameters can be dynamically reconfigured by Device
Mobility. Thus, the phone keeps its user-specific configuration, such as directory number,
speed dials, and call-forwarding settings, but adapts location-specific settings like region,'
location, orSRST reference to the actual physical location. Device Mobility can also be
configured in such away that dial plan-related settings, such as the device calling search space
(CSS), AAR group, and AAR CSS, are modified.
4-6
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
»2010 Cisco Systems, Inc.
Issues with Roaming Devices
When phones move between different Cisco Unified Communications Manager sites, some
settings can become inaccurate.
Issues with Roaming Devices
Ifa user is moving between sites and is using a
phone:
• Location-dependent settings are set per phone (MAC).
• The physical device location is assumed to be constant.
• The constant settings become inaccurate:
Region
- Location
SRST reference
AAR group
- CSS
MRGs and MRGLs
Other settings
The configuration ofan IP phone includes personal settings and location-dependent settings
thai are all bound statically to the MAC address ofthe phone and hence to the device itself. The
phv sical dev ice location isassumed to be constant.
Ifa phone or. more likely, asoftphone is moved between sites, the location-dependent settings
become inaccurate. Some of these settings and theirerrors arcas follows:
•m
•
Region: Might cause wrong codec settings
• Location: Might cause wrong Call Admission Control (CAC) and bandwidth settings
• Survivable Remote Site Telephony (SRST) reference: Might cause malfunction ofCisco
Linified SRS I
•
Automated alternate routing (AAR) group: Might cause malfunction ofthe call
redirection on no bandwidth
• Calling search spaee (CSS): Might cause usage of remote gateways instead ofloeai
gateways
• Media Resource Groups (MRGs) and Media Resource Group Lists (MRGLs): Might
cause allocation ofwrong media resources, such as conference bridges oriranseoders
To maintain the correct settings. Cisco Unitied Communications Manager needs lo be aware of
the ph> sical location ofall phones, including moving devices.
© 2010 Cisco Systems. Inc
Implementation ofFeatures and Applications for Multisite Deployments
Issues with Devices Roaming Between Sites
This topic describes issues that occur when users roam between sites with their devices.
Roaming Devices
Device can be an IP phone or, more likely, a softphone (such as
Cisco IP Communicator) ofa roaming user.
Cisco
Unified
Com municatj oris
Remote
Manager
Gateway
•s.
WAN
Roaming Device
When users roam between sites, they might take their phones with them. This situation
typically does not apply lo Cisco IP phones, but is very common with softphoncs such as Cist
IP Communicator orCisco Unified Wireless IP phones.
4-4
Implementing Cisco Unified Communications Manager. Part 2(CIPT2) V8.0
>2010 Cisco Systems, Inc.
Lesson 11
Implementing Device Mobility
Overview
St is common in multisite em ironments that some users roam between sites regularly. When
such users lake their Cisco Unified Communications endpoints, such as aCisco Lnified
Wireless IP Phone or Cisco IP Communicator (a softphone) phone, with them, the standard
configuration of their endpoints must be adapted to suit the needs ot the current physica
location. It is important that aprofessional unified communicalions system provides such a
solution.
This lesson describes Cisco Unified Communications Manager Device Mobility, afeature of
Cisco Unified Communications Manager that allows its endpoints lo be dynamically
reconfigured based on their actual location as determined by the IP address that is used bv the
device.
Objectives
Upon completing this lesson, you will be able to describe how Device Mobility works and how
it is implemented. This ability includes being able to meet these objectives:
Identify the issues with devices roaming between sites
Describe the Device Mobility feature
Describe the Device Mobility configuration elements and their interactions
Describe Device Mobility operation
Describe Device Mobility interaction with globalized call routing
Contigurc Device Mobility
ttfc
mm
4-2
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
>20f0 Cisco Systems, Inc.
Module 41
Implementation of Features
and Applications for Multisite
Deployments
Overview
Users in multisite environments often roam between sites. They cither take devices (such as a
Cisco Unified Wireless IP Phone or Cisco IP Communicator) with them or use guest phones at
the siles that they roam to.
This module describes Cisco Unified Communications Manager Device Mobility and Cisco
Extension Mobilitv and their implementation. The implementation provides users with the
freedom to roam and still be reachable bv their own extensions, no mailer where they are or
what dev ice thev use.
Module Objectives
Upon completing this module, you will be able to implement Device Mobility and Cisco
Fxtension Mobility. This ability includes being able to meet these objectives:
• Dcscnbc how Device Mobilitv works and how il is implemented
• Describe how Cisco Extension Mobility works and how il is implemented
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
)2010 Cisco Systems, Inc.
Monitoring Leamed Routes in Cisco Unified Communications Manager Express
CCD PSTN Backup—CSS
SRST Considerations
CCD and Static Routing Integration Considerations
Cisco IOS SAF Client Considerations When Using Globalized Call Routing
Solution for PSTN Backup Advertised in E.164 Format Without Leading +
TEHO Considerations
Trunk Considerations When Using Globalized Call Routing
Cisco Unified Communications Manager Clusters and CCD Configuration Modes
Other SAF and CCD Considerations
„
Summary
References
jj-58
J"^
£-bi
5-b^
5-63
™
^
5-bb
5-67
d-oo
_g8
,„
Module Summary
5]6g
References
,. -,.
Module Self-Check
Module Self-Check Answer Key
••c 2010 Cisco Systems. Inc
5-57
Implemenling Cisco Unified Communications Manager, Part 2(CIPT2) v8 0
r~
Step 7b: Subscribe Phone to Cisco Extension Mobility Phone Service
4-67
Summary
~
References
Tro
Module Summary
7™
References
71^
Module Self-Check Answer Key
4"'4
Module Self-Check
Jy
Call Control Discovery
.
Module Objectives
Implementing SAF and CCD
.
Objectives
SAF and CCD Overview
5~1
$£
^
5-4
Dial Plan Scalability Issues in Large Networks
Scalable Dial Plan Solution for Large Networks
Call Control Discovery Overview
SAF Characteristics
SAF Client Types
SAF Message Components
5-5
5"6
5-7
5-8
5"9
5-10
SAF Client and SAF Forwarder Functions
CCD Characteristics
CCD Characteristics
CCD Services in Cisco Unified Communications Manager
5-13
5-14
SAF Routing Characteristics
SAF Neighbor Relationships
Processing Received Routes in Cisco Unified Communications Manager
CCD Operation
5-11
5"12
5-15
5-17
5-19
5_2|
CCD—Propagation of HQ Routes
CCD—Propagation of BR Routes
5"22
5-23
CCD—Call from HQ to BR
CCD—Link Failure at BR
5-24
5"25
CCD—Call from HQ to BR During Link Failure
SAF and CCD Implementation
External SAF Client Configuration Elements
Relationship ofExternal SAF Client Configuration Elements
Internal SAF Client Configuration Elements
Relationships ofInternal SAF Client Configuration Elements
SAF Forwarder Configuration Procedure
Step1: Configure SAF Forwarder
Step2: Configure SAF Forwarder to Support External SAF Clients
External SAF Client Configuration Procedure
Step1: Configure SAF Security Profile
Step2: Configure SAF Forwarder
Step 3: Configure SAF Trunk
Step 4: Configure Hosted DN Group
Step 5: Configure Hosted DN Pattern
Step6: Configure CCD Advertising Service
Step 7: Configure CCD Requesting Service and Partition
Step 8: Configure CCD Blocked Leamed Patterns
Step 9: Configure CCD FeatureParameters
Internal SAFClient Configuration Procedure
Step 1: Configure Trunk Profile
Step 2: Configure Directory Number Blocks
Step 3: Configure Call Control Profile
Step 4: Configure Advertising Service
Step 5: Configure Requesting Service
Step 6: ConfigureVoIP Dial Peer
CCD Considerations
Monitoring Leamed Routes in Cisco Unified CommunicationsManager
ii
—
5-1
5-1
Overview
ImplementingCisco Unified Communications Manager, Part 2 (CIPT2)v8.0
5-26
5-28
5-3°
5-31
5-32
5-33
5-34
5-35
5-36
5-37
5-38
5-39
5-40
5-41
5-42
5-43
5-44
5-45
5-46
5-48
5-49
5-50
5-51
5-52
5-53
5-54
5-55
5-56
© 2010 Cisco Systems, Inc.
Table of Contents
Volume 2
Implementation ofFeatures and Applications for Multisite Deployments
mll
Overview
4-1
Module Objectives
Implementing Device Mobilitv
m
**
^
mm
**•
mm
mm
ww
**
m
±1
4-1
___
4.3
Objectives
Issues with Devices Roaming Between Sites
Issues with Roaming Devices
Using Device Mobility to Solve Roaming Device Issues
Device Mobility Overview
Device Mobility—Dynamic PhoneConfiguration Parameters
Device Mobility—Dynamic Configuration by Location-Dependent Device Pools
4^
4_4
4_5
4_6
4_7
4-8
4-10
Relationship of Device Mobility Configuration Elements
Device Mobility Operation
Device Mobility Operation: Flowchart
Device Mobility Considerations
Device Mobility and Calling Search Spaces
Examples ofCall-Routing Paths Based on Device Mobility Groups and TEHO
Device Mobility Interaction with Globalized Call Routing
Advantages ofUsing Local Route Groups and Globalized Call Routing
4-12
4_14
4_1g
4^1 g
4_20
4-22
4_24
4-26
Example: No Globalized Call Routing—Same Device Mobility Group
Example: Globalized Call Routing
Device Mobility Configuration
Steps 1 and 2. Configure Physical Locations and Device Mobility Groups
Step 3: Configure Device Pools
4-28
4_29
4_30
4-31
4.32
Device Mobility Configuration Elements
Example: No Globalized Call Routing—Different Device Mobility Group
4_11
4-27
Step 4. Configure Device Mobility Infos
4.33
Step 5a: Set the Device Mobility Mode Cisco CallManager Service Parameter
Step 5b: Set the Device Mobility Mode for Individual Phones
4-34
4-35
Summary
References
Implementing Cisco Extension Mobilitv
4_3g
4.37
4.39
Objectives
Issues with Users Roaming Between Sites
Issues of Roaming Users
Cisco Extension Mobility Solves Issues of Roaming Users
Cisco Extension Mobility Overview
Cisco Extension Mobility: Dynamic Phone Configuration Parameters
Cisco Extension Mobility: Dynamic Phone Configuration by Device Profiles
Cisco Extension MobilityConfiguration Elements
Relationship of Cisco Extension Mobility Configuration Elements
Cisco Extension MobilityOperation
4.39
4-42
4-43
4-45
4-46
4-47
4-49
4-50
Issues in Environments with Different Phone Models
Default Device Profile and Feature Safe
How Cisco Extension Mobility Handles Phone Model Differences
Cisco Extension Mobility and CSSs
Alternatives for Mismatching Phone Models and CSS Implementations
Cisco Extension MobilityConfiguration
Step 1: Activate the Cisco Extension MobilityService
Step 2: Set Cisco Extension MobilityService Parameters
Step 3: Add the Cisco Extension MobilityPhone Service
Step 4: Create Default Device Profiles
Step 5a: Creale Device Profiles
4-52
4-53
4-54
4-56
4-57
4-58
4-59
4-60
4-61
4-62
4-63
4-40
4-41
Step 5b: Subscribe Device Profile to Cisco Extension Mobility Phone Service
4-64
Step 6: Associate Users with Device Profiles
Step 7a: Configure Phones for Cisco Extension Mobility
4-65
4-66
3-110
Implementing CiscoUnified Communications Manager. Part 2 (CiPT2) v8.0
) 2010 Cisco Systems, Inc.
Module Self-Check Answer Key
QD
D
Q2)
A
Q-D
B
Q4)
C
05)
D
06)
B
Q-)
Standard locations-based CAC docs nol allow tiic configuration of a dilTerenl limit per pair of locations.
Onh a tola! limit for all calls coming in to or going oul of a location can be configured.
Q8)
B
Q9)
B. [•
QIO)
B
qui
A
©2010 Cisco Systems, Inc.
Bandwidth Management and CAC Implementation
3-109
Q7)
Q8)
What is alimitation ofstandard locations-based CAC? (Source: Implementing CAC)
Which statement is false about CAC when using RSVP-enabled locations? (Source:
Implementing CAC)
A)
CAC adapts to the actual topology and considers network changes.
B)
RSVP is used for CAC and toprovide QoS for each individual stream.
C)
The RSVP agent to be used by aphone is determined by the Media Resource
Group Listof the phone.
D)
The RSVP agent isconfigured as an MTP in Cisco Unified Communications
Manager.
09)
AAR reroutes calls to the PSTN for which two types ofcalls? (Choose two.) (Source:
Implementing CAC)
A)
calls rejected by an H.323 gatekeeper
B)
calls rejected by standard location-based CAC
C)
D)
calls placed tounregistered phones
calls placed toa gateway that is busy
E)
F)
callsrejected by RSVP-enabled location-based CAC
calls rejected bySIPprecondition-based CAC
Q10) When using end-to-end RSVP with SIP preconditions, RSVP is used between the
originating and the terminating phone. (Source: Implementing CAC)
A)
B)
true
false
QII) How can calls that are rejected by an H.323 gatekeeper be rerouted by using adifferent
path? (Source: Implementing CAC)
A)
B)
by configuring route lists and route groups with backup devices
by putting the gatekeeper-controlled intercluster trunk or H.225 trunk into a
C)
by configuring a second route pattern in the same partition that refers to a
backup device cannot be done, because AAR only supports internal calls
location that is set to unlimited
3-J08
implementing Cisco Unrfied Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems, Inc.
Module Self-Check
Use the questions here to review what you learned in this module. The correct answers and
solutions are found in the Module Self-Check Answer Key.
Ql)
Which feature does not conserve bandwidth in the IP WAN? (Source: Managing
Bandwidth)
A)
B)
02)
RTP headercompression
low-bandwidth codecs
C)
local media resources
D)
quality of service
How can the bandwidth per call be limited in Cisco Unitied Communications
Manager? (Source: Managing Bandwidth)
A)
B)
C)
D)
b\ specifying the maximum permitted codec bandwidth between pairs of
regions
bv specifying the maximum permitted codec bandwidth between pairs ot
locations
by specifying the maximum bandwidth per stream with the bandwidth zone
local command
by specifying the maximum permitted codec bandwidth tor calls going out ot
or coming into a region
Q3)
When dcploving local conference bridges at each site, what is the minimum number of
Media Resource Group Lists that arc required? (Source: Managing Bandwidth)
A)
number ofsites *(number ofsites - l)/2
B)
one per site
D)
one
C)
Q4>
one per site and one per conference bridge
Which device requires access to Ihe transcoder from its Media Resource Group List
when transcoding is required for acall? (Source: Managing Bandwidth)
A)
B)
both endpoinls of the call
the callingdevice
D)
the called device
C)
Q5)
ade% ice that supports only codecs that are not permitted tor the call
Which statement is true about multicasl MOI 1from branch router flash? (Source:
Managing Bandwidth)
A)
B)
C)
D)
Multicast MOH from branch router flash requires an SRST router to be in
fallback mode to work.
Multicast MOH from branch router flash can also be used tor unicast MOH.
The branch router supports G.711 and G.729 only for MOH.
Regions in Cisco Unified Communications Manager have to be configured in
such away that G.711 is allowed between the Cisco Unified Communications
Manager MOH server and the branch phones.
Q6)
Which CAC-related feature applies to intercluster calls? (Source: Implementing CAC)
A)
B)
C)
D)
locations
H.323gatekeeper CAC
AAR
RSVP-enabled locations
Bandwidth Management and CAC Implementation
) 2010 Cisco Systems. Inc
3-107
3-106 implementing Cisco Unified Communicalions Manager, Par. 2(CIPT2) v8.0
©2010 Cisco Systems, Inc.
Module Summary
This topic summarizes the key points that were discussed in this module.
Module Summary
• Cisco Unified Communications Manager supports several
features that reduce bandwidth requirements in multisite
environments.
* Cisco Unified Communications Manager CAC mechanisms
include locations, RSVP-enabled locations for calls within a
cluster, RSVP-enabled locations for calls through SIP trunks,
and H.323 gatekeeper-based CAC.
This module described the available design options and features that are recommended tor
deplovment in amultisite environment in order to reduce bandwidth requirements mthe II
WAN The module also described the different ways of implementing Call Admission Contro
(C \C) within acluster and beyond cluster boundaries. Finally, the module explained how calls
can be rerouted over the public switched telephone network (PSTN) ifthere is insufficient
bandwidth.
References
For additional infomiation. refer to these resources:
• Cisco Svstems. Inc. Cisco Unified Communications System 8.x SRMX April 2010.
http-,'Uuw.cisco.c<>iii'cii/"US/docs.'\oicejp_.co.nin/cucm/snid/8x/uc8x.htinl
• Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0/1). February 2010.
_
lmp:/^vNu.asco.coni/en.'l.S.'doc>.\oicc_ip_comin/cuem/ailn.m/8
O.J/ecmdg.bccm-801cm.html
• Cisco Svstems Inc. Cisco Unified SRSTSystem Administrator Guide, December 2007.
IUtp:.vw;w.cisco.coni/eivUS/partner/docs/voice ip^onitn/aisrst/adtnin/srst/configurtmc.,..
guide srr.lsa.hlml
. Cisco Svstems. Inc. Cisco IOS H.323 Configuration Guide Release 15.0- Configuring
H3"3 Gatekeepers and Proxies. February 2008. October 2009.
lu.p.
x,uxv.cisco.co,n.cn/US/partner/docs/.os/voice/l,323/conngu,-at.on/gu,dc.vh.h32, gk
config psl05'Jt TSD Products Configuration Guide.Chupler.lilm!
Bandwidth Management and CAC Implementation
) 2010 Cisco Systems. Inc
3-105
Summary
This topic summarizes the key points that were discussedin this lesson.
Summary
• CAC limits the number ofcalls in order to avoid voice quality
issuescaused bybandwidth oversubscription due totoomany
voice calls.
• Cisco Unified Communications Managerlocations can be used
for CAC within a Cisco Unified Communications Manager
cluster that has a hub-and-spoke topology.
1Cisco Unified Communications Manager RSVP-enabled
locations provide topology-aware CAC between RSVP agents.
AAR allows calls that were denied by locations-based CAC to
be rerouted over the PSTN.
SIP Preconditions allows calls between clusters toflow through
dedicated routers.
H.323 gatekeepers can provideCAC on Cisco Unified
Communications Manager H.323 trunks.
References
For additional information, refer to these resources:
• Cisco Systems, Inc. Cisco Unified Communications System 8.x SRND, April 2010.
•
http:/Avuu.cisco.com/en/US/docs/voice_ip .coinm/cticm/srnd/8N/uc8\.himl
Cisco Systems, Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0(1), February 2010.
h!tp:/Av\u\.cisco.c(>m/enAJS/doLVvoicc_ip_comnVcucin/admin/8_0_l/ccmcfe/bccni-80I-
em.html
• Cisco Systems, Inc. Cisco IOS H.323 Configuration Guide Release 15.0 - Configuring
H323Gatekeepers and Proxies, February 2008, October 2009.
http://\\u^.cisco.com/en/L!S/partner/docs/ios/voicc/h323/coniiguration/guidc/vh_h323 gk
contIg_psl059i_TSD_Products_Connguration_Guide_Chapter.html
3-104
Implementing Cisco Unified Communications Manager. Part 2(CIPT2) v8.0
©2010 Cisco Systems, Inc.
Note
The command syntax and a sample configuration were shown earlier in this topic.
To provide backup paths for the gatekeeper-controlled trunk, perform this step:
Step 1
Add PSTN gateways lo route groups, and add these gateways to the route lists that
are using the gatekeeper-controlled trunks.
Note
You configured PSTN backup in earlier lab activities ofthis course
Bandwidth Management andCAC Implementation
© 2010 Cisco Systems. Inc
3-103
Configuration Procedure for Implementing H.323 GatekeeperControlled Trunks with CAC
The figure shows the required steps for implementing gatekeeper-controlled trunks for call
routing only. It also shows you how to add gatekeeper CAC functionality and how to provide a
backup pathfor the gatekeeper-controlled trunk.
Configuration Procedure for
Implementing H.323 GatekeeperControlled Trunks with CAC
Implement gatekeeper-controlled trunks:
1 Configure gatekeeper (Cisco IOS router) for call routing.
2. Add gatekeeper to Cisco Unified Communications Manager.
3 Add and configure gatekeeper-controlled trunk (gatekeeper4
controlled intercluster trunk or H.225 trunk) inCisco Unified
Communications Manager.
Configure route patterns, routelist, and route group pointinq to
gatekeeper-controlled trunk.
Implement gatekeeper CAC:
•
Configure gatekeeper (Cisco IOS router) for CAC.
Provide PSTN backup for gatekeeper-controlled trunk;
•
Modify route list and route groups to provide a PSTN backup
path for the gatekeeper-controlled trunk.
To implement gatekeeper-controlled trunks, perform these steps:
Step 1
Enable gatekeeper functionality at aCisco IOS router and configure the gatekeeper
for call routing. This configuration typically includes zones, zone prefixes, and the
default technology prefix.
Note
More information about gatekeeper configuration is provided in the Implementing Cisco
Voice Communications and QoS (CVOICE) course.
Step 2
Step 3
Add the gatekeeper to Cisco Unified Communications Manager.
Add the gatekeeper-controlled trunk (either agatekeeper-controlled intercluster
trunk or an H.225 trunk) to Cisco Unified Communications Manager, and configure
the trunk.
Step 4
'
&
Configure route groups, route lists, and route patterns in order to route calls that
match acertain route pattern (for example, 9.5[l2][l2].„ , for the examples that
were shown earlier in this topic) tothe gatekeeper-controlled trunk.
Note
You performed the last three steps of the preceding procedure in earlier lab activities of this
course.
To implement gatekeeper-controlled CAC, perform ihis step:
Step 1 Configure the Cisco IOS gatekeeper with bandwidth commands to enable bandwidth
-~n .^
imitations (typically required only for interzone calls).
limitations (tvniYallv rpni n r**H rmlw frt- :«<^
3-102
Implementing Cisco Unified Communicalions Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems, Inc.
Note
For a PSTN backup, you need to perform digit manipulation msuch away that the calling
number and (more importantly) the called number are transformed to always suit the needs
of the device that isactually used. This transformation can be done atthe route list, where
digit manipulation can be configured per route group. In the example, the called number 91
511 555-1234 has to bechanged to a 10-digit number for the H.225 trunk, because the
gatekeeper is configured with area code prefixes without the long distance 1 The called
number must also bechanged to an11 -digit number if rerouting the call to the PSTN
gateway is necessary. Abetter solution would be using global transformations atthe egress
devices (H 225 trunk and PSTN gateways). In a large multisite environment or in an
international deployment, the implementation of globalized call routing would be the best
solution
Bandwidth Management andCAC Implementation
© 2010 Cisco Systems, Inc.
3-101
Providing PSTN Backup for Calls Rejected by CAC
This subtopic describes how backup paths can beprovided for calls that are rejected by a
gatekeeper because of CAC.
Providing PSTN Backup for Calls
Rejected by CAC
Route lists and route groups provide backup paths if the
preferred path fails. In the example, ifthe calls cannot be
established over the H.225 trunk because of CAC, the calls use
the PSTN gateway (HQ-1) as backup.
Cisco Unified Ccmmmieaiors
Ci sco Un ified C om mu n ica t ions
Manager Cluster
Manager Cluster
PSTN Prefixes 511.521
PSTN Prefixes: 512, 522
HQ-1
Route Pattern:
9.15112J2XXXXXXX
Route Group. H.225
Trunk. Clusterl
Route Group PS I N
Gateway HQ-1
A call that is placed to a gateway or a trunk can fail for many reasons:
• The appropriate device can be down. Timeouts occur when a call isplaced toan H.323
gateway, when an ARQ message issent toan H.323 gatekeeper, orwhen keepalive
messages are notexchanged with an MGCP gateway.
• Communication problems can occur with the gateway. H.323 messages can be sent tothe
IP address of thewrong interface. Gatekeeper registration can fail because of an invalid
zone name orbecause the call isrejected due toa lack ofresources. Acall might be
rejected when no channel is available on an EI or Tl trunk, when an administratively
configured limit ofcalls isreached atadial peer, orwhen a call isdenied by CAC.
Cisco Unified Communications Manager uses the same backup method—route lists and route
groups—for all ofthese types ofcall failures. Ifthe currently attempted device ofaroute group
cannot extend the call (for whatever reason), Cisco Unified Communications Manager will try
the next device according to the route group and route list configuration.
Therefore, providing abackup for calls that have been rejected due to H.323 gatekeeper CAC is
as simple as having aroute list and route groups that prefer the gatekeeper-controlled trunk
overoneor more PSTN gateways. If thecall cannot besetup overthe trunk, Cisco Unified
Communications Manager will reroute the call to the PSTN gateways. Instead ofreferring to a
dedicated PSTN gateway that should be used as abackup, the local route group feature can be
used.
3-100
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
52010 Cisco Systems, Inc.
The maximum audio bandwidth is limited to 64 kb/s. Calls requiring more bandwidth {for
media such as wideband audio codecs or video calls with a video call bandwidth of more
than 64 kb/s) are not permitted in any zone.
Thetotal of all calls(interzone andintrazone calls) in zone ClusterB must notexceed 688
kb/s. As anexample, ibis configuration allows three G.729 calls to ClustcrA (three times
twice the codec bandwidth of 8 kb/s) and five G.711 calls within ClusterB (five times twice
the audiocodec bandwidth of 64 kb/s). Intrazone calls in zone ClusterA are unlimited.
Note
Some ofthe bandwidth commands inthe example are forillustration only andare not useful
inthis scenario. Forexample, youcould change the bandwidth interzone default 64
command to bandwidth interzone ClusterA 64 because interzone default appliesonly to
zone ClusterA; zone ClusterB is explicitly configured, and no other zones exist. Furthermore,
intrazone limitations have been configured but would never apply inthis scenario. The
reason is that all calls are interzone calls.Thegatekeeper is used only forintercluster calls,
and the two clusters are in different zones.
mm
©2C1Q Cisco Systems. Inc
Bandwidth Management and CAC Implementation
3-99
Example: H.323 Gatekeeper Also Used for CAC
The figure shows an example ofaCisco IOS H.323 gatekeeper that has CAC enabled.
Example: H.323 Gatekeeper Also Used
for CAC
Cisco Unified Communications
Manager Cluster
PSTN Prefixes 511.521
Cisco Unified Communications
Manager Cluster
PSTN Prefixes 512,522
192.168.3.254
pow Trunk: Cluster^ ^mm^ Ji.225 Trunk: CL-^
10.1 1.1
10.1.1.1
gatekeeper
10.1.1 2
ions local ClusterA lab.ccm 192.168.3.254
10.2.1 2
ions local ClusterB lab.cm 192.169.3.254
H 225 Trunk'Clusterl
DeviceType. Gateway
Zone ClusterA
Technology Prefix Mr
GK 192 158 3 254
•one prefn ClusterA 511-
H.225 Trunk; Cluster2
lone prefix ClusterA 521"
ions prefin ClusterB 513«
DeviceType.Gatenvay
rone prefix ClusterB 522*
Zone1 ClusterB
bandwidth interzone default a
bandwidth intertone sons CI ustsrB
bandwidth session default 128
Technology Prefix: Mr
IS
GK 192.168 3.254
bandw4.dth total zone ClusterB 688
9"-type-prefix lt« default- technology
no shutdown
The example is based on the previously illustrated example, but now the H.323 gatekeeper also
performs CAC.
The bandwidth interzone default 64 command specifies that 64 kb/s is permitted for calls
going out of and coming into azone. Because no specific zone is specified but the keyword
default is used, this setting applies to all zones that are not explicitly configured with a
different setting.
The bandwidth interzone zone ClusterB 48 command specifics that the previously configured
default interzone bandwidth limit should not apply to ClusterB but that ClusterB should instead
be limited to 48 kb/s.
The bandwidth session default 128 command limits the bandwidth to be used per call to a
codec that does not require more than 64 kb/s (for example, G.711 or G729) Because no
d.fferent session bandwidth is configured for any specific zone, this default applies to all zones.
The bandwidth total zone ClusterB 688 command limits all calls of ClusterB (that is calls
within the cluster and intercluster calls) to atotal of688 kb/s. Because there is neither a
bandwidth total default command nor aspecific bandwidth total command for ClusterA
ClusterA has nototal limit applied.
IfG.729 is used for inter/one calls and G.711 is used for intrazone calls, this configuration
effectively would permit these limitations:
> There can be amaximum of three G.729 calls between ClusterA and ClusterB because
CusterB is limited to 48 kb/s (that is, three times twice the codec bandwidth of8kb/s)
ClusterA could have four G.729 calls to other zones. However, because the example shows
only two zones and the other zone (ClusterB) is limited to three G.729 calls ClusterA will
never beable to use thepermitted interzone bandwidth.
3-98
Implemenling Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems, Inc.
Bandwidth limitations are configured differently on different Cisco products and for different
features. The table summarizes how to configure bandwidth limitations in Cisco Unified
Communications Manager.
Cisco Unitied
Cisco Unified
Cisco IOS H.323
Communications
Communications
Gatekeeper
Manager
Manager
Region
Location
Audio-Only Call
Configuration
Audio codec only
Audio codec bit rate +
Twice the audio codec
Layer3 overhead
bit rate
Example: G 711 Call
G.711
80 kb/s
128 kb/s
Audio codec and video
Video call speed
Twice the video call
Video Call
Configuration
call speed
Example 384-kb/s
G.711 and 384 kb/s
speed
384 kb/s
768 kb/s
Video Call
Note
Video calls have not been discussed in this course but are also shown for completeness.
Bandwidth Management and CAC Implementation
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Using an H.323 Gatekeeper for CAC
To use an H.323 gatekeeper for CAC, you have to configure bandwidth limitations, as shown
the figure.
Using an H.323 Gatekeeper for CAC
router(config-gk)#
bandwidth {interzone | total | session}
{default | zone zone-name} bandwidth-size
H
1Sets the maximum bandwidth (in kb/s) permitted per zone (or the
default for all zones not explicitly configured):
• interzone: Applies to all calls coming into and going out of the
specified zone (interzone calls)
- total: Applies toall calls in the specified zone (interzone and
intrazone calls)
• session: Applies toeach call in thespecified zone (specifies
the maximum bandwidth per call)
Bandwidth ofa call is twice the bandwidth ofthe audio codec:
- G.711:128 kb/s
• G.729: 16 kb/s
In Cisco IOS Software, you implement H.323 gatekeeper CAC by using the bandwidth
command.
bandwidth Command Parameters
Syntax
Description
interzone
Specifies the total amount of bandwidth for H.323 traffic from the zone to any
other zone.
total
'
Specifies the total amount ofbandwidth for H.323 traffic that isallowed in the
zone.
session
Specifies the maximum bandwidth that isallowed for a session in the zone
default
Specifies the default value for all zones.
zone
Specifies a particular zone.
zone-name
Names the particular zone
bandwidth-size
Maximum bandwidth. For interzone and total, the range isfrom 1to
10,000,000 kb/s. For session, therange isfrom 1to 5000 kb/s.
The bandwidth that is calculated per call is twice the bandwidth of the audio codec AG729
call consumes 16 kb/s of the configured bandwidth, and a G.711 call consumes 128 kb/s of
the configured bandwidth.
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Thev all use different H.323 IDs because different trunk names have been contigured in the two
clusters and because Cisco Unified Communications Manager adds the _1 and _2 to the trunk
name to uniquely identify the call-processing servers per cluster.
Note
If the same trunk name was configured in the two clusters, registrations would fail because
ofduplicate H.323 IDs.
The call-processing servers of ClusterA registered in /one ClusterA. and the call-processing
servers of ClusterB registered in zone ClusterB. You can verify this situation by using the
command show gatekeeper endpoints. All endpoints are registered with the prefix 1# which
is configured to be the default technology prefix. You can verify this situation by using the
command show gatekeeper gw-type prefix. The output of these two commands is shown m
one table in the figure.
Note
For more information regarding gatekeeper configuration and operation, refer to the
Implementing Cisco Voice Communications and QoS (CVOICE) course.
____
Ifthe gatekeeper receives an Admission Request (ARQ) message from one ofthe H.323
aatewavs (a call-processing Cisco Unified Communications server, in this case), it looks up its
call-routing table (list of configured /one prefixes) to find out in which /.one the requested
prefix can be found.
You can verifv the list of configured prefixes and their /ones by using the command showgatekeeper zone prefix.
If an ARQ message was sent from 10.1.1.1 to the gatekeeper that requests acall to
512555P34 the gatekeeper will determine that the call has to be routed to zone ClusterB. Inc
onlv prefix that is reg.stered by gateways in this zone is 1#*. which is the default technology
prefix and is registered bv 10.2.1.1 and 10.2.1.2. Therefore, the gatekeeper chooses one of these
two eatcwa% s(in round-robin fashion) to be the terminating gateway. It will inform the
originating gatewav (the call-processing server ofClusterB that sent the ARQ message) to set
up an 11.323 call with the determined terminating gateway (10.2.1.1 or 10.2.1.2).
Note
At this point, the gatekeeper is configured only to perform call-routing address resolution. It
resolves adialed number to the IP address where the call has to be routed. No CAC is
performed by the gatekeeper in this example
Bandwidth Management and CAC Implementation
j 2010 Cisco Systems, Inc.
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Example: H.323 Gatekeeper Used for Call Routing (Address
Resolution) Only
The figure shows an example for gatekeeper-controlled trunks in adistributed Cisco Unified
Communications Manager deployment.
Example: H.323 Gatekeeper Used for
Call Routing (Address Resolution) Only
Cisco UnrfiedCommunications
u„„,„ r-,
.
Manager
Cluster
i-i,™ 11„-« .. r-
Cisco Unified
Communications
,,,
„, .
PSTN Pr*« 511.52,
101 11 -"
PSTN pXe"'512^22
192.168.3.254
H.225Trur^Clus^ ^ ^ J^Tn** C,uste/2
10.2.1.1
gatekeeper
10.1.1.2
H 225 Trunk Clusleri
DeviceType Galeway
Zone CluslerA
Technology Prefix If
GK. 192168.3254
zone local ClusterA lab.con
zone local ClusterB lab.com
10.2.1.2
zona prefix ClusterA 511*
zone prefix ClusterA 521-
H.225 Trunk: Cluster2
zone prefix ClusterB 512*
DeviceType. Gateway
zone prefix CluaterB 522*
g»-type-prefix 1#« default-technology
no shutdown
Zone: ClusterB
Technology Prefix MT
GK. 192 16S.3.254
GATEKEEPER BKDPOINT HEGISTRATIOH
H323-ID
IPAddr
ZonaName Type
Prefii
Clusterll
Clusterl_2
Cluster2_l
ClusterJ_2
10.1.1.1
10.1.1.2
10.2.1.1
10.2.1.2
ClusterA VOIP-GW
ClusterA VOIP-GW
ClusterB VOIP-GW
ClusterB VOIP-GW
1#*
1#'
If
1#*
In the example, two Cisco Unified Communications Manager clusters arc shown Each cluster
hasan H.225 trunk configured.
The 11.225 trunks use different names per cluster in order to keep the H323 IDs unique
ClusterA uses Clusterl. and ClusterB uses Cluster2. In each cluster, there are two call-
processing nodes (10.1.1.1 and 10.1.1.2 in ClusterA, and 10.2.1.1 and 10.2.1.2 in ClusterB).
The trunk in ClusterA with the name Clusterl is configured with zone ClusterA and technology
prefix 1# . The trunk in ClusterB with the name Cluster2 is configured with zone ClusterB and"
mVITi6^!??010^ PrCfiX ('**}" B°th tnmks refer t0 **1P address oUhc sa™ gatekeeper:
I yl. 168.3.254.
The gatekeeper has two local zones: ClusterA and ClusterB. It is configured to route calls to
prefixes 511 and 521 to zone ClusterA, and calls to prefixes 512 and 522 to zone ClusterB In
addition, the gatekeeper is configured to use 1#* as the default technology prefix 'fhat is calls
to prefixes for which the gatekeeper does not know which gateway to use are routed to the
gateway orgateways that registered a technology prefix of I#*).
This gateway configuration means that the gatekeeper will have four gateways registered:
' Manager
MUSterL^*Group
WhiCuthatfS isconfigured
tHe ^ cal|-Processing
server of the Cisco Unified Communications
in the device pool ofthe trunk
• Cluster ]_2. which is the second call-processing .server ofClusterA
•
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Thetwo call-processing servers of ClusterB
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) vB.O
©2010 Cisco Systems, inc.
Note
The H.323 ID has to be unique. Cisco Unified Communications Manager keeps the H.323 ID
that is used by the members of acluster unique by adding the individual ending _x
Furthermore because Cisco Unified Communications Manager does not allow multiple
trunks to use the same name, no duplicate H.323 IDs can be presented to the gatekeeper
from a duster However, if the same trunk name is configured in multiple clusters, the call-
processing servers of two or more clusters will try to register with the same H.323 ID. The
gatekeeper will not allow these duplicate H.323 IDs to register, so the trunk will not be
operational Therefore, it is important to use unique trunk names across all Cisco Unified
Communications Manager clusters that register with a gatekeeper.
_
II 323 zone: H323 zones are used to group devices. You perform call routing and CAC
based on these /ones. For instance, you could configure aso-called detault technology
prefix per zone that identifies the gateway (or gateways) to which calls should be routed
when the gatekeeper does not know which gateway to use. Also. CAC can be configured
differentlv for calls within a zone versus interzone calls.
Note
The H323 zone name that is configured at the gatekeeper-controlled trunk is case-sensitive
and hastoexist atthe gatekeeper
^
__
• Technology prefix: H.323 gateways (including Cisco Unified Communications Manager)
can register prefixes (that is. number ranges that they can route calls to) at the gatekeeper.
The prefix can consist onlv ofnumbers (for example. 511). or it can include atechnology
prefix (such as IM. 2#. and so on). One way of using an H.323 technology prefix is tor a
»atewav to indicate the services that it provides by specifying an appropriate technology
prefix ('for instance. 1* for voice services, 2* for fax services, and so on). Calls that include
the technology prefix in their numbers (for example, acall that is placed to 1*5115551000)
can be muted'to the gateway in the zone that registered the appropriate technology prefix.
\s mentioned earlier, agatekeeper can be configured to route calls-for which it does not
know which gateway to use-to the gateway or gateways that register with aprefix that is
configured to be the default technology. For example, if there is only one Cisco Unified
Communications Manager cluster registering per zone, you can configure the trunk in each
cluster with atechnology prefix of 1*#. and configure the gatekeeper lo send all calls to the
gateway that registered with the configured default technology prefix (1*# in this case).
The eat'ekeeper needs onlv aconfiguration ofnumber prefixes (which number to find in
which /one). When the outgoing zone is determined, the gatekeeper just sends the call to
one of the gateuaj s(Cisco Unified Communications Manager systems) that registered in
the zone with the default technology prefix.
Note
More information about how agatekeeper routes calls is provided in the Implementing Cisco
Voice Communications and QoS (CVOICE) course.
_
Bandwidth Management and CAC Implementation
©;010 Cisco Systems. Inc
3-93
H.323 Gatekeeper CAC
This topic describes 11.323 gatekeeper CAC support in Cisco Unified Communications
Manager.
Cisco Unified Communications
Manager Support for H.323 Gatekeepers
Cisco Unified Communications Manager supports H.323
gatekeeper networks using:
• Gatekeeper-controlled intercluster trunk
- To be used with Cisco CallManager versions earlier than v3.2
• H.225 trunk
- To be used with Cisco CallManager v3.2 or later, Cisco Unified
Communications Manager, and all other H.323 devices
Trunks register with gatekeeper and provide the following
information:
• Type of device (Usually, Cisco Unified Communications Manager
registers as gateway.)
• H.323 IDof trunk is built from trunk name (plus _x, where x is a
number that identifies each call-processing server in cluster).
• H.323 zone.
• Technology prefix.
Cisco Unified Communications Manager can connect to other Cisco Unified Communications
Manager clusters or to any other H.323 devices via H.323 trunks. H.323 trunks can be
configured on their own—without the use of a gatekeeper for address resolution and CAC—or
as gatekeeper-controlled trunks. You can configure two gatekeeper-controlled trunks in Cisco
Unified Communications Manager:
•
Gatekeeper-controlled intercluster trunk: This trunk is used to connect to Cisco
CallManager versions earlier than 3.2.
•
H.225 trunk: This trunk can be used to connect to Cisco Unified Communications
Manager Version 3.2 or later and to all other H.323 devices. The H.225 trunk features a
peer discovery mechanism and hence can identify the device that is located at the other end
of the trunk and use the appropriate feature set.
A Cisco Unified Communications Manager gatekeeper-controlled trunk usually registers as an
H.323 gateway with the gatekeeper. Alternatively, it can beconfigured to register asa H.323
terminal. When a trunk isregistered. Cisco Unified Communications Manager provides this
information to the gatekeeper:
•
•
H.323 device type: Thedevice typecan be eithergateway or terminal. Cisco Unified
Communications Manager isusually configured to register asa gateway.
H.323 ID: The H.323 ID isbased on the name of the trunk that isconfigured inCisco
Unified Communications Manager with the string jr at theend. Thex is a number that
uniquely identifies each call-processing server of the cluster (that is,Cisco Unified
Communications Manager servers where Cisco CallManager service isactivated) that is
listed in the device pool that is assigned to the trunk.
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Step 2b: Apply SIP Profile to Trunk
The figure shows you how to applv the previously configured SIP profile to a trunk.
Step 2b: Apply SIP Profile to Trunk
trunk LonlquHlun
^^^^^B
;tr.-^i;- iOi'tis i-,g
n
4
.L
Apply SIP Profile.
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At the SIP trunk, set the SIP profileto the profilethat you createdearlier.
When the RSVP OverSIPparameter of the SIPprofile is set to Local QoS, or fall back to local
RSVP is enabled at the SIP profile, the SIP trunk needs to have an MRGL assigned sothatit
can allocate an RSVP agent for intracluster RSVP-enabled CAC. You can setthe MRGL
directK at the SIP trunk configuration page. If it is notsetat the trunk, you must setthe MRGL
at the device pool that is applied to the SIP trunk.
© 2010 Cisco Systems. Inc.
Bandwidth Managementand CAC Implementation
3-91
Step 2a: Configure SIP Profile
The figure shows how to configure SIP Preconditions settings at a SIP profile.
Step 2a: Configure SIP Profile
Check the checkbox
Chocse E2E from
to allowfatlbackto
the drop-down
local RSVP
list.
Tuinl Specific Confi»
Re'mjtt l-KcmiT
Trjnkbasedon' jpkever
RSVP Over SIP"
W Fill back Lclocal B5VP
SIP P.e ll XX Onboii"
Send PRACK f l . i Contains SDP
SetSIPRellXXOptbns
to Send PRACKif Ixx
Contains SDP
The necessary configuration for SIP Preconditions is applied to SIP trunks via SIP profiles. At
the SIP profile, you have to set the SIP RellXX Options parameter to Send PRACK if Ixx
Contains SDP.
Then you have to set RSVPOver SIP to E2E (end-to-end) whenyou want to enable SIP
Preconditions. If you want the trunk to use local QoS only, you wouldset the parameter to
Local QoS instead of to E2E.
WhenSIP Preconditions is configured (RSVPOver SIP is set to E2E),you can check the check
box fall back to local RSVP. This option allows a fallback to local QoS if the far end does not
support SIP Preconditions. If SIP Preconditions is supported by the far end and the RSVP
reservation fails, there is no fallback to local RSVP.
Note
When the other side ofthe SIPtrunk is CiscoUnified Communications Manager, there will
never be a fallback to local RSVP. SIP Preconditions is never considered to be unsupported
between two Cisco Unified Communications Manager clusters, regardless whether it has
been enabled at the other side or not. As a consequence SIP Preconditions always fails and
never falls back to local RSVP in such a scenario.
When the other side of the SIP trunk is Cisco IOS device—for example, Cisco Unified
Communications Manager Express—and end-to-end RSVP is not enabled at that remote
router, then a fallback to local RSVP is performed, if configured at the local Cisco Unified
Communications Manager cluster. Ifend-to-end RSVP is not configured on a Cisco IOS
device, SIP Preconditions is considered to be unsupported and therefore local fallback is
possible.
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SIP Preconditions Configuration Procedure
This subtopic describes the SIP Preconditions configuration procedure.
SIP Preconditions Configuration
Procedure
!
Follow standard procedure of RSVP-enabled locations.
a
Configure RSVP service parameters.
b
o
Configure RSVP agents in Cisco IOS Software.
Add RSVP agents to Cisco Unified Communications
d
Enable RSVP between location pairs.
e
f
Configure Media Resource Groups.
Configure Media Resource Group Lists.
Manager.
g
2
Assign Media Resource Group Lists to devices.
Configure SIP profileand apply SIP profile to trunk.
The configuration for SIP Preconditions is identical to theconfiguration of RSVP-enabled
locations. In addition to thestepsrequired for RSVP-enabled locations, youhave to configure
the SIP trunks that should use SIP Preconditions for end-to-end QoS.
Note
The RSVP agentthatis associated with the IPphone is usedfor the call leg to the far-end
SIP device. IfQoS fallback is not enabled, the SIP trunkwill never allocate an RSVPagent.
IfQoS fallback mode is enabled, two local RSVP agents are required in a fallback scenario:
one forthe IP phone and one forthe SIP trunk. Therefore, the MRGL at the SIP trunk is
required only for QoS fallback mode orfor when theSIP trunk is not configured for SIP
Preconditions at all but is configured to use local QoS.
Thefirst configuration step was described earlier inthe lesson and is notdescribed again.
Refer to the"Configuration Procedure for Implementing RSVP-Enabled Locations-Based
CAC" subtopic inthis lesson for a description of Step 1.
© 2010 Cisco Systems. Inc
Bandwidth Management and CAC Implementation
•
When preconditions fail on the far end, QoS fallback has no effect. Consequently, the call
continues without RSVP when the RSVP policy is Optional (Video Desired), and the call
fails when the RSVP policy is either Mandatory or Mandatory (Video Desired).
•
When receiving an INVITE with no preconditions and QoS fallback is off, the call fails
when the RSVP policy is Mandatory or Mandatory (Video Desired). When the RSVP
policy is Optional (Video Desired), the call continues without RSVP.
•
When receiving an INVITE with no preconditions and QoS fallback is on, the configured
RSVP policy is applied to local RSVP.
•
When receiving an INVITE with preconditions, and local QoS (instead of SIP
Preconditions) is configured at the receiving SIP trunk, the call fails when the received
RSVP policy is Mandatory. If the received RSVP policy is Optional and the local policy is
No Reservation, the call proceeds with no RSVP. If the received RSVP policy is Optional,
the locally configured policy is applied to local QoS.
If QoS fallback or local QoS configuration, the policies that are applied to local QoS are
managed the same way that they are managed for intracluster calls with RSVP-enabled
locations.
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©2010Cisco Systems, Inc.
Fallback from End-to-End RSVP to Local RSVP
This subtopic describes the fallback mechanism from end-lo-end RSVP to local RSVP when
the far end of the SIP trunk does not support SIP Preconditions,
Fallback from End-to-End RSVP to
Local RSVP
QoS fallback can be enabled to use local RSVP instead
of end-to-end RSVP when end-to-end RSVP is not
supported at the far end.
• Applicable only when SIP Preconditions is not supported by far
end (not on reservation failure).
* Call is reattempted without SIP Preconditions, in this case.
CAC reverts to local RSVP.
- RSVP policy configuredfor SIP Preconditions is used for local
RSVP
- Cluster-internal RSVP agents are used.
• Originating phone to its RSVP agent (no RSVP)
• RSVP agent oforiginating phoneto RSVP agent ofSIP
trunk (RSVP)
• RSVP agent tofarend (no RSVP)
You can configure QoS fallback to use local RSVP when end-to-end RSVP is notsupported by
the farend. fallback applies only in the casewhere the far enddoesnolsupport SIP
Preconditions. If il does supportSIP Preconditions and the RSVP reservation fails, there is no
fallback to local RSVP.
If there is QoS fallback, the call is reattempted without SIP Preconditions. CAC reverts to local
RSVP. uhich means that two cluster-internal RSVP agents arc used. The call is split into three
local call legs:
•
One from the originating phone to its RSVP agent
•
One from that RSVP agent to the RSVP agentthat is associated with the SIPtrunk
•
One from the RSVP agent that is associated with the SIP trunk, toward the othercall-
routing domain (where the same action con happen inthe ease ofa Cisco Unified
mm
Communications Manager)
However, the call leg between the two clusters or between the local cluster and the SIP device
on the other end does not use RSVP-based CAC.
The configured RSVP policy determines how calls areprocessed in certain scenarios:
•
•
When the far end does not support preconditions and QoS fallback is off, the call fails
when the RSVP policy is Mandatory, or Mandator,' (Video Desired). When the RSVP
policy is Optional (Video Desired), the call continues without RSVP.
When the farend does notsupport preconditions and QoS fallback is on,the configured
RSVP policy is applied to local RSVP.
>2013 Cisco Systems, Inc
Bandwidth Management and CAC Implementation
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SIP Preconditions Call Flow Summary
(Cont.)
Cisco Unified Communications
Manager Media component
initiates renegotiation of media
capabilities.
Media Request. forFull Sat
ot Codec Capabilities
Answ3witti SlnflteCode
NegotiatedbyMedia
Session is established (RSVP
bandwidth is adjusted if
necessary).
Session is established (RSVP
banOwidth Is adjusted if
necessary).
Whenthe call is answered, the terminating side sends an OK messagethat is confirmed from
the other side with an ACK message.
Now. where the call is formally set up. the terminating side triggersa renegotiation of media
capabilities with an INVITE message with no SDP attached.
Theoriginating sidesends an OK message, including the capabilities of the enddevice, in its
SDP.
The terminating side selects a codec and informs the originating side with an ACK message
with anattached SDP, including theselected capabilities (codec, packetization size, and soon).
Ifneeded. RSVP reservations areupdated between the two RSVP agents.
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SIP Preconditions Call Flow Summary
This subtopic illustrates a summarized call flow forSIP Preconditions calls.
SIP Preconditions Call Ffow Summary
QoS "Preconditon"
m=amJio 20O0G RTPfAVP 0
0=!NlP4 192022
m=auara10000 rtf/wP 0
e=MIP4 1920 21
a=currqose2ei
>
a=des qos mandatory e2e
a=wjrE.flose2e '.vi'-t
a=desqos mandatory e2a
^H.II.J.I Ii>Kiaa*gaa^ai
flendrecv
<
sendrecu
a^conl qos e2e recv
SIPUA1 initiates
RSVP reservation in
the 1 -» 2 direction
SIP UA2 initiates
RSVP reservation in
the 2 ^ 1 direction.
nvauOio 10000 RTPNWP 0
c=!NIP4 192021
a^currqos s2e';e--.i
allies qos mandatory e2e
sendiecv
7
m=audio 20000 RTP/AVP 0
C=INIP4192 02 2
a=curr qtra e2f ser **•;•.•
aides' qos mandalory e2e
•^f—BJSIfriJt'Jii'lWI
<i
sendrec*<
Precondition Complete
Precondition Complete
"fhe figure shows the most important components ofthe first phase ofthe call setup over a SIP
trunk that iseonligured for SIP Preconditions. The phase starts with the initial INVITE
message with the IP address of the originating RSVP agent and the request for RSVP CAC in
the SDP.
Then it show s the 183 response message that confirms the received RSVP CAC request inits
SDP. The SDP furtherincludes the IP address of the terminating RSVP agent and the request
for RSVP CAC for the reverse direction.
This negotiation isthen completed by the PRACK message that issent from the originating
side toward the temiinating side.
RSVP reser\ ations arethen setup In each RSVP agent for thedirection to theother RSVP
agent, using RSVP PATH and RSVP Resv messages.
The originating side then informs the temiinating side about the successful RSVP reservation in
the SDP of an UPDATH message. The terminating side confirms this information in anOK
message with SDP that includes the same status information for the other direction.
The precondition phase is now completed, and the terminating device can now send a
RINGING message to the originating side.
© 2010 Cisco Systems. Inc.
BandwidthManagement and CAC Implementation
3-85
SIP Preconditions Operation (Cont.)
After call is answered, terminating Cisco Unified
Communications Manager requests renegotiationof media
capabilities (INVITE, no SDP).
Originating Cisco Unified Communications Manager sends full
set of supported media capabilities (OKwith SDP).
Receiving Cisco Unified Communications Manager sends ACK
with SDP, including selected codec.
Ifselected codec has different bandwidth requirements, RSVP is
updated accordingly.
Call is established with three call legs:
- Originating phone to originating RSVP agent (no RSVP)
- Orig inating RSVPagent to terminating RSVP agent (RSVP)
- Terminating RSVP agent to terminating phone (no RSVP)
6. When the call isanswered, the terminating Cisco Unified Communications Manager
requests a renegotiation of media capabilities by sending a SIP INVITE message without
SDP.
7. The originating Cisco Unified Communications Manager responds with aSIP OK message
with SDP. Thecomplete set of supported media capabilities is included in the SDP.
8. The receiving CiscoUnified Communications Manager sends a SIPOK withSDP
message, including the selected codec. This codec is nowactually usedfor the end-to-end
call.
9. Ifthe selected codec has bandwidth requirements that are different from the requirements
that were used during the SIP Preconditions phase, the RSVP reservation is updated
accordingly.
10. The call is now established with three call legs (like with RSVP-enabled locations for calls
within a cluster):
—
The call leg between the originating IPphone and its RSVP agent, where no RSVPbased CAC was performed
—
The middle call legbetween thetwo RSVP agents, where RSVP-based CAC was
performed, as described earlier
—
The call leg between the terminating IP phone and its associated RSVP agent, where
againno RSVP-based CAC was performed
Note
Standard locations-based CAC is performed between the IP phones andtheir associated
RSVP agents As a result, thecall leg from theIPphone to itsRSVP agent iscounted
against themaximum bandwidth thatisconfigured at thelocations thatare applied to theIP
phone and to the RSVP agent.
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SIP Preconditions Operation
This subtopic describes the operation ofSIP Preconditions inCisco Unified Communications
Manager.
SIP Preconditions Operation
Phone places call to destination reachable via SIP trunk.
Originating Cisco Unified Communications Manager sends INVITE
messagewith SDP:
-• IP address of originating RSVP agent is used.
• RSVP is requested.
Terminating CiscoUnified Communications Manager sends SESSION
PROGRESS message with SDP:
IP address of terminating RSVP agent is used.
RSVP request is confirmed for forward direction.
RSVP request is sent for reverse direction
After confirmation (PRACK and OK), each RSVP agent attempts
RSVP reservation for its forward direction.
Ifsuccessful, standard call setup is performed (RINGING, OK, ACK)
A call with SIP Preconditions follows the message sequence of RFC 3312 to establish a
precondition. Here isa summarv' ofthe session establishment phases:
1. The originating IP phone places a call to adestination that isreachable through a SIP trunk.
According to the location configuration at the originating IP phone location, RSVP has to
beused between the location ofthe originating IP phone and the SIP trunk where the call
mm
should be routed to.
2, The originating Cisco Unified Communications Manager sends a SIP INVITE message
with Session Description Protocol (SDP). The IP address for the media stream in the SDP
is setto the IP address of the originating RSVP agent. RSVP is requested in the SDP.
3 The tenninating device—for example aCisco Unified Communications Manager server of
another cluster—responds with a SIP SESSION PROGRESS message with SDP. Itwill
provide the IP address of the terminating RSVP agent, confirm the RSVP request for the
forward direction, andsend an RSVP request forthe reverse direction.
4. The negotiation of SIP Preconditions for RSVP CAC is completed by SIP PRACK and OK
messages. Then each of the two RSVP agents attempts an RSVP reservation tor i+s forward
direction (that is. toward the other RSVP agent) ofthe preconditioned bandwidth.
5. Ifthe RSVP reservation issuccessful, a standard call setup isperformed by SIP RINGING,
OK. and ACK messages.
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Bandwidth Managementand CAC Implementation
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CAC with SIP Preconditions
This subtopic describes how you can implement end-to-end RSVP-based CAC for SIP trunks
with SIP Preconditions.
CAC with SIP Preconditions
For SIP calls going out of the cluster, RSVP can be used end-to-end
between different domains.
With Cisco Unified CommunicationsManager, RSVPagent of the
phone is used (not the one of the trunk).
Cisco IOS gateways and Cisco Unified Communications Manager
Express also support SIP Preconditions.
SIP Trunk
SCCP
RTP
• RSVP
_ Anarogor
Digital Voice
When both endsof a SIPtrunk support SIP Preconditions and the IP phone and the SIPtrunk
are in different locations and RSVP is enabled between thesetwo locations, then end-to-end
RSVP is used. Asa result, only the RSVP agent thatis associated with the IP phone is invoked;
there is nosecond local RSVP involved. TheRSVP agent of thephone now uses RSVP-based
CAC toward the other end of the SIP trunk.
Iftheother endis another Cisco Unified Communications Manager cluster, then the same result
happens at that farend: only one RSVP agent is invoked. If the otherend is a CiscoIOS router,
then that router (either Cisco Unified Communications Manager Express or a Cisco IOS SIP
gateway) terminates RSVP at the far end.
With SIP Preconditions, RSVP is now virtually end-to-end. Itspans the two call- routing
domains and is not limited to the local cluster.
Note
Due to proprietary extensions, SIP Preconditions for RSVP-enabled CAC iscurrently
supported only between Cisco Unified Communications Manager, Cisco Unified
Communications Manager Express, andCisco IOS SIP gateways. Third-party SIPdevices
are currently not supported.
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CAC Without SIP Preconditions
This subtopic describes how you can implement local RSVP-based CAC for SIP trunks when
SIP Preconditions is not used.
CAC Without SIP Preconditions
For SIP calls going out ofthe cluster, RSVP can be used only
between local RSVP agents associated with phone and SIP
trunk.
No end-to-end RSVP between separate domains.
When not using SIP Preconditions, you can use RSVP only within the local Cisco Unified
Communications Manager cluster. Such an implementation islike RSVP-enabled locations, as
discussed in an earlier topic ofthis lesson, except that the two devices that are involved in the
local Cisco Unified Communications Manager cluster arean IP phone and a SIP trunk (or two
SIP trunks).
The IP phone and the SIP trunk are in different locations, and RSVP is enabled between these
two locations. The IP phone refers to its RSVP agent by its MRGL, and the SIP trunk refers to
its RSVP agent b> its MRGL. RSVP CAC applies between these two RSVP agents. Because all
devices are local 'to the Cisco Unified Communications Manager cluster, this implementation
model is called local RSVP. Ifanother Cisco Unified Communications Manager cluster is atthe
other end of the SIP trunk, local RSVP can beused also at that end. However, the call leg
between the two RSVP agents that are associated with the SIP trunk ateach cluster isnol
subject to RSVP. Therefore, there isno end-to-end RSVP in this ease.
Ifthe other end ofthe SIP trunk isa third-party device, a Cisco IOS SIP gateway, orCisco
Unified Communicalions Manager Express, then local RSVP applies only tothe end ofthe SIP
trunk where Cisco Unitied Communications Manageris used.
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BandwidthManagement and CAC Implementation
SIP Preconditions
This topic describes SIP Preconditions and how it is used in Cisco Unified Communications
Manager to implement RSVP-based CAC for calls through SIP trunks.
SIP Preconditions Overview
Based on RFC 3312: Integration of Resource Management and SIP
- RFC allows for several types of precondition signaling.
- Cisco Unified Communications Manager currently supports RSVP
only.
lntercluster(also known as end-to-end) RSVP
-
Built on the capabilities of intracluster RSVP.
-
Allows RSVP to be used with devices outside the Cisco Unified
Communications Manager cluster.
- Supports RSVP CAC for calls over SIP trunks to
• Cisco Unified Communications Manager
• Cisco Unified Communications Manager Express
* Cisco IOS SIP gateway
• Cisco Unified Border Element
The Cisco Unified Communications Manager implementation of SIP Preconditions is based on
RFC 3312. Integration of Resource Management and SIP. The RFC describes severaltypes of
precondition signaling. Cisco Unified Communications Manager is currently supporting
precondition signaling for RSVP only.
SIP Preconditions appliesto SIPtrunks and henceappliesto calls going out of the cluster. Like
RSVP-enabled locations, it allows RSVPagentsto be used for calls through SIP trunks. It is
therefore also referred to as intercluster RSVP.
Another term that is used to refer to SIP Preconditions is "end-to-end RSVP." This term does
not mean that RSVPis implemented in the actualendpoints (IP phones),but it refers to
intercluster calls. Before SIP Preconditions, intercluster callsusing SIPwereableto useonly
local RSVP within a cluster. In thiscase, an RSVP agent that is associated with the IP phone,
and another RSVPagent that is associated with the SIP trunk, are used. Sucha configuration
requires the phone and the trunk to be in separate locations, and RSVP needs to be enabled
between these two locations. Thesetwo RSVPagents,however, were both local to the Cisco
Unified Communications Managerclusterand hence were not spanningto the other end of the
cluster. With SIP Preconditions, RSVP can be used between both ends of the SIP trunk; hence
the name end-to-end RSVP.
SIP Preconditions is not limited to intercluster trunks (that is, calls between two Cisco Unified
Communications Manager clusters). It can be used also for SIP trunks to Cisco Unified
Communications ManagerExpress. Cisco IOS gateways, and Cisco Unified Border Elements.
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Step 4: Configure Phones for AAR (Cont.
Forward calls to
Set (verify) exiemal
phone number mask
otdireclory numbef
voce mail if directory
Set individual destination number
number cannot be
mask (CFNB) to be used instead of
reached due lo
external phone number mask and
locai ms-based
AAR group prefix
CAC.
You need lo configure the IP phone directory numbers for AAR. The Directorv- Number
Configuration windowdisplays these relevantoptions:
•
•
Voice Mail: If this check box is checked, calls to this phone are forwarded to voice mail if
this directory numbercannotbe reached due to locations-based CAC.
AAR Destination Mask: If thisoption is set. the number where callsare rerouted to if this
directory number cannot be reached due to locations-based CAC iscomposed of this mask
and this director, number. Otherwise, the number would be composed of this directorv
number, the external phone number mask, and an AAR group prefix. Because 1his setting is
configured for each directory number, it allows any destination to be specified. (Ifthere arc
not n wildcard digits inthe mask, then calls arererouted to the specified number without
considering any digits ofthe directorv' number.) Therefore, this setting isoften referred lo
as CFNB.
•
AARGroup: An AAR group at the directory number has to be set in orderto allow AAR
calls to thisdirector.' number. The AAR group that is configured at thedirectorv number is
the destination AAR group.
•
External Phone Number Mask: This mask is the external phone number mask of the
directory number. Itshould always be set, because it is used by other features (such as digit
manipulation at route patternsor route lists).
© 2010 Cisco Systems. Inc
BandwidthManagement and CAC Implementation
3-79
Step 4: Configure Phones for AAR
The figure shows how to configure phones for AAR.
Step 4: Configure Phones for AAR
Cisco Unified Communications Manager Administration:
Call Routing > Phone
When you enable AAR on a phone, there are two possible settings in the Phone Configuration
window:
•
AAR CSS: This CSS is used ifa call that originated at this phone is rerouted using AAR.
•
AAR Group; The AAR group of the phone is the source AAR group, while the AAR
group that was set at the directory numberis the destination AAR group. It is important to
understand this distinction for the configuration of AAR prefixes,becausethey are
configured separately foreachpairof AAR source anddestination group. If no AAR group
is set at the phone, then the AAR group of the directory number is used as the AAR source
group for this phone.
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Step 3: Configure AAR Groups
As mentioned before. Step 2. "Configure partitions and CSSs." is not shown, because the
configuration of partitions and CSSs has been covered earlier. This figure illustrates Step 3. the
configuration of AAR groups.
Step 3: Configure AAR Groups
Cisco Unified Communications Manager Administration:
Call Routing > AAR Groups
Enler the name of me AAR group.
onfiguie the prefix that is added to the
external phone number mask of the callefl
phone for AAR calls within the same AAR
group
For each of the other AAR groups,
configure the dial prefix that is added to the
external phone number mask of the called
phone
Specify the dial prefix for bolh directions
(call from this AAR group to the other AAR
group and vice versa)
You configure AAR groups from Cisco Unified Communications Manager Administration
underCall Routing > AAR Groups. Each addedAAR group can be configured with a dial
prefix for itsown group and twodial prefixes foreachof the otherAARgroups (one forcalls
going to theother group andone forcallsbeingreceived from theothergroup).
Note
Inthis example, there are onlytwoAAR groups. For AAR calls from HQto BR a prefix of
0001 is used. For calls tn the other direction, a prefix of 901149 is used. The AAR
configuration that is shown would fit to a scenario where the HQ site is in Germany and the
BR site is in the United States The external phone number mask at both sites would use
national format.
As a result, an AAR call from Germany to the United States would be placed to 0001
followed by the national number (10 digits). 0 is the PSTN access code in Germany, 00 is
the international access code, and 1 is the country code for the United States. An AAR call
from the United States to Germany would be placed to the national number of a Germany
phonethat is prefixed with 901149. 9 is the PSTN access code inthe United States, 011 is
the international access code, and 49 is the country code of Germany.
Tip
The configuration thatisshown in thefigure does not useglobalized call routing. Globalized
call routing is recommended in larger multisite environments, especially ininternational
deployments. With globalized call routing, all sitesuse thesame AAR group and noprefixes
are required within thatgroup The external phone number mask is specified in globalized
format(E.164 number with + prefix).
) 2010 Cisco Systems. Inc.
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Step 1: Configure AAR Service Parameters
The figure illustrates how to enable AAR and how to set AAR-related parameters.
Step 1: Configure AAR Service Parameters
Cisco Unified Communications Manager Administration:
System > Service Parameters > Cisco CallManager
You enable AAR by setting the Cisco CallManager service parameter Automated Alternate
Routing Enable to True (False is default).
Other AAR-related service parameters aretheOut-of-Bandwidth Text parameter, where you
can set the text that is displayed on an IP phone when a call fails due to no available bandwidth,
and the AAR Network Congestion Rerouting Text parameter, whereyou can set the text that is
displayed on an IP phone when AAR reroutes a call.
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AAR Configuration Procedure
The implementation of AAR includes these steps.
AAR Configuration Procedure
1
Configure AAR service parameters (Cisco CallManager
service).
2
Configure partitions and CSSs.
3
Configure AAR groups.
A
Configure phones for AAR:
a
ApplyAARCSS and (source) AARgroup to IP phones.
& Configure IPphone directory numbers:
1) Apply (destination) AAR group.
2)
Set individual AAR destination mask (CFNB),
s,
Forward to voice mail if no bandwidth is available.
Step 2.-Configure partitions and CSSs." is notdiscussed in this topic, because the
configuration of partitions and CSSs was discussed in detail in the Implementing Cisco Unified
Communications Manager, Part 1 (CIP'1'1) course,and both of these itemshave already been
used se\eral limes in this course.
You need to precisely design partitions and AAR CSS. The AAR CSS of the calling device
must include the partition that is necessary to route the redirected call. Thecall is routed to the
number that iscomposed of the destination directory number, external phone number mask, and
AAR prefix (according tothe AAR group configuration). Ifyou configure an individual AAR
destination mask or forward to voice mail, the AAR CSS has to provide access to these
numbers (numbers that arccomposed of the called directory number and AAR destination mask
or voice-mail pilot number).
As mentioned earlier, in globalized call routing, AAR configuration is simpler when you use
the globalized format at theexternal phone number mask.
© 2010 Cisco Systems. Inc
BandwidthManagement and CAC Implementation
AAR Considerations
Ihis subtopic discusses important considerations when you are implementing AAR.
AAR Considerations
AAR supports these call scenarios:
- Call originates from an IP phone within one location and
terminates at an IP phone within another location.
- Incoming call through a gateway device within one location
terminates at an IP phone within another location.
AAR does not work with SRST:
- AAR is not activated by WAN failure.
- AAR requires CAC failure.
Use of globalized call routing simplifies AAR implementation in
international deployments.
AARdoes not support CTI route points as the origin or destination
of calls.
AAR does not workto Cisco Extension Mobility users roaming to
different sites.
AAR supports these call scenarios:
•
Call originates from an IP phone within one location and terminates at an IP phone within
another location.
•
Incoming call through a gateway device within one location terminates at an IP phone
within another location.
AAR does not work with Survivable Remote Site Telephony (SRST). AAR is activated only
after a call is denied by CAC, not by WAN failures.
Using globalized call routing simplifies the implementation of AAR substantially—especially
in international deployments.
AAR does not support CTI route points as the origin or destination of calls, and AAR is not
compatible with Cisco Extension Mobility for users who roam to different sites.
Note
When tail-end hop-off (TEHO) is used, itisimportant toconfigure AAR in such a way that
the local gateway is always used for callsbeing rerouted by using AAR. This automatically
occurs when you use local routegroups. When youare not using local routegroups, you
have to configure AAR CSS so that the focal gateway is used forAAR calls. Ifthe AAR CSS
refersto the TEHO gateway, AAR callswill fail, because the callleg to the (remote) PSTN
gateway again has the same issue that the initial call had: Itneeds to go overthe IP WAN
(which typically meansthat itgoes outofthe location ofthe originating phone), butdoing
thatis not possible becauseno bandwidth is left for the location (which was the reason why
the initial called ended up in a CACfailure).
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In the example, all phones are in the same AAR group (System). Noprefix is configured for
callswithin thissingle AAR group. There is a single route pattern in Ii.164 format: (\+!). The
route pattern refers to the only configured route list, which is configured to usethe local route
group. Each gateway is referenced from a site-specific route group. U.S. phones use a U.S.specific device pool with the local route group setto U.S., and German phones use a device
pool specific to theircountry. where the local route group refers to the DE route group. The
external phone number mask in globalized format is +15115222xxx at U.S. phones and
+4969125xxxx at Gemian phones. The AAR CSS is the same for both phones and provides
access to the '*+.! route pattern.
When a call from a U.S. phone to a German phone is notadmitted because of no available
bandwidth, theexternal phone number mask of the German phone is merged withthe directory
number ofthe phone (in thiscase, the result is +49691253001). No AAR prefix is added, so a
call isplaced to that number. It matches the \+.! route pattern, and the local route group isto be
used, fherefore. the call is sent to the U.S. gateway, where the called numbercan be localized,
using called-party transformation (that is. the number ischanged lo49691253001 with a
number type of international) settings thatare configured at the gateway.
The same thing happens for calls in the other direction. As a result,+15115552001 is called,
and after the called numberis localized at call egress—again provided by global
Iransformations at the gateway—a call with a number type of international is placed to
15115552001. this time through the Gennan gateway.
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Bandwidth Management and CAC Implementation
3-73
AAR Example with Local Route Groups and Globalized
Numbers
The figure shows an AAR examplewhere globalized call routing and local routegroups are
used.
AAR Example with Local Route Groups
and Globalized Numbers
,.
Site 2
Sitel
IP WAN
PSTN
Route PaBem;**!
1
Single Roule List
Default Local Route Group
DN" 2001
Ext Phone
Number Mask
+ 15115552XXX
DN": 3001
Single Route Pattern in One
Partition, Single AAR CSS, Single
AAR Group, No Prefixes Within AAR
Group
Ext. Phone
Number Mask
+4969125XXXX
' Unified CM = Cisco Unitied Communications Manager, " DN = directory number
If the AAR destination mask is entered in the globalized form, and if every AAR CSS is able to
route calls to destinations in the globalized form, then system administrators can forego the
configuration of AAR groups, because their sole function is to determine which digits to prefix
based on the local requirements of the PSTN access of the calling phone to reach the specific
destination. With globalized call routing, Cisco Unified Communications Manager can route
calls to the PSTN in E. 164 format with a + prefix. When you configure the external phone
number mask, in this format, no prefixes are required for AAR. To localize the called- and
calling-party numbers, implement global transformations for each egress PSTN gateway (like
for normal PSTN calls).
Without local route groups, the AAR CSS is used to route the call through the colocated
gateway of the calling phone by matching a site-specific route pattern that refers to a sitespecific route list, route group, and gateway. When local route groups and globalized call
routing are implemented, the egress gateway does not need to be selected by site-specific AAR
CSS. because the egress gateway is determined by the local route group feature.
In summary. \\ hen you are using globalized call routing with local route groups, AAR
implementation is extremely simple: Only a single AAR CSS and AAR group are required and
applied to all phones, regardless of their location.
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In the other direction, an AAR call from a German phone to a U.S. phone composes a dial
string of 00015115552001. which is the format that is used for inlemalional calls to the United
States. It matches the 0.! route pattern and is sent out using the German gateway.
In summary, this two-site example requires two route patterns in different partitions, two AAR
CSSs. and two AAR groups. In a large, worldwide deployment with lots of different numbering
plans, the configuration of AAR groups can be relatively complex.
)2010 Cisco Systems, Inc
Bandwidth Management and CAC Implementation
AAR Example Without Local Route Groups and Globalized
Numbers
The figure shows an example of AAR when globalized call routing and local route groups are
not being used.
AAR Example Without Local Route
Groups and Globalized Numbers
Second call fails locations-based CAC
UnifiedCW
Site 1 (+1):
international
Dialing: 9.011
and should use PSTN.
Location Configuration
Hu6_None. Unlimited
Site 2 (+49):
International
Dialing: 0.00
BR-LOC 24 kb/s
AAR Group Configuration.
ON" 2001
Prefix U.S. lo DE. 901149
DN": 3001
E<t Phone
Prefix DE to U S.. 0001
Ext Phone
Number Mask:
511S552XXX
Number Mask:
Two Roule Paltems in Different
69125XXXX
Partitions, Two AAR CSSs,
Two AAR Groups
' Urafied CM - Cisco Unified Communications Manager, " DN = directory number
There are two sites, one in the United States and the other one in Germany (country codes 1 and
49). At site 1 (country code 1) the access code is 9, at site 2 (country code 49) the access code
is 0. Both countries use 10-digit numbers. There are two route patterns(9.@ for site 1 and 0.!
for site 2). Each route pattern is in a site-specific partition, and the phones use site-specific
CSSs.
Fromthe perspective of AAR, U.S. phones are configured with a 10-digitexternal phone
number mask, and phones at Germany also use national format for the external phone number
mask. U.S. phones are in AAR group U.S., and German phones are in AAR group DE. AAR
prefixes are configured in this way:
•
Prefix from AAR group U.S. to AAR group DE: 901149
•
Prefix from AAR group DE to AAR group U.S.: 0001
The AAR CSS of U.S. phones has accessto the 9.@ roule pattern; the AAR CSS of German
phones has access to the 0.! route pattern.
When a call from a U.S. phone to a Gennan phone is not admitted because of no available
bandwidth, the extemal phone numbermask of the German phone is merged with the DN of the
phone (in this case, the result is 691253001). Then the prefix 901149configured from AAR
group U.S. to DE is appended, resulting in a call to 901149691253001, which is processed by
the 9.(S) route patternthat refersto the U.S. gateway.
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AAR Characteristics
This subtopic describes the characteristics of AAR.
AAR Characteristics
• Provides a fallback mechanism for calls denied by CAC:
- Reroutes calls over PSTN.
- Works only for locations-based CAC (standard locations
and RSVP-enabled locations).
• Works only for cads placed to internal directory numbers.
• Alternate number is composed of dialed directory number, a
prefix configured per AAR source and destination group, and
the external phone number mask of the called device:
- AAR destination mask can be configured per device
(a.k.a. Call Forward No Bandwidth [CFNB]) to reroute
calls to other phone numbers, such as cell phones.
- Forward to voice mail can be configured per device to
reroute calls to voice mail.
AAR pro\idesa fallback mechanism for callsthat are denied by locations-based CAC or
RSVP-cnablcd locations-based CAC bv rerouting calls over the PSTN in the event of CAC
failure.
AAR works only for calls that arc placed to internal directory numbers. It docs not apply to
calls that are placed to route patterns or feature patterns such as Meet-Me or Call Park.
Ho\\e\ er. it does work for hunt pilots and computertelephony integration (CTI)ports.These
entities can be configured with an AAR group andan AAR calling search space (CSS).
The alternate number that is used for the PSTN call is composed of the dialed directory
number, a prefix that is configured perAAR source and destination group, and theexternal
phone number mask of the called device.
Alternatively, calls can berouted tovoice mail, oryou can configure an AAR destination mask
for each device that allows an> numberto be used for the rerouted call. The numberthat is
specified at the AAR destination mask isalso known as the Call Forward No Bandwidth
(CFNB) destination.
Note
AAR is a fallback mechanism for calls that are denied by locations-based CAC or RSVPenabled locations-based CAC. It does not apply to calls that are denied by gateways d le to
exceeding the available or administratively permitted number ofchannels, or to calls tut
have been rejected on trunks (for example, ongatekeeper-controlled H.225 or intercluster
trunks) If suchcalls fail (for whatever reason), fallback mechanisms are provided by route
lists and route groups.
) 2010 Cisco Systems, Inc
Bandwidth Management and CAC Implementation
Automated Alternate Routing
This topic describes how to implement AAR in Cisco Unified Communications Manager.
AAR Overview
Headquarters
Branch B
ys Cisco Unrfied CommunicationsManager CACblocks a call over the IP WAN.
9 The call isautomatically rerouted over thePSTN.
AAR allows calls to be rerouted through the PSTN by using an alternate number when Cisco
Unified Communications Manager blocks a call due to insufficient location bandwidth. With
AAR. the caller does not need to hang up and redial the called party. Without AAR, the user
wouldget a reordertone and the IP phone woulddisplay "Not enoughbandwidth."
AAR applies to centralized call-processing deployments. For instance, ifa telephone in a
company headquarters calls a telephone in branch B and the available bandwidth for the WAN
linkbetween the branches is insufficient (as computed by the locations mechanism), AAR can
reroute the call through the PSTN. The audio path of the call would be IP-based from the
calling phone to its local (headquarters) PSTN gateway, time-division multiplexing (TDM)based from thatgateway through the PSTN to the branch B gateway, and IP-based from the
branch B gateway to the destination IP phone.
AAR is transparent to users. Itcan beconfigured so that users dial only theon-net (for
example, four-digit) directory number of the called phone. (Noadditional userinput is required
to reach the destination through an alternate network such as the PSTN.)
In the examplethat is shown here, a call is placed from PhoneA to Phone B, but the locationsbased CAC denies the call due to insufficient bandwidth. Cisco Unified Communications
Managernow automatically composes the required route patternto reach Phone B via the
PSTN and sends the call off-net.
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Afteryou click Save,the changes are displayed in the Location RSVP Settings section of the
window. Only the locations thatarcnotconfigured to use the system default arelisted.
Note
You can also enable RSVP within a location. For the currently configured location, Use
System Default is not an option. Youcan choose only No Reservation, Optional (Video
Desired), Mandatory, or Mandatory (VideoDesired)within a location. The defaultfor calls to
own location is No Reservation, and to all other locations, the default is Use System Default.
Note
When RSVP-enabled locations are used, it is extremely important that the phones use the
appropriate RSVP agent. Note thattherewill be threecall legs, phone to its RSVP agent;
that RSVP agent to another, remote RSVP agent; and finally, that remoteRSVP agent to its
phone.
How does Cisco Unified Communications Manager determine which RSVP agent is the
RSVP agent to be used by a given phone?The selection of the RSVP agent is based solely
on the MRGLs that are assigned to the phones that attempt to establish a call. Errors in the
MRGL configuration can resultinsuboptimal traffic flows. Therefore, whenyouimplement
RSVP-enabled locations, you must properly assign phones to RSVPagents by using
MRGLs and MRGs. The sample scenarioat the beginning ofthis configuration subtopic
provided all the information that is needed for assigning the RSVP agentstothe phones
Theappropriate configuration is notshown here becausethe configuration ofMRGLs and
MRGswas covered in detail in the Implementing Cisco Unified Communications Manager.
Part 1 (CIPT1) courseand becauseMRGLs and MRGs havealready beendiscussed inthis
course.
©2010 Cisco Syslems. Inc.
Bandwidth Management and CAC Implementation
3-67
Step 4: Enable RSVP Between Location Pairs
The figure illustrates how to enable RSVP between pairs of locations.
Step 4; Enable RSVP Between Location
Pairs
After configuring the RSVPagentsin Cisco IOS routersand adding them to Cisco Unified
Communications Manager, you need to enable RSVP between one or morepairsof locations.
You performthis task in the Location Configuration window, whichyou access from Cisco
Unified Communications Manager Administration by choosing System > Location. Choose
the location for which RSVP should be enabled for calls to one or more other locations. In the
Location Configuration window underModify Setting(s) to OtherLocations, the currently
configured location is listed. All other locations are also listed.
Choose the location to which RSVP should be used, and then choose the RSVP setting. You
will find the same options thatyou found at the Default interlocation RSVP Policy service
parameter:
•
\o Reservation: No RSVP reservations are made between any two locations.
•
Optional (Video Desired): A callcanproceed as a best-effort, audio-only callif failure to
obtain reservations forboth audio and video streams occurs. The RSVP agentcontinues to
attempt an RSVP reservation for audio and informs Cisco Unified Communications
Manager if the reservation succeeds.
•
Mandatory: Cisco Unified Communications Manager does notringtheterminating device
until RSVP reservation succeeds for the audiostream and, if the call is a video call, for the
video stream as well.
•
Mandatory (Video Desired): A video call can proceed as anaudio-only call ifa
reservation for the audio stream succeeds but a reservation for the video stream does not
succeed.
Inaddition, there is theoption Use System Default, which applies thevalue of the Default
interlocation RSVP Policy service parameter for calls to the chosen location.
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©2010 Cisco Systems, Inc.
Step 3: Add RSVP Agents to Cisco Unified Communications
Manager
The figure shows how to add an RSVP agent in Cisco Unitied Communications Manager.
Step 3: Add RSVP Agents to Cisco
Unified Communications Manager
Cisco Unified Communications Manager
Administration:
Media Resources > Media Termination Point
•tAilt-n.-iaoifcrtTrB"
ttuioaon
;,SCO105 Fnh.nr.H 5«ft««« PWd.aT.rni.njtion 1 .
H« 1 HSiF lot 1
•
^\
Choose media termination point type and device
pool. Enter media termination point name and
description.
Note The configured media termination point name has to matctittie name assignefl lothe
media resource at the Cisco IOS router
After configuring the RSVP agent function at theCisco IOS gateway, youneed to add the
corresponding media resource toCisco Unified Communications Manager. In Cisco Unitied
Communications Manager, choose Media Resources > Media Termination Point and click
Add New.
Inthe Media Termination Point Configuration window, choose the type of the MTP (currently
there is only oneoption. Cisco IOS Enhanced Software Media Termination Point), enter a
name anddescription, and then choose the device pool that should be used.
Note
The name of the MTP has to match the name that was configured at the Cisco IOS router
with the associate profile idregister command entered inseep ccm groupidconfiguration
mode. The name is case-sensitive.
Note
Because RSVP-enabled locationsallow RSVPto be used between two RSVPagents lat
are betweenthe twoendpoints ofa call, at least two RSVP agents have to be configured in
a cluster to make itwork. Inthe example, these agents are HQ-1 and BR-1 The figure that
is shown with this step is an example that uses the HQ-1 router.
***w
© 2010 Cisco Systems. Inc.
Bandwidth Management and CAC Implementation
3-65
Note
The bandwidth that is reserved for a call depends on the codec that is used. As with
standard (non-RSVP-enabled) locations, it is 80 kb/s for G.711 and 24 kb/s for G.729.
During the call setup, however, the RSVP agent will always request an additional 16 kb/s,
which is released immediately after the RSVP reservation is successful. Therefore, the
interface bandwidth has to be configured in such a way that it can accommodate the desired
number of calls (considering the codec that will be used) plus the extra 16 kb/s. If, for
example, two G.729 calls are permitted on the interface, 64 kb/s must be configured; for two
2 G.711 calls, 176 kb/s is required.
In the example, only one G.729 call is permitted.
Note
Because RSVP-enabled locations allow RSVP to be used between two RSVP agents that
are between the two endpoints of a call, you need to configure at least two RSVP agents in
a cluster to make it work. In the example, these agents would be HQ-1 and BR-1, The figure
that is shown with this step is an example that uses the HQ-1 router.
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Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Step 2: Configure RSVP Agents in Cisco IOS Software
The figure shows how to configure a Cisco IOS router to enable RSVP-agent functionality.
Step 2: Configure RSVP Agents
in Cisco IOS Software
:p local FastBtheraetO/C
;p ccp 10.1.1.1 identifier 1
;p ccm group 1
SBOciate ecu 1 priority 1
Buociatf profile
1 register hq-1_mtf
Pass-through codec is
used, which allows a
CiscolOS Software
MTP to be used.
MTP is used as RSVP
agent.
Software MTP is used;
interface SerialO/1
description
IP-WAN
ip address 10.1.4.101 255.255.255.
duplex sutc
speed auto
ip rsvp bandwidth 4C
no hardware DSPs are
required.
RSVPis enabled on
interface; permitted
bandwidth is specified.
As with other mediaresources that arc provided by Cisco IOS Software (conference bridges
and transcoders). the configuration starts with global Skinny Client Control Protocol (SCCP)
settings, followed by the Cisco Unified Communications Manager group configuration. In
configuring the media resource itself (which you perform indspfarm profile configuration
mode). \ou use three commands that arc specificto the implementation of a software MI P
RSVP agent:
• codec pass-through: This command specifies that the actual content ofthe RTP stream is
not modified. Mediaresources usually have to interpret and modify the audio stream:
examples are transcoders that change the codec ofthe audio stream, orhardware MTPs that
are used to convertout-of-band signaling to in-hand dual tone multifrequency (DTMF).
The RSVP agent repackages RTP only at Layer 3 and Layer 4. Itterminates the incoming
call leg by dc-encapsulating RTP and then re-encapsulating the identical RTP into a new-
call leg. Because this simple repackaging does not require interpreting and modifying the
audio payload (which isrequired with transcoders or hardware MTPs that are used for
DTMF). the router can perfonn this function in software.
* rsvp: This command specifics that this MTP isused as an RSVP agent that will be used to
set up a call leg to another RSVP agent where RSVP with IntServ over DiffScrv has to be
used.
•
maximum sessions software sessions: This command specifies the maximum numberof
sessionsfor the mediaresource. Note that the keyword software has been used. This
keyword indicates that this RSVP agent should nol use digital signal processors (DSPs) but
that itshould perform its function in software. You can use software MTP only when codec
pass-through has been configured.
After setting up the Ml"P RSVP agent, you need toenable RSVP on the WAN interface or
interfaces b\ using the ip rsvp bandwidth bandwidth command. The specified bandwidth
determines how much bandwidth can be reserved by RSVP.
© 2010 Cisco Systems. Inc.
Bandwidth Management and CAC Implementation
3-63
Step 1: Configure RSVP Service
Parameters (Cont.)
Cisco Unified Communications Manager Administration:
System > Service Parameters > Cisco CallManager
t
"* *.r
-l^.. j_f-i^...—K£urj
_
Set default interlocation RSVP
- r- *
. ••>*,
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-t
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. .
rr-
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enor-handling option.
_S«t
or**,* DSCP<OO00O0l
v d6T*ultDS
0c**uK DSCP10CQ00D)
* d»<Jult 05
Set DSCP values for calls where
RSVP failed (either immediately at
Ihe cal setup or m id-call).
The figureshows the configuration of the previously mentioned serviceparameters, as well as
the service parameters that are used to set the differentiated servicescode point (DSCP) values
thatshould be used forthe RTP packets of callsfor which RSVP failed. These parameters can
be audiochannel (for which RSVPfailed at the call setup if the policywas set to Optional) or a
videoor audiochannel(if the RSVPfailure occurred mid-call and the Mandatory RSVPmidcall error handle option is set to Call becomes best effort).
Youcan configure all these service parameters from Cisco Unified Communications Manager
Administration by choosing System > Service Parameters under the Cisco CallManager
service. All these parameters are clusterwide parameters; that is, they apply to all serversof the
cluster that is running the Cisco CallManager service.
3-62
Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0
>2010 Cisco Systems, Inc.
The table shows the interactions of these policy settings.
RSVP Policy
Mandatory
When Reservation
Mandatory
Mandatory
Behavior or Call
Failure (Non-Multilevel
RSVP
RSVP
Result
Precedence and
Mid-Call Error
Mid-Cail
Preemption [MLPP])
Handle
Retry
Occurs
Option
Counter
Audio or video RSVP
Call rejected.
—
—
failure in initial call setup
Mandatory
Audio or video RSVP
failure in mid-call
following retry
Call released if
reservation does not
counter
succeed after n retries.
Call fails
n
exceeded
Mandatory
Audio or video RSVP
failure in mid-call
Call becomes
best effort
Mandatory
(Video Desired)
Audio RSVP failure in
Mandatory
Video RSVP failure in
(Video Desired)
initial call setup
Mandatory
(Video Desired)
Audio RSVP failure in
Call fails
mid-call
following retry
—
n
Call proceeds as best
effort, and reservation
is retried infinitely.
—
Call rejected.
~ -
Call proceeds as audio-
initial call setup
—
only call.
n
counter
exceeded
Mandatory
(Video Desired)
Video RSVP failure in
Mandatory
Audio RSVP failure in
(Video Desired)
mid-call
Call proceeds as audioonly call.
—
—
initial call setup
Call becomes
best effort
Call released if
reservation does not
succeed after n retries.
n
Call proceeds, audio
stream in call becomes
best effort if reservation
does not succeed after
n retries.
Mandatory
Video RSVP failure in
Call fails
(Video Desired)
mid-call
following retry
n
Call proceeds as audioonly call.
counter
exceeded
Mandatory
Video RSVP failure in
(Video Desired)
mid-call
Call becomes
best effort
n
Call proceeds; video
stream in call becomes
best effort if reservation
does not succeed after
n retries.
Optional (Video
Desired)
Call proceeds as audioonly call with best
Audio or video RSVP
failure in initial call setup
effort; RSVP is trie.,
until reservation
succeeds or call is torn
down
Optional (Video
Desired)
Call proceeds as audioonly call; if audio
Audio or video RSVP
failure in mid-call
stream had reservation
failure, it becomes best
effort.
Mandatory
Audio or video RSVP
failure in initial call setup
) 2010 Cisco Systems, Inc
Call rejected.
—
~
BandwidthManagement and CAC Implementation
3-61
RSVP Retry Timer: This parameter defines the interval (in seconds) after which the
RSVP agent will retry the reservation if there is a failure. If you set this parameter to 0, you
disable RSVP retry on the system. If the RSVP policy is optional, the call can still proceed
even if there is RSVP failure during call setup time. An RSVP failure indicates insufficient
bandwidth at the time of setup, so the call is likely to begin with poor voice quality.
However, this condition may be transient, and the automatic reservation retry capability
may succeed during the course of the call, at which point adequate bandwidth will be
ensured for the remainder of the call. The icon or message that is displayed to the user
initially should convey something like, "Your call is proceeding despite network
congestion: you may experience impaired audio quality; if this condition persists, you may
want to try your call again later." If reservation retry succeeds, the icon or message should
be removed or should be replaced by one that conveys a return to normal network
conditions and ensured audio quality.
Mandatory RSVP Mid-Call Retry Counter: This parameter specifies the RSVP mid-call
retry counter when the RSVP policy specifies Mandatory and when the mid-call error
handle option "call fails following retry counter exceeded" is set. The default value
specifies one time. If you set the service parameter to-1, retry continues indefinitely until
either the reservation succeeds or the call gets torn down.
Mandatory RSVP Mid-Call Error Handle Option: This parameter specifies how a call
should be processed if the RSVP reservation fails during a call. You can set this service
parameter to these values:
—
Call Becomes Best Effort: If RSVP fails during a call, the call becomes a besteffort call, if retry is enabled, RSVP retry attempts begin simultaneously.
—
Call Fails Following Retry Counter Exceeded: If RSVP fails during a call, the call
fails after n retries of RSVP if the Mandatory RSVP Mtd-Call Retry Counter service
parameter specifies n.
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Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Step 1: Configure RSVP Service Parameters
Ihis step describes RSVP service parameters and how to configure them.
Step 1: Configure RSVP Service
Parameters
Default Interlocation RSVP Policy:
• No Reservation: No RSVP resen/ations are made between locations.
• Optional (Video Desired): Iffailure to obtain reservations for both audio and video
streams occurs, an audio-only call will be placed with best-effort service.
• Mandatory: Both audio and video (if wdeo call) reservations must succeed for
Cisco Unified Communications Manager to ring terminating device.
• Mandatory (Video Desired): A video call can proceed as an audio-only call if a
reservation (or the audio stream succeeds but a reservation for the video stream
does not succeed
RVSP Retry Timer: Defines how often (in seconds) the RSVP agent
will retry the reservation if there is a failure.
Mandatory RSVP Mid-Call Error Handle Option: Ifa mid-call failure
occurs, defines whether call becomes best effort or fails (after n
retries).
Mandatory RSVP Mid-Call RetryCounter: Defines the n tries for midcall error processing.
You can configure these important RSVP service parameters:
•
Default Interlocation RSVP Policy: This parameter sets the clusterwide default RSVP
polic\. You can set thisservice parameter to oneof these values:
—
No Reservation: No RSVP reservations arc made between any two locations.
—
Optional (Video Desired): Acall can proceed asa best-effort, audio-only call if
failure to obtain reservations for both audio and video streams occurs. The RSVP
agent continues toattempt an RSVP reservation Tor audio and informs Cisco Unified
Communications Manager if the reservation succeeds.
—
Mandatory: Cisco Unified Communications Manager docs notring the terminating
device until RSVP reservation succeeds for the audio stream and, if the call is a
video call, for the video stream as well.
—
Mandatory (Video Desired): Avideo call can proceed as an audio-only call ifa
reservation for the audio stream succeeds but a reservation for the video stream does
not succeed.
© 2010 Cisco Systems. Inc
Bandwidth Managementand CAC Implementation
3-59
Example: RSVP-Enabled Locations Configuration
The figure shows an example of RSVP-enabled locations-based CAC implementation.
Example: RSVP-Enabled Locations
Configuration
Location BR
Location HQ
HQ
10.1.1.1
Cisco Unified
Communications Manager
BRMRGL
1
BR RSVP
MRG
l^VtfWt
BR
In the example, there are two sites: headquarters (HQ) and branch (BR). Phones that are located
in the headquarters are in location HQ, and phones that are located at the branch are in location
BR. RSVPagentsexist at each site (HQ-1_MTP is provided by router HQ-1, and BR-1_MTP is
provided by router BR-1). The RSVPagentsare assigned to their respective locations.
Headquarters phones have the MRGL HQ MRGL applied; this MRGL includes the MRG
HQ RSVP_MRG, which includes the HQ-1MTP RSVP agent media resource. Branch phones
have the MRGL BR MRGL applied; this MRGL includes MRG BR RSVP_MRG, which
includes the BR-1_MTP RSVP agent media resource.
Regions (not shown in the figure) are configured in such a way that G.729 has to be used for
calls between headquarters phones and branch phones.
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Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Configuration Procedure for Implementing RSVP-Enabled
Locations-Based CAC
To implement Cisco Unified Communications ManagerRSVP-enabled locations, you will need
to follow these steps.
Configuration Procedure for implementing
RSVP-Enabled Locations-Based CAC
1
2
3
Configure RSVP service parameters.
Configure RSVP agents in Cisco IOS Software.
Add RSVP agents to Cisco Unified Communications
4
Enable RSVP between location pairs.
Manager,
5
Configure Media Resource Groups.
6
r
Configure Media Resource Group Lists.
Assign Media Resource Group Lists to devices.
Because the implementation of Media Resource Groups (MRGs) and MRGLs has been
discussed in detail in the Implementing Cisco Unified Communications Manager, Part I
(CIPT1) course and has been used in earlier lessons of this course, only Steps 1 lo 4 are
discussed in this topic.
) 2010 Cisco Systems, Inc
Bandwidth Management and CAC Implementation
Here are other RSVP messages:
•
Krror and confirmation messages: Reservation-request acknowledgment messages are
sent as the result of the appearance of a reservation-confirmation object in a reservationrequest message. This acknowledgment messagecontains a copy of the reservation
confirmation. An acknowledgment message is sent to the unicast address of a receiver host,
and the address is obtained from the reservation-confirmation object. A reservation-request
acknowledgment messageis forwarded to the receiverhop by hop to accommodate the
hop-by-hop integrity-check mechanism.
—
Path-error messages result from Path messages, and they travel toward senders.
Path-error messages are routed hop by hop using the path state. At each hop, the IP
destination address is the unicast address of the previous hop.
—
Reservation-request errormessages result from reservation-request messages and
tra\ cl toward the receiver. Reservation-request error messages are routed hop by hop
using the reservation state. At each hop, the IP destination address is the unicast
address of the next-hop node. Here is some of the information that can be carried in
error messages:
•
•
Admission failure
•
Bandwidth unavailable
•
Service not supported
•
Bad flow specification
•
Ambiguous path
Teardown messages: RSVP teardownmessages remove the path and reservation state
without waiting for the cleanup timeout period. Teardown messages can be initiated by an
application in an end system (sender or receiver) or a router as the result of state timeout.
RSVP supports two types of teardown messages:
—
Path-teardown: Path-teardown messages delete the path state (which deletes the
reservation state), travel toward all receivers downstream from the point of
initiation, and are routed like Path messages.
—
Reservation-request teardown: Reservation-request teardown messages delete the
reservation state, travel toward all matching senders upstream from the point of
teardown initiation, and are routed like corresponding reservation-request messages.
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Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0
© 2010CiscoSystems, Inc.
How RSVP Works
The figure reviews the basic operation of RSVP.
How RSVP Works
Resv
•; ( Dest 10505050 !!
Res
/
!
Res*
; Dea '.320 20 20
Res.
3est 10 10 10 10
„.„.
Dest 10
,„„
30 30 30
. Desl 10 30 30 30 .
\NHop 10 SO 60 8'
NHop 10 50 50 5o]!\NHop 10 SOSOSrJ
\MHap '330 33 30
In Hot. 1020 202\
= RSVP processing occurs
= BanAwitltti is reserved on interface
As shown in the figure, the RSVP-enabled sender (in this case, an RSVPagent)sends a Path
message towardthe RSVP-enabled receiver(again,an RSVPagent in this case) along the path
that requests bandwidth for the call to be set up. The receiver responds with a Resv message
that is routed back along the path. Each RSVP-enabled device cheeks to see if the requested
bandwidth is available and sends the appropriate information in the downstream path toward
the sender.
If no RSVP-enabled device on the path had lo deny the reservation because of insufficient
bandwidth, the reservation was successful: the call was admitted by RSVP CAC.
Here is a more detailed description of the key RSVP messages:
•
Path messages (Path): An RSVP Path message is sent by each sender along the unicast or
multicast routesthat are provided by the routing protocol. A Path messageis used to store
the pathstate in each node. The pathstateis used to route Resv messages in thereverse
direction.
•
Reservation-request messages(Resv): A reservation-request message is sent by each
receiver host toward the senders. This message follows in reverse the routes that the data
packets use. all the way to the sender hosts. A reservation-request message must be
delivered to the sender hosts so that the hosts can setup appropriate traffic-control
parameters forthe first hop. RSVP does notsend any positive acknowledgment messages.
) 2010 Cisco Systems. Inc
Bandwidth Management and CAC Implementation
3-55
Characteristics of RSVP Agent-to-RSVP Agent Call Leg
This subtopic describes the characteristics of the call leg between two RSVP agents.
Characteristics of RSVP Agent-to-RSVP
Agent Call Leg
• Based on standard Cisco IOS RSVP.
• IP network between RSVP agents is RSVP-enabled.
• Each interface is configured with maximum bandwidth to be
reserved by RSVP.
- If RSVP is not enabled on any hop in the path, the
appropriate link is ignored by CAC algorithm.
• IntServ and DiffServ models are used:
- RSVP only for CAC (control plane)
- LLQ for QoS (data plane)
• Call is set up only after successful RSVP CAC.
The call leg between two RSVP agenLs uses standard RSVP, as implemented in Cisco IOS
routers. The IP network between the RSVP agents is RSVP-enabled. In other words, each
interface is configured with a maximum amount of bandwidth that can be used for RSVP calls.
When not enough bandwidth is available end-to-end (between the two RSVP agents, in this
case). RSVP CAC denies the call,
If RSVP is not enabled on any hop in the path, the appropriate link is ignored by the CAC
algorithm {that is. it is always admitted on this link).
Cisco Unified Communications Manager RSVP agent CAC uses the Integrated Services
(IntServ) and Differentiated Services (DiffServ) models for the RSVP call leg. In other words.
RSVP is used only for CAC (the "control" plane), and not with RSVP-reservable queues for
providing QoS to the streams. Instead, standard low-latency queuing (LLQ) configuration is
required to provision QoS for the voice stream (the "data" plane).
The end-to-end call—that is, the incorporation of all three call legs—is established only after
the RSVP call leg has been admitted. If the RSVP call leg is not admitted, the call fails due to
CAC denial (not enough bandwidth).
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ImplementingCisco Unrfied Communications Manager, Part 2 (CIPT2)v8.0
)2010 Cisco Systems, Inc.
Characteristics of Phone-to-RSVP Agent Call Legs
This subtopic describes the characteristics of the call legs between phones and RSVP agents.
Characteristics of Phone-to-RSVP
Agent Call Legs
* Based on Cisco Unified Communications Manager locations:
- Usually phones and their RSVP agents are in the same
location.
If phone and its RSVP agent are in separate locations,
standard location-based CAC is performed for this call leg.
• Phones have to use their RSVP agent:
- RSVP agent that is used by a certain phone should be as close
as possible to the phone,
- RSVP agent to be used by a certain phone is determined by
the Media Resource Group List of the phone.
• RSVP agent is an MTP:
- Pass-through codec is supported:
• No changes to RTP payload
• Allows secure RTP to be used
Standard locations algorithms apply lo the call leg between an IP phone and its RSVP agent,
which are usually in the same location. If they are in separate locations, standard locationsbased CAC is performed for this call leg (phone to RSVP agent) first. The two RSVP agents
will tr> to set up their call leg by using RSVP only if enough bandwidth is available for the IP
phones to reach their RSVP agents.
An RSVP agent registers with Cisco Unified Communications Manager as a special MTP
device. Cisco Unified Communications Manager uses the Media Resource Group List (MRGL)
of the IP phone to determine which RSVP agent is to be used by which IP phone. The
association of a phone to its RSVP agent does not occur as the result of a search for an RSVP
agent in the same location of the phone. As mentioned earlier, the IP phone and its RSVP agent
can be in separate locations. Only MRGLs arc used to identify the RSVP agent to be used by an
IP phone.
From a design perspective, the RSVPagent that is used by a certain IP phone or group of
phones should be as close as possible to the IP phone or phones. Such a design ensures that
there are optimal pathswhere the phones ideallydo not use the IP WANto accesstheir RSVP
agents. This design also ensuresthat RSVPis used at the IP WAN and that the call legs that do
not use RSVP utilize only LAN infrastructure.
1he RSVP agent supports pass-through codec configuration, which allows any codec to be used
(the codecdocs not have to be known or supported by the RSVP agent). Pass-through codec
configuration includes Secure Real-Time Transport Protocol (SRTP), where the RTP payload is
encrypted.
) 2010 Cisco Systems, Inc
Bandwidth Management and CAC Implementation
3-53
Three Call Legs with RSVP-Enabled Locations
When RSVP-enabled locations are used, the end-to-end call is split into three separate call legs.
Three Call Legs with RSVP-Enabled
Locations
Location A
Location B
.••'
..-'
Cisco Unified''-.. ""•..
SCCP or
SCCP or ..•''*..-•*' Communications ''''..''••-._ SCCP
'--'"..-•SCCP
Mana9er
SCCP-C'V
RSVPAgentl to RSVPAgent2: RSVP-RTP
— SCCP or SIP
•*•
RSVP
**
RTP
In the figure, Phonel. which is in Location A, places a call to Phone2, which is in Location B.
The Cisco Unified Communications Manager location configuration specifies that RSVP has to
be used for calls between these two locations.
Cisco Unified Communications Manager instructs the two involved RSVP agents (one in
Location A. and one in Location B) to use RSVP to try to set up the call between each other. If
the call is admitted (that is, if enough bandwidth is available in the network path between these
two devices), the RSVP agents inform Cisco Unified Communications Manager that the RSVP
call leg was successfully set up.
Cisco UnifiedCommunications Managernow tells the phonesto set up their call legs,each to
its respective RSVP agent. If the RSVP call setup between the two RSVP agents is denied,
Cisco Unified Communications Manager considers the call to have failed CAC.
It is important to realize that there are three separate RTP streams: Phonel talks to RSVP
Agentl. RSVP Agentl talks to RSVP Agent2, and RSVP Agent2 talks to Phone2.
RSVP CAC is used between the RSVP agents only.
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Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.D
>2010 Cisco Systems, Inc.
RSVP-Enabled Locations
"Ihis topic describes RSVP-enabled locations in Cisco Unified Communications Manager.
RSVP-Enabled Locations
Characteristics
• Based on Cisco Unified Communications Manager locations
• Allows RSVP to be enabled selectively between pairs of locations
• Uses RSVP agents:
Devices (MTPs) through which call has to flow
'-•^b*
• RSVP used between RSVP agents
• Topology-aware:
- Works well with all topologies (full mesh, partial mesh, hub
and spoke)
- Adapts to network changes; considers actual topology:
• Link failures?
• Backup links?
• Load-share paths?
RSVP-enabled locations are based on Cisco Unified Communications Manager (standard)
locations. RSVP-cnahled locations differ from standard locations in two ways. First. RSVP can
be enabled selectively between pairs of locations. Because endpoints such as Cisco IP phones
do not supportRSVP. the solution uses so-called RSVP agents.
An RSVP agent is a Media Temiination Point (MTP) through which the call hasto flow. RSVP
is then used only betueen the two RSVP agents, while the Real-Time Transport Protocol (RTP)
stream from IP phone to RSVP agent does not use RSVP.
Thesecond andmostimportant difference between RSVP-enabled locations andstandard
locations is that the use of RSVP makes the location-based CAC mechanism topology-aware. It
workswell with all topologies (full mesh, partial mesh, and hub and spoke)and adapts to
network changes by considering the actual topology. Theadvantages include these
considerations:
•
Link failures: If one link in the IP network goes down and packets are routed on different
paths. RSVP is aware of the change and considers the bandwidth thatis now available at
the path that is actually routed.
•
Backup links: Ifbackup links are added after link failures, orif bandwidth ondemand is
used to add dial-on-demand circuits. RSVP again is fully awareof the routing path that is
currently used and thebandwidth that is available on each link along thatpath.
•
Load-share paths: If load sharing is used. RSVP is aware of the overall bandwidth thatis
provided by multiple load-sharing links.
Using RSVP for CAC allows admitting ordenying calls that are based onactual
oversubscriptions. The result is always based onthe currently available bandwidth and
interfaces, not on a logical configuration that ignores the physical topology.
© 2010 Cisco Systems. Inc
Bandwidth Management and CAC Implementation
Step 2: Assign Locations to Devices
The figure shows how you can assign locations to devices.
Step 2: Assic
Cisco Unified
Communications
Manager
Administration:
n Locat ons to Device s
Product Tr»»:
tltt*
OtvHe Proton 1
SCCP
Location is
indirectly applied
BBQi[tr*(*n
via device pool
(each device
pool Is
ur-tnown
Device > Phone
H"*"™'1
cm*
—=
T
location).
Itisr-f &L*ron Tarrtpl&U*
j If location is
1 selectedhere.
J
location of
device pool is
1
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N„„*m0w«om.u*.s
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•"
"""• -
Locations are a mandatory setting in a device pool, and you must assign a device pool to each
device. Therefore, a device always has a location that is assigned indirectly through its device
pool. If a device uses a different location from the one specified in its device pool, that location
can be chosen at the device itself. A location that is assigned at the device levelhas higher
priority than the location of the device pool.
3-50
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
12010 Cisco Systems, Inc.
Step 1: Configure Locations
The figure shows the configuration of the locations in Cisco Unified Communications
Manager.
Step 1: Configure Locations
Cisco Unified Communications Manager
Administration: System > Location
Enter location name.
Set bandwidth permitted for
cals coming into and going
out of location
You configure locations b\ choosing System> Location in Cisco Unified Communications
Manager Administration. One location exists bydefault: the HubNone location. This location
is the default location for all devices. To add a new location, click Add New.
In the Location Coniiguration window, entera name forthe location andset the bandwidth for
audio calls (the default is Unlimited).
Calculations of call bandwidth include Layer 3 overhead: A G.729 call is calculated with 24
kb/s. and a G.711 call consumes 80 kb/s. Be aware that unless RSVP-enabled locations (which
are discussedlater in this lesson)are used, the bandwidth limit that is configured at a location
applies only to calls coming into and going out ofthe location. Calls within a location are
ignored b\ standard location-based CAC.
© 201C Cisco Systems, Inc
BandwidthManagement and CAC Implementation
3-49
Locations Configuration Example: Hub-and-Spoke Topology
The figure shows an exampleof a hub-and-spoke locations-based CAC implementation.
Locations Configuration Example:
Hub-and-Spoke Topology
Location HQ 96 kb/s
Location BR1: 24 kb/s
Location BR2:72 kb/s
There are three sites: the headquarters and two branches. Each site has its own location (IIQ,
BR I. and BR2). The physical topology is a hub-and-spoke topology (headquarters is the hub).
The link between branch I and the headquarters should not carry more than one G.729 call, and
the link between branch 2 and the headquarters should not carry more than three G.729 calls.
The next two subtopics describe how to implement locations-based CAC for this scenario.
3-48
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8,D
>2010Cisco Systems, Inc.
Configuration Procedure for Implementing Locations-Based
CAC
To implement Cisco Unified Communications Manager locations-based CAC. you need to
follow two steps.
Configuration Procedure for
Implementing Locations-Based CAC
Add locations and configure CAC bandwidth limit.
2
Note
Assign locations to devices.
To know how much bandwidth you need to calculate per call, you should design and
configure regions before implementing locations
© 2010 Cisco Systems, inc
Bandwidth Management and CAC Implementation
3-47
Locations: Full-Mesh Topology
The figure shows a full mesh topology with locations-based CAC.
Locations: Full-Mesh Topology
What if there are no
Bandwidth is not
limited between pairs
of locations, which can
Location HQ:
Urt mited
calls to or from BR1
are attempted, but four
causeprobtems in
calls are attempted
non-hub-and-spoke
from 8R2 toHQ?
topologies.
Location BRl: 48 kb/s
Location BR2:96kb/s
This example is based on the previous example, but a direct IP WAN link has been added
between BRl and BR2. The idea is that one G.729 call is allowed on the WAN link from BRl
toward the headquarters, one G.729 call is allowed on the WAN link between BRl and BR2,
and three G.729 calls are allowed on the WAN link from BR2 toward the headquarters.
Such a scenario reveals issues that arise when locations-based CAC is used in topologies other
than hub-and-spoke topologies. To allow the additional G.729 call that is permitted on the
WAN link between BRl and BR2, the bandwidth limit of these two locations has been
increased by 24 kb/s. Increasing the bandwidth, however, can lead to these undesirable
situations:
•
Two G.729 calls from BRl to HQ: Because the BRl location now has a limit of 48 kb/s.
it allows two G.729 calls. Location bandwidth limits are not configured per destination; any
call coming into or going out of a location is considered, regardless of the other location
that is involved in the call. Therefore, there is no way to divide the available 48 kb/s into
one call toward the HQ and one call to BR2.
•
Kour G.729 calls from BR2 to HQ: The same problem occurs with the BR2 location: The
additional bandwidth that was added to accommodate the desired call toward BRl can be
used toward I IQ. occupying that link with one more call than intended.
Note
3-46
The problems that are described here are caused by the fact that the bandwidth limit is
configured per location, regardless of the other location (where the call goes or comes from)
Implementing Cisco Unitied Communications Manager, Part 2 (CIPT2) v8.0
12010 Cisco Systems, Inc.
Locations: Hub-and-Spoke Topology
"fhe figure shows a hub-and-spoke Cisco Unified Communications Manager topology with
locations-based CAC.
Locations: Hub-and-Spoke Topolo*
Locations-based CAC works relatively well in
hub-and-spoke topologies.
i tjkt™
idlBaB
lxG.729
I
Location HQ.
Unlimited
\
3xG729
IPWANV
Location BR2: 72 kb/s
Location BRl: 24 kb/s
As shown in the figure, there are three sites: the headquarters (HQ) and two branches (BRl and
BR2). There is no direct connection between the branches; all traffic goes by way of the
headquarters.
Ihis scenario is ideal for locations-based CAC If the intention is lo allow only one G.729 call
on the link between BRl and HQ and three G.729 calls on the link between BR2 and HQ. this
location configuration would suit these needs:
•
Location HQ: Unlimited
•
Location BRl: 24 kb/s
•
location BR2: 72 kb/s
This coniiguration ensures that no more than oneG.729 call will be sentoverthe IP WAN
toward location BRl and that no more than three G.729 calls will be sent over the IP WAN
toward location BR2.
Note
The configuration also allows one G.729 call betweenBR1 and BR2. Because the
configured bandwidth limit does notconsider the destination location, the 24-kb/s limit of
BR1 allows any call to go out (orcome in) regardlessof where itgoes (orwhere itcomes
from) The headquarters limit is not affected at all bysucha call. Only locations BR1 and
BR2 will subtract 24 kb/s from their limits. Because locations-based CAC does not provide
topology awareness, Cisco Unified Communications Manager is not even aware that thecall
physically flows through the headquarters.
©2010 Cisco Systems, Inc
BandwidthManagement and CAC Implementation
^^
3-45
Standard Locations
This topic describes how to implement CAC in Cisco Unified Communications Manager by
using standard locations.
Locations Characteristics
Each device has one location assigned.
You limit calls by permitting a certain bandwidth for calls coming in
and going out of a location:
- Audio bandwidth is calculated by actual codec plus IP
overhead (assuming 20 ms packetization period).
• Examples: 80 kb/s for G.711, 24 kb/s for G.729.
- Calls within a location are unlimited.
- Bandwidth limit of source and of destination location are
checked individually.
Works within a Cisco Unified Communications Manager cluster
(including exit points):
- Trunks and gateways can be put into a location, allowing some
control for calls leaving the cluster.
Locations-based CAC is unaware of topology
Each device has one location assigned. The assignment can be direct or via a device pool. If
both types of assignment are used, the device configuration has higher priority.
You limit calls by permitting a certain bandwidth for all calls coming into and going out of a
location. Cisco Unified Communications Manager calculates the actual audio codec bandwidth
plus IP overhead (assuming a packetization period of 20 ms). This means that each G.711 call
reduces the bandwidth that is configured for a location by 80 kb/s, while a G.729 call reduces
the available bandwidth by 24 kb/s.
Note
Calls withina location do not decrease the bandwidth limit; they are unlimited. Only calls that
go out of a location or that are received from outside the location are considered by the
locations-based CAC algorithm.
The bandwidth limitsthat are configured at the location of the originating device (the source
location) as well as at the location of the terminating device (the destination location) are
checked individually. Unlike with regionconfiguration, where the maximum permittedcodec is
configured perpairofregions, the bandwidth limitof a location applies to all (both placed and
received) interlocation calls. If the bandwidth limit of the source or of the destination location
(or of both) is exceeded, the call is not admitted. Locations provideCAC for calls within
clusters; however, because locations can alsobe configured for gateways and trunks, locations
do allow some control for calls leaving the cluster.
Locations-based CAC in Cisco Unified Communications Manager is completely unaware of the
topology of the network. It is a purely logical assignment and does not reflect the actual
topology or the actual bandwidth available.
3-44
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
) 2010 Cisco Systems, Inc.
CAC in Cisco Unified Communications Manager
Cisco Unified Communications Manager supports various CAC methods.
CAC in Cisco Unified Communications
Manager
Cisco Unified Communications
Manager supports the following
H.323 Gatekeeper or SIP
CAC features:
«_CAC-K
• CAC within a cluster:
RSVP-enabled locations
If CAC denies call, automated
alternate routing (AAR) can be
used lo reroute call over PSTN
CAC for calls leaving a cluster.
H.323 gatekeeper or SIP
Locations
.
CAC
i Locations
I
CAC „,
Preconditions
If CAC denies call and no other
entnes in route list are left to
try. call fails.
In centralized call-processing deployments, you can use standard locations and Resource
Reservation Protocol (RSVP)-enabled locations lo provide CAC within a Cisco Unified
Communications Manager cluster. If a call is not admitted by one of these two CAC methods
due to bandwidth limitations, you can use AARto reroutethe call over the PSTN (off-net)
instead of denying the call. AAR provides a service like PSTN backup, except thatthe reason
for call backup is not that the call failedon the on-ncl path, but that there is no available
bandwidth from a CAC point of view.
In distributed call-processing environments, you canuse H.323 gatekeeper CAC with H.323
trunks (gatekeeper-controlled intercluster trunks and H.225 trunks). If Session Initiation
Protocol (SIP)trunks, you can use SIP Preconditions, which allows RSVP-based CAC.
If calls are not admitted by the H.323 gatekeeper, standardbackup functionalitv of route lists
and route groups is applied, for example, to route calls thathave not been admitted by the
gatekeeper to be sent over the trunk, you can configure one or more PSI'N gateways inanother
(lower-priority) route group of the same route list. In this way, the gatekeeper-controlled trunk
ispreferred over the PSTN aslong ascalls are admitted; after admission isrejected, calls are
sentoverthe PSTN. Thesame principle applies to callsthat arc placed through SIP trunks that
are configured for SIP Preconditions.
) 2010 Cisco Systems. Inc
Bandwidth Management and CAC Implementation
CAC Overview
This topic describes the CAC optionsthat are available in Cisco Unified Communications
Manager.
Call Admission Control
CAC limits the number of calls between certain parts of
the network in order to avoid bandwidth oversubscription:
• QoS can be used to give priority to Voice over Data.
• QoS cannot solve the problem of too much prioritized traffic
(caused by too many voice calls).
• Oversubscription results in delayed packets and packet drops:
- Any packets of any voice stream are affected (not just packets
of the call that exceeds bandwidth limit).
- Results in quality degradation of all voice calls.
• CAC avoids such problems by limiting the number of voice calls.
CAC limits the number of calls between certain parts of the network in order to avoid
bandwidth oversubscription with too many voice calls. QoS is not able to achieve this result
because QoS provides only the means to prioritize Voice over Data traffic. QoS does not avoid
the situation in which too many (prioritized) voice streams are sent over the network.
If oversubscription occurs, any packets of any voice stream can be affected, not just packets of
the particular call or calls that exceed the bandwidth limit. The result in this case is packet
delays and packet drops of all voice calls, and hence oversubscription degrades the quality of
all voice calls.
Therefore, in order to ensure good voice quality, you need to use CAC to limit the number of
voice calls.
3-42
ImplementingCisco Unified Communications Manager, Part 2 (CIPT2)v8.0
>2010 Cisco Systems. Inc.
Lesson 2
Implementing CAC
Overview
Implementing multisite IP telephony deployments over an IP WAN requires additional
planning to ensure the quality and availability of voice calls.
When an IP WAN connects multiple sites in a Cisco Unified Communications deployment.
quality of sen. ice (QoS) hasto be implemented in orderto prioritise voice packets overdata
packets. Ilow ever, to avoid an oversubscription that is caused by too many voice calls, a
mechanism is necessary to limit the numberof calls that are allowed at the sametime between
certain locations. Call Admission Control (CAC) is the mechanism that ensures that voice calls
do not oversubscribe the IP WAN bandwidth and thus impact voice quality.
This lesson describes how to implement CAC mechanisms that are provided by Cisco Unified
Communications Manager, andexplains how automated alternate routing (AAR) can be used in
some scenarios lo reroute calls that were denied by CAC over the public switched telephone
network (PSTN).
Objectives
Upon completing this lesson, you will be able to describe and configure CAC mechanisms and
AAR in Cisco Unified Communications Manager and in gatekeepers. Ihis ability includes
being able to meet these objectives:
,^M
u Describe the CAC options that are provided by Cisco Unified Communications Manager
•
Implement locations-based CAC in Cisco Unified Communications Manager
•
Implement RSVP-enabled locations-based CAC in Cisco Unified Communications
*•*
Manager
•
Implement AAR inorder to reroute intracluster calls over the PSI'N if notenough
bandwidth is available for an on-net call
•
•
Implement SIP Preconditions onSIP trunks inCisco Unified Communications Manager
Implement 11.323 gatckceper-based CAC in Cisco Unified Communications Manager
3-40
Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
Bandwidth management methods include techniques that reduce
required bandwidth of voice streams, techniques that keep voice
streams off the IP WAN, and other techniques such as deploying
transcoders.
The highest bandwidth-consuming codec for a call between
two devices is determined by the Cisco Unified Communications
Manager region configuration of the two involved devices.
When you are deploying local conference bridges at multiple
sites, use Media Resource Groups and Media Resource Group
Lists to control which conference bridge is used by which device.
Transcoders allow low-bandwidth codecs to be used over the
IP WAN when low-bandwidth codecs are not supported by both
endpoints.
Multicast MOH from branch router flash is a feature that allows
MOH streams to be generated locallyat the remote site instead
of being sent across the IP WAN from the main site.
References
for additional infonnation. refer to these resources:
•
Cisco Svstems. Inc. Cisco Unified Communications System 8.x SRND. April 2010.
litip:.,',www.ci,bCo.com/en.llS'/docs/\oicc ip eomm/cucm/srnd/8\/ue8\.htm)
•
Cisco S\ stems. Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0(1). February' 2010.
iilip:';www.cisc(XC<)m;cn;US^docs^(iiec_ip_coniin/cucni/admin/8_0_l/ecnicfg''1iccm-801cm.him]
•
Cisco Systems. Inc. Cisco Unified SRST System Administrator Guide. December 2007.
hup:.' www,cisco.eoni/en'US/pji'lner'docs/voice ip conim/cusrst/admiii/srst/coiifiguratioii.'
sjuide.-'srstsa.html
>2O10Cisco Systems, Inc
BandwidthManagement and CAC Implementation
3-39
Step 4c: Disable Multicast Routing on IP WAN Router Interface
The figure shows how to configure a Cisco IOS router to disable multicast routing at the IP
WAN interface.
Step 4c: Disable Multicast Routing on
IP WAN Router Interface
ip multicast-routing
interface FastEthernetO/0
description HO-Voice-Servers
ip address 10.1.1.101 355.255.255.0
ip pim sparse-dense-mode
interface FastEthsrnatO/0
description HQ-Phones
ip address 10.1.2.101 255.255.255.0
ip pim sparse-dense-mode
interface SerialO/l
description ip wan
1Disable multicast
ip address 10.1.4.101 255.255.255.0
" .
,
,
,
no ip pim sparse-dense-mode
.^___
..
,.,».,
_J routingD on WAN
-! interface.
If no other multicast applications are used over the IP WAN, the simplest way of preventing the
multicast MOH packets from being sent to the WAN is to disable multicast routing at the WAN
interface.
3-38
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
) 2010 Cisco Systems, Inc.
Step 4b: Use IP ACL at IP WAN Router Interface
The figure shows how you can use an IP ACL to configure a Cisco IOS router to drop multicast
MOI I packets.
Step 4b: Use IP ACL at IP WAN Router
Interface
ip multicast-routing
ip access-list extended drop-moh
deny ip any host 239.1.1.1 range 16384 163B5
permit ip any any
interface PastEtoernetO/0
^^"-
description HQ-Volce-Servers
ip address 10,1.1.101 255.255.255.0
IPACL Filtering MOH
Multicast Packets
ip pim sparse-dense-mode
interface FastEthernetO/1
description HO-Phones
ip address 10,1.2.101 255.255.255.0
ip pim sparse-dense-mode
interface SeriolO/1
description IP MAN
ip address 10.1.4.101 255.255.255.0
ip access-group drop-moti out
ip pim sparse-dense-mode
^___
["
1
i
PACLAppliedtoWAN
nterface
The ACL matches the MOH group address and port numbers that are used by the MOII server
for the MOH RTP and RTCT packets. The ACL is applied to the IP WAN interface in the
outgoing direction and therefore doesnot allow multicast MOH packets to be sentout on the IP
WAN.
Note
As stated earlier in this lesson, the multicast address and port range that must be filtered
depend on several parameters, such as the audiosource number, the enabledcodecs, and
the increment method.
)2O10 Cisco Systems, Inc
Bandwidth Management and CAC Implementation
Step 4a: Configure the Maximum Hops to Be Used for MOH
RTP Packets
Continue the multicast MOH server configuration by setting the maximum hop value, as shown
in the figure.
Step 4a: Configure the Maximum Hops
to Be Used for MOH RTP Packets
Cisco Unified Communications Manager Administration:
Media Resources > Music on Hold Server
* tr.*hl* >**«UI* L.M &*urcol 6" P%ri
NOtiSVW
(Utdroi*
&*u»U&t4U»Brt
16SW
*""*"
numbn null)
ta™.»tau ^ hnHfan*,. '
if Mai it
Selected *utoc««r Avdvsaortct-
Set maximum hops (time to
live) value for each audio
source for multicast packets.
All MOH audio sources that have been configured for multicasting are listed in the Selected
Multicast Audio Sources section of the MOH Server Configuration screen. You can set the
Max Hops value for each audio source; the default is 2. This parameter sets the TTL value in
the IP header of themulticast MOH RTP packets to the specified value. TTLin an IP packet
indicates the maximum numberof routersthat an audio source is allowedto cross. If Max Hops
is set to 1. the multicast MOH RTP packets remain in the subnet of the multicast MOH server.
Whenyou use multicast MOHfrom branch router flash, you can set Max Hopsto a value that
is lower than the actual hop count from the MOI 1server toward the WAN interface of the main
site router. This value, however, might conflict with the needs within the main site when IP
phonenetworkshave the same or a higher distance—that is, a higherhop count—to the MOH
serverthan the WAN network. In such a case, one of the other possible methodsof preventing
the multicast MOH packets that aregenerated by the MOH server have to be used. Theyare
shown on the following pages.
3-36
Implementing CiscoUnified Communications Manager, Part 2 (C1PT2) v8.0
>2010 Cisco Systems, Inc.
Step 3: Enable Multicast MOH from Branch Router Flash at the
Branch Router
The figure shows how to configure a Cisco IOS routerfor multicast MOI I from branch roi ter
flash.
Step 3: Enable Multicast MOH from Branch
Router Flash at the Branch Router
call-manager-fallback
mai-epbones 1
nax-dn
1
ip source-address 10.1.5.102
nob mob-file.au
multicast mob 239.1.1.1 port 16364
interface FastEtharnatO/0
description BF-Phones
ip address 10.1.5.102 255.255.255.0
interface SerlalO/1
description IP WXN
ip address 10.1.4.102 255.255.255.0
Multicast MOH from branch router flash is part of the SRST feature. Therefore, SRST must
already beconfigured before youcan enable multicast MOII from branch router flash.
Note
SRSTconfiguration options are discussed inthe module "Centralized Call -Processing
Redundancy Implementation."
Rased on anexisting SRST configuration, you need only two commands to enable multicast
MOH from branch router Hash:
•
moh file-name: fhis command specifies the MOH audio source file. The specified file has
to be stored in flash memor> of the SRST gateway.
•
multicast moh multicast-group-address portport: This command specifies themulticast
address and port that are used for the multicast MOH packets. The specified address and
port have toexactly match the values that have been configured at the MOH server in
Step 2b.
Note
TheSRST gateway will permanently streamMOH, regardless ofan IPWAN failure or IP
phones being registered with the SRSTgateway.
^^
You can configure an additional five MOH streams using MOH group configuration. Refer to
the module "Centralized Call-Processing Redundancy Implementation" for more information
about MOII group configuration.
i 2010 Cisco Systems. Inc.
BandwidthManagement and CAC Implementation
3-36
Step 2c: Enabling Multicast MOH at the Media Resource
Groups
This figure shows that the configuration of a Multicast is enabled at the Media Resource Group
(MRG).
Step 2c; Enabling Multicast MOH at the
Media Resource Group(s)
Cisco Unified Communications Manager Administration: Media
Resources > Media Resource Group
Multicast MOH only works if the multicast enabled MOH server is assigned to a multicast
enabled MRG. This MRG will be configured to be a member of an MRGL. The MRGL will
then be associated with devices such as phones.
%m*
3-34
ImplemenlingCisco Unified Communications Manager, Part 2 (CIPT2)v8.0
>2010 Cisco Systems, Inc.
«*»
Step 2b: Configure Multicast MOH in Cisco Unified
Communications Manager
After allowing multicast MOH on audio sources, you must enable the MOH server for
multicast MOH. as shown in the figure.
Step 2b: Configure Multicast
Cisco Unified Communications Manager
Cisco Unified Communications Manager Administration:
Media Resources > Music on Hold Server
r^ttmtlkijvltvtr InlwnilMn-
•iaMtMHfcWiiliwi.in rmm itniif
D
J
Enable multicast MOH on the
MOH server.
'.ferine**'
«Lttte4 HuKKuf AvrfiD 4
Configure multicast
parameters.
fhe figure shows how to enable multicastMOH on a MOH server: In the Multicast Audio
Source Infonnation section of the MOH server configuration screen, check the finable
Multicast Audio Sources on this MOH Server check box. 'fhe Base Multicast IP Address, Jase
Multicast Port Number, and Increment Multicast On parameters are automatically populated
when you enable multicast MOH on the server. You can modify these values as desired.
Note
To avoid network saturation in firewall situations, it is recommended that you choose to
increment multicast MOH on the IP address instead of on the port number. Choosing this
option means that each multicastaudio source will have a unique IP address, and helps to
avoid network saturation If multiplecodecs are enabled for the MOH server, additional IP
addresses will be in use (one per codec and per audio source).
>2010 Cisco Systems, Inc.
Bandwidth Management and CAC Implementation
3-33
Step 2a: Configure MOH Audio Sources for Multicast MOH
To enable multicast MOH. you first have to allow multicast MOH on MOH audio sources, as
shown in the figure.
Step 2a: Configure MOH Audio Sources
for Multicast MOH
Cisco Unified Communications
Cisco Unified Communications
ManagerAdministration: Media
ManagerAdministration: Media
Resources > Music On Hold
Resources > Fixed MOH Audio
Audio Source
Source
-Nuikh W d l r m r h A t b V f l l**W1**Oan-
-*OH *j*e 5ff»«*n*fc*r**r' E
v
O-.-X-*
$«.,..*
Flu * HOH *«»• I
l>f>
>b
Mm'
|l 'u»Mc«»i| |
("0-T*.(
HemMi (M^jksJ.Ni™ ntouhrd j
Tr#ftjl»4tfift Coft*l*«
DilfcSjKI-6
Btf«
LO-Ctf •T.-i*
11 )I»H J 1 i
\
MOH audio sources do not allow multicast
MOH by default.
tan pi* Audi iSotii ca Lul*- t*a y
irra^'udi^i^t* (?28.i««v
MOH audio sources and fixed MOH audio
"
sources (if used) must be enabled for
multicast MOH.
Check the Allow Multicasting check box for each MOH audio that is allowed to be sent as a
multicast stream. This instruction applies to MOH audio sources and to fixed MOH audio
sources.
Note
3-32
More information about configuring the MOH server and MOH audio sources is provided in
the ImplementingCisco Unified Communications Manager, Part 1 (CIPT1) course.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) vB.O
© 2010 Cisco Systems, Inc.
Step 1: Enable Multicast Routing on Cisco IOS Routers
The figure shows how to enable multicast routing on Cisco IOS routers at the main site an<: at
the remote site.
Step 1: Enable Multicast Routing in
Cisco IOS Routers
ip multicast-routing ~
—
Enable multicast
routing on router.
interface FaatEthernetQ/O
description HQ-Voice-Servers
ip address 10.1.1.101 255.255.255.0
ip pin sparse-dense-mode
interface FaatEtnernetO/1
description HQ-Phonea
ip address 10.1.3.101 255.255.255.0
ip pim sparse-dense-mode
Enable multicast
routing on interfaces.
interface SerialO/l
description IP WAN
ip address 10.1.5.101 255.255.255.0
ip pim sparse-dense-mode
You use two commands to enable multicast routing in the network so that multicast MOH
streams can he sent:
•
anip multicast-routing: Configure thiscommand in global configuration mode. This
command enables multicast routing on the Cisco IOS router in general.
•
ip pim sparse-dense-mode: Configure thiscommand on eachinterface where multicast
routing should he enabled.
Note
The configuration that is shown in the example enables multicast routing in the whole
network. When multicast MOH from branch router flash is used, multicast streams will not be
sent to the IP WAN They can be blocked based on the maximum hops parameter (TTL field
in the IP header) or by IP ACLs. You can also block multicast streams by disabling multicast
routing on the interface, butonlyifno other multicast routing applications are required inthe
network.
In the example, the maximum hops parametercannot be used because the HQ-Phones
network and the IP WAN network have the same distance to the HQ-Voice-Servers network
To allow multicast MOH to be sent to the HQ phones, a maximum hop value of 2 is required.
This value, however, will allowthe multicast MOH packets to be sent out on the WAN
interface. Therefore, IP ACLs have to be used, or multicast routing has to be disabled at the
WAN interface (if multicast routing is not required by otherapplications).
© 2010 Cisco Systems, Inc
BandwidthManagement and CAC Implementation
3-31
Configuration Procedure for Implementing Multicast MOH from
Branch Router Flash
Implementing multicast MOH from branch router Hash includes the following steps.
Configuration Procedure for implementing
Multicast MOH from Branch Router Flash
1
2
Enable multicast routing in the network.
Configure multicast MOH in Cisco Unified Communications
Manager:
a.
b
c
3.
Configure MOH audio sources for multicast MOH.
Configure MOH audio server for multicast MOH.
Enable multicast MOH at the media resource group(s)
Enable multicast MOH from branch router flash at the
branch router.
4
Implement a method to prevent multicast MOH streams
from being sent over the IP WAN:
a
Configure maximum hop value to prevent multicast
MOH streams from being sent over the IP WAN.
b
Use IP ACL at IP WAN router interface.
c.
Disable multicast routing on IP WAN router interface.
The configuration procedure describes the implementation of multicast MOH from branch
router flash by first enabling multicast MOH (steps 1 and 2). Once this works as desired, the
configuration is modified so that the multicastMOH streamis generated locallyat the branch
router (Step 3) and the multicast MOH stream that is generated by the MOH server is prevented
from being sent to the IP WAN.
WhenenablingmulticastMOH at the MOHserver,make sure that you set the maximum hop
value of the multicast-enabled MOH audiosource(s)to a high enough value to allow the
multicast MOH packets to be sent all the way to the remote phones.
When choosing option 4a to preventing the multicast MOH stream of the MOH server from
being sent to the IP WAN.you have to use a low enough value to ensure that the multicast
MOH packets generated by the MOH server do not reach the IP WAN.
Note
All IP phones must be able to access to the main site Cisco Unified Communications
Manager MOH server from their MRGL. This access is required as soon as multicast MOH
is configured, whether multicast MOH from branch router flash is used. If the remote site IP
phones do not have access to the Cisco Unified Communications Manager MOH server
from theirMRGL, CiscoUnified Communications Manager cannot instruct the IP phones to
join the multicast group and will make the phone use tone on hold instead of MOH.
Furthermore, you need to check the Use Multicast for MOH Audio check box at the MRG
that includes the multicast-enabled MOH server (see Step 2c).
Finally, make sure that the G.711 codec is used between the MOH server and the branch
phones, because SRST multicast MOH supports only G.711.
3-30
Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
•
Disable multicast routine at the IP WAN interface: By disabling multicast routing at the
IP WAN interface, multicast packets are not routed out on that interface.
At the branch router, the multicast MOH stream is sent out on the interface that is specified in
the ip source address command in call-manager-fallback coniiguration mode(or in telephonyserver configuration mode, when Cisco Unified Communications Manager Express in SRST
mode is used). Therefore, the multicast MOII stream that is generated at the branch router does
not have to be blocked at the branch router WAN interface.
©2010 Cisco Systems, Inc
Bandwidth Management and CAC Implementation
3-29
Example: Implementing Multicast (VIOH
from Branch Router Flash (Cont.)
Cisco Unified
Communications
Manager MOH
Configuration
DA 239 1 1 1
DP. 16384
(a) Max Hops
(TTLV1
Main Site
ill-manager-fallback
i-epbones 1
(b)
ip
accaaa
-liat
deny udp any ho
16384
sirtended
drop
3t
1.1
239.1
mob
i ange
majt-dn
1
ip aouroa-address 10.1.5.102
16385
moh moh-fila.an
per•Bit
ip
any
any
multicast moh 239.1.1.1
inte rface
ip
access
serial
0/0
-group
drop-moh out
port 16394
The figure shows the configuration of the previously discussed scenario.
It is assumedthat the baseline configuration provides multicast routing in the wholenetwork
and that the Cisco Unified Communications ManagerMOHserver is alreadyconfigured for
multicast MOH.
Now the multicast MOI1stream that is sent toward the remote site needs to be blocked, and
multicast MOH from branch router flash needs to be implemented at the remote site.
Therefore, the SRST configuration of the remote site router is extended to include multicast
MOH. The SRST configuration uses the same multicast IP address and portthat are configured
at the Cisco Unified Communications Manager MOH server that is located at the main site.
To stopmulticast MOH generated by themain siteCisco Unified Communications Manager
MOH server from beingsentoverthe IP WAN, you canchoose oneof three options:
•
Set Time to Live (TTL) to a low enough value at the Cisco Unified Communications
Manager MOH server: If the TTL value in the IP headerof the generated multicastMOH
packets is set to a low enough value, the packets will not be routed out to the IP WAN.
However, if the IP WAN link is one hop away from the Cisco Unified Communications
Manager MOII server, andif the main site phones are alsoone hop away from the server.
this method cannot beused, because themain site IPphones would also be affected by the
dropped packets. In the current example, TTL is set to 1, and it is assumed that the IP
phones are inthe same VLAN, like the Cisco Unified Communications Manager MOI I
server.
•
Filter the packets by an IP access control list (ACL): At the main site router, an ACL
can be configured that drops the multicast MOH packets at the IP WAN interface.
Note
Make sure thatyou verify the actually used multicast IPaddresses and ports. Asdescribed
earlier, itdepends on the base address and port configuration, the method that is used to
increment the base number {on IP address or port), the codecs that are enabledfor MOH,
and the audio sources that are multicast-enabled.
3-28
Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Example: Implementing Multicast MOH from Branch Router
Flash
The figure shows a sample scenario for implementing multicast MOH from branch router flash.
Example: Implementing Multicast MOH
from Branch Router Flash
Multicast MOH packets generated by the Cisco Unified
Communications Manager MOH server should not be sent over the
IP WAN. Branch router should generate multicast MOH packets.
Cisco Unified
Communicalions
Manager MOH
Wain Site
WAN
SRST MOH
Remote Site
In the example, a MOH server is located at themain site. It is configured for multicast MOH.
Multicast routing has been enabled in the whole network—including inthe IP WAN link to the
remote site.
The main siterouter, however, should no longer route multicast MOH to the remote site. The
remote site SRSI gateway should instead generate multicast MOII streams to the phones that
are located at the remote site.
Cisco Unified Communications Manager is not aware that the multicast packets that are
generated by the MOH server atthe main site are filtered onthe IP WAN interface and then are
locally generated by the remote site SRST gateway. Therefore, Cisco Unified Communications
Manager will instruct the IP phones that are located at the remote site lojoin the multicast
group IP address that isconfigured at the Cisco Unified Communications Manager MOH
server. To allow the phones to receive MOH for the multicast group IP address that they join,
you must configure the SRST gateway touse the same multicast address and port that isused
bv the Cisco Unified Communications Manager MOH server that is located at the main site.
© 2010 Cisco Systems. Inc.
Bandwidth Management and CAC Implementation
3-27
As you can see from the table, audio sources are incremented in ascending order, starting with
audio source 1 (live audio at source 0 is not multicast-capable and hence is excluded in the
calculation). Codecs are enumerated in the order shown (G.711 mu-law, G.711 a-law, G.729,
wideband). For each audio stream, two ports are used: the first one (the even-numbered port)
for the actual RTP transmission and the subsequent one (the odd-numbered port) for the
corresponding RTCP.
If you are not sure about the used multicast addresses and ports, you can configure traces for
the Cisco IP Voice Media Streaming Application service. Make sure that you check the Service
Initialization check box in the trace configuration. Then restart the Cisco IP Voice Media
Streaming Application service.
When analyzing the trace output, you will find this kind of information:
CMOHHgr: :KickStartMultiCastStreaiii {1} Starting Multicast
stream, asID = 1, conferencelD = 1, cocleeType = iKtSw,
Multicast ip:port - 239.1.1.it 163841
<CLID::Cluster><NID::10.1,i,l>
CMOHMgr::KickStartMultiCastStream <1) Starting Multicast
stream, asID = 1, conferencelD = 1001, GQdecType = «!#"$,
Multicast ip:port - 239.1,1.2il6384|
<CLID: :ClusterxHID: :1Q.1.1.1>
Note
The output that is shown does not match the example in the figure. It is used only to
illustrate which information you will find in the trace output. The number after the
KickStartMultiCastStream identifies the audio source. For each enabled codec, you will find
information about the used multicast IP address and port. The NID (node ID) shows the IP
address of the MOH server. In this example, only G.711 mu-law and G.711 a-law codecs
are enabled. Onlyone audio source (audio source 1) is multicast-enabled. There is a single
MOH server at 10.1.1 1
Tip
It is importantto know the used multicastIP addresses and ports when you choose the
option to prevent multicast traffic from entering the IP WAN by access lists.
3-26
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems, Inc.
Multicast MOH: Address and Port Increment Example
The table shows an example of IP address and port increments for multicast MOH.
Multicast MOH: Address and Port
Increment Example
You can verify used IP addresses and ports by using traces:
- Configure trace for the Cisco IP Voice Media Streaming
Applicationservice, and check the Service initialization
check box
Restart Cisco IP Voice Media Streaming Application service.
- Analyze trace output.
increment on IP Address
Base IP 239.1 1.1
Increment on Port
Base Port 16384
Enabled Codecs
G711a-law, G729
239.1.1.2
Audio Source 1
Audio Source 2
No
Audio Source 3
"Ybs"
Audio Source 4
No
Audio Source 5
No
Audio Source 6
Yes
239.1.1.23
239113
16386
239 1*1 11
1S402
239 1 1 23
16404
16428
The base multicast group iseonligured for IPaddress 239.1.1.1 and port 16384. G.711 a-law
and G.729 codecs are enabled: audio sources 1. 3. and 6 are multicast-enabled.
The figure slums the IP addresses orports that are used for the actual multicast MOII streams.
The next table shows how these numbers were derived.
Note
The yellow highlighted numbers mthetable present thevalues thatare used in thefigure.
The gray highlighted and bold numbers present theIP address and port number that are
actually used in this example.
Audio
Increment on IP Address
Increment on Ports
Source
{239.1.1.x—Only Last Octet Shown)
(163xxx—Only Last Octet Shown)
G.729
Wideband
386/387
388/389
390/391
392/393
394/395
396/397
398/399
12
400/401
402/403
404/405
406/407
15
16
408/409
410/411
412/413
414/415
18
19
20
416/417
418/419
420/421
422/423
22
23
24
424/425
426/427
428/429
430/431
G.711
G.711
mu-law
a-law
4
364/3B5
7
8
10
11
13
14
5
17
6
21
G.729
G.711
G.711
mu-law
a-law
1
1
2
3
2
5
6
3
9
4
© 2010 Cisco Systems. Inc
Wideband
Bandwidth Management and CAC Implementation
3-25
Multicast MOH from Branch Router Flash: Address and Port
Considerations
This subtopic describes what you need to consider regarding the multicast group coniiguration
of MOH streams.
Multicast MOH from Branch Router
Flash; Address and Port Considerations
You can configure MOH streams to be incremented on IP
addresses or ports.
Increment on IP address is recommended.
For each audio source, four streams are considered (one per
codec: G.711 mu-law, G.711 a-law, G.729, and wideband) for the
increment.
- independent of codecs that are actually enabled.
Numbers are incremented differentlyon IP address versus port.
- IP addresses are incremented by one per stream.
- Ports are incremented by two per stream (RTP and RTCP).
When you increment IP address or port, consider audio sources
that are not enabled for multicast.
- Appropriate range is reserved for future multicast enablement.
Because a single MOH server can stream multiple multicast MOH files, you have to specify an
initial multicast address and portthat is used forthe firststream. In addition, you have to
choose whether to increment the IP address or port on additional streams. It is recommended
thatyou increment on IP addresses instead of on ports. If there aremultiple MOH servers
within a network, you have to make sure that they do not use overlapping multicastIP
addresses and ports for their streams.
Foreach audio source, fourstreams are considered for the increment—one per codec: G.711
mu-law, G.711 a-law, G.729, and wideband. Thisprinciple always applies, regardless of which
MOH codecs have been enabled in the CiscoIP Voice Media Streaming Application service.
When you are incrementing on IP addresses, each stream consumes one IP address. In other
words, eachaudio source requires fourIP addresses. When incrementing on ports, you have to
considerthe Real-Time Transport Control Protocol (RTCP). For each audio stream,two
separate RTP ports are reserved: one forthe actual audiotransmission andone for (the
optional) RTCP. Therefore, when you areincrementing multicast MOH on ports, each stream
consumes two ports. You have to calculate eight port numbers peraudio source (two ports per
codec).
Audio sources that are not enabled for multicast MOH should nevertheless be considered for
the increment ofaddresses orports. Audio source 1,which starts with the configured base
address and port, requires four IPaddresses oreight ports. The same principle applies toeach
consecutive audio source (audio source 2, audio source 3, and so on), regardless of whether
these audio sources are multicast-enabled.
3-24
Implementing CiscoUnified Communications Manager, Part2 (CIPT2) v8.0
© 2010 Cisco Systems, Inc.
Multicast MOH from Branch Router Flash: Region
Considerations
Ihis subtopic discusses how to configure codecs when you are using multicast MOH from
branch router flash.
Multicast MOH from Branch Router
Flash: Region Considerations
* Locally generated MOH streams have to be identical to the
streams that are generated by Cisco Unified Communications
Manager MOH servers:
Same destination IP address (multicast address).
- Same destination port number.
- Same codec.
- Same packetization period.
* Multicast MOH from branch router flash supports only G.711.
• G.711 must also be used for the stream generated by Cisco
Unified Communications Manager MOH servers:
Put Cisco Unified Communications Manager MOH server
into a dedicated region.
Allow G.711 between the region of the MOH server and
the region used by branch phones.
- Limit all other intersite calls to G.729.
When multicast MOH is used. IP phones andCisco Unified Communications Manager are not
aware that the IP phones listen to locally generated MOH streams. From a signaling
perspective, the IP phone isinstructed to listen to acertain multicast stream, and the local
SRSI gateway has to generate amulticast MOH stream by using identical settings, such as
destination address (multicast group), destination port, codec, and packetization period.
Multicast MOH in SRST gateways and Cisco Unified Communications Manager support only
the G.711 codec, 'fherefore. G.711 must also be configured between the Cisco Unified
Communications Manager MOH server and the branch IPphones. IfCisco Unified
Communications Manager signals a codec other than G.711 tothe IP phone, the IP phone could
not play the locally generated MOH stream because ofa codec mismatch (the signaling would
be G.729. but the received RTP stream would be G.711).
Toensure that Cisco Unified Communications Manager sends signaling messages to the phone
and instructs it to listen to a G.711 stream,configure regions in this way:
•
Put the Cisco Unified Communications Manager MOH server or servers into a dedicated
region (for example. MOH).
•
Putall branch devices intoa site-specific region (forexample. Branch-1).
•
Allow G.711 between regions MOH and Branch-1.
• Make sure that region Branch-1 islimited to G.729 for calls loand from all other regions.
© 20i0 Cisco Systems. Inc.
Bandwidlh Managementand CAC Implementation
3-23
Each SRST or Cisco Unified Communications Manager Express router can stream up to six
different MOH files. You can configure each of them for multicast MOH or unicast MOH.
Therefore, the maximum number of multicast MOH audio sources diat can be used per remote
site is limited to six. By providing different MOH files for each site, site-specific MOH files
can be played for each site. Only G.711 codec is supported by SRST and Cisco Unified
Communications Manager Express.
When using multicast MOH also within the main site, you must enable multicast routing in
order to allow the multicast stream to be routed from the Cisco Unified Communications
Manager server network to the phone network or networks. If the MOH server is on the same
network that the IP phones are on, multicast routing is not required, but such a scenario is not
recommended, for security reasons (servers should be separated from endpoints).
3-22
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) w8.0
©2010 Cisco Systems, Inc.
Multicast MOH from Branch Router Flash
Implementation
1his topic describes how to implement the multicast MOH from branch router flash feature.
Multicast MOH from Branch Router
Flash Characteristics
Works only with multicast MOH.
Based on MOH capabilities of SRST (Cisco IOS router requires feature
set that supports SRST)
Cisco Unified Communications Manager must be configured to use
multicast MOH
IP phone is not aware that it listens to locallygenerated MOH.
Stream generated by MOH server is prevented from reaching the IP WAN.
Identical stream is generated locally at the branch sites.
Branch router can stream up to six MOH files.
Only G.711 is supported.
- Each stream can be selectively enabled for multicast.
• Up to six different MOH sources supported per branch router.
Multicast MOH from branch router flash is a feature that allows multicast MOI I streams to be
generated b\ gateways that are located at remote sites instead ofbeing streamed from the main
site to the remote site over the IP WAN.
Cisco Unified Communications Manager is configured for standard multicastMOH. Neither
Cisco IInilied Communications Manager northe phones that are located at theremote siteare
aware that the stream generated at the centra! site is replaced by a locally generated stream. The
multicast MOH stream that is generated by the centrally located MOH server is prevented from
traversing over IPWAN. and the remote site router generates a stream that has the same
attributes (codec, multicast address, and port).
As mentioned earlier, multicast MOII from branch router flash is based on multicast MOH. so
\ou must configure Cisco Unified Communications Manager to use multicast MOH instead of
unicast MOH. This configuration is recommended anyway in order to reduce load at the MOH
server by multicasting one stream that can be received by all devices, instead ofstreaming
MOH indi\idually for each endpoint in separate RIP sessions.
To generate a multicast MOI Istream at the remote site, you use features ofSurvivable Remote
Site Telephom (SRST) orCisco Unified Communications Manager Express, fherefore. the
remote site router thatwill generate the multicast MOI I stream forthedevices that are located
at the remote site has to beconfigured for SRST or Cisco Unified Communications Manager
Express. SRSI" does not have to be active (there is no need for a fallback scenario), because an
SRST gatewaj that is configured for multicast MOH streams MOH all the time, regardless of
its state (standby mode or SRST mode). The same principle applies toCisco Unified
Communications Manager Express: Only multicast MOH has tobe enabled, no further features
have to be enabled,and no phones have to be registered.
; 2010 Cisco Systems. Inc.
Bandwidth Managementand CAC Implementation
3-21
•
associate profile: To associate a DSP farm profile with a Cisco Unified Communications
Managergroup, use the associate profile command in SCCPCisco Unified
Communications Manager configuration mode.
Tip
The name that is specified in the Cisco IOS device must match the name in the Cisco
Unified Communications Manager exactly; the names are case-sensitive.
Note
When a Cisco IOS Enhanced Media Termination Point is being configured, any name can
be configured with the associate profile command. When a Cisco IOS conference bridge is
being configured, the name cannot be configured; it is MTP(AMC), where (MAC) is the MAC
address of the interface that was specified at the seep local command.
•
dspfarm profile: To enter DSP farm profile configuration mode and define a profile for
DSP farm services, use the dspfarm profile command in global configuration mode.
•
codec (dsp): To specify call density and codec complexity that is based on a particular
codec standard, use the codec command in DSP interface DSP farm configuration mode.
•
maximum sessions (DSP farm profile): To specify the maximum number of sessions that
are supported by the profile, use the maximum sessions command in DSP farm profile
configuration mode.
•
associate application seep: To associate SCCP to the DSP farm profile, use the associate
application seep command in DSP farm profile configuration mode.
•
no shutdown: If you fail to use the no shut command for the DSP farm profile, it will be
displayed in the gateway but will fail to operate.
To verify the Cisco IOS media resource configuration, use fhese show commands:
3-20
•
show seep: To check whether the Cisco IOS router successfully established a TCP
connection with the configured Cisco UnifiedCommunications Managersystem or systems
in order to exchange SCCP signaling messages, use the show seep command.
•
show seep ecm group [group-number]: To see which media resources are registered with
the Cisco Unified Communications Managersystem or systemsthat are configured in the
specified group, use the show seep ccm group / command.
•
show dspfarm profile [group-number]: To see the status of the media resource of the
specified profile at the Cisco IOS router, use the show dspfarm profile / command.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Step 2: Configure Transcoder Resource in Cisco IOS Software
The figure shows how to configure Cisco IOS Software to provide transcoding resources to
Cisco Unified Communications Manager.
Step 2: Configure Transcoder
Resource in Cisco IOS Software
Name lo Register Wilti at Cisco
cp ccm group 1
^iate ccm 1 priority 1
Unitied Communications
:iata profile 1 register HQ-1 XCODEB-*^-! Manager
dspfarm profile 1 transcode
~| The profile IDhas to match
codec g711ulaw
codec gTllalaw
For verification use:
codec g729ar8
codec g729abrfl
show seep
maximum
sessions
2
associate application SCCP
no
show seep ccm group 1
show dspfarm profile 1
shutdown
In theexample that is shown in the figure, a Cisco IOS Enhanced Media Termination Point
type transcoder is configured:
•
dspfarm (DSP farm): To enable DSP farm service, use the dspfarm command in global
configuration mode. The DSP farm service is disabled by default.
•
dsp sen ices dspfarm: To enable DSP farm services for a particular voice network
module, usethe dsp servicesdspfarm command in interface configuration mode.
•
seep local: To use Skinny Client Control Protocol (SCCP) to select the local interface that
is used to register the media resources with Cisco Unified Communications Manager, enter
the seep local command in globalconfiguration mode.
•
seep ccm: To use SCCP to add a Cisco Unified Communications Manager server tothe list
of available serversand set variousparameters—including IP addressor Domain Name
System (DNS) name, port number, and version number—use the seep ccm command in
global configuration mode.
•
•
seep: To enable the SCCP protocol and its related applications (for example, transcoding
and conferencing), use theseepcommand inglobal configuration mode.
seep ccm group: To create a Cisco Unified Communicalions Manager group and enter
SCCPCisco Unified Communications Managerconfiguration mode, use the seep ccm
group command in global configuration mode.
•
associate ccm: To associate a Cisco UnifiedCommunications Manager with a Cisco
Unified Communications Manager group and establish itspnority within the group, use the
associate ccm command in SCCPCisco Unified Communications Managerconfiguration
mode.
© 2010 Cisco Systems. Inc.
Bandwidth Management and CAC Implementation
Step 1: Add Transcoder Resource in Cisco Unified
Communications Manager
The figure shows how to add a transcoderresourcein Cisco Unified Communications Manager.
Step 1: Add Transcoder Resource in
Cisco Unified Communications Manager
You can freely choose the device name when using Cisco
IOS Enhanced Media Termination Point hardware.
In all other cases, it must be MTP followed by MAC
address.
Select transcoder type.
TrtMWdBi
Tip,"
C'tai (OS Mivad Hnba Ttrmnuliui Punf
DttcVJon
HQ-ilEhrtcfbdiisf Dttouro*
*>•..« Pwl*
DbUJI
^*\
|_
Enter device name and
description.
— i tf 1
< «tont >
V".
ttdqrm*ht)ti
Select device pool.
Navigate to Media Resources > Conference Bridge and click Add New. The Transcoder
Configuration window opens. Choose the type of Cisco transcoder media resource from these
options:
•
Cisco IOS Enhanced Media Termination Point
•
Cisco IOS Media Termination Point
•
Cisco Media Termination Point Hardware
•
Cisco Media Termination Point
Note
The type depends on the hardware that is used. For example, NM-HDV would require Cisco
IOS Media Termination Point to be selected while newer DSP hardware such as NM-HDV2
is configured as Cisco IOS Enhanced Media Termination Point.
Choosethe type of the Ciscotranscoder mediaresource, enter a device name and a description
for the transcoding resource, and then choose a device pool.
The device name has to match the name that is entered at the Cisco IOS router thai provides the
mediaresource. The name is case-sensitive. If the transcoding resource is provided by Cisco
IOS Enhanced Media Termination Point hardware, you can freely choose the name. In all other
cases,the name is MTPfollowed by ihe MAC addressof the interface that is configured to be
used for registering the media resource with Cisco Unified Communications Manager.
3-18
Implementing Cisco Unrfied Communications Manager, Part 2 (CIPT2)v8.0
>2010 Cisco Systems, Inc.
Configuration Procedure for Implementing Transcoders
"fhis subtopic presents the procedure for implementing transcoders.
Configuration Procedure for
Impiementing Transcoders
•
Add transcoder resource in Cisco Unified Communications
Manager.
2
Configure transcoder resource in Cisco IOS Software.
3
Configure media resource groups (MRGs).
4
Configure media resource group lists (MRGLs).
G
Assign MRGLs to devices.
To implement transcoders. perfonn these steps:
Step 1
Add transcoder resource in Cisco Unified Communications Manager.
Step 2
Configure transcoder resource in Cisco IOS Software.
Step 3
Configure MRGs.
Step 4
Configure MRGI.s.
Step 5
Assign MRGLs to devices.
Note
Only the first two steps of the procedure are presented on the following pages, because the
last three steps—the configuration of MRGs, the configuration of MRGLs, and the
assignmentof MRGLs to devices—are discussed inthe Implementing Cisco Unified
Communications Manager, Part 1 (CIPT1) course.
©2010 Cisco Systems. Inc
Bandwidth Management and CAC Implementation
3-17
Example: Implementing a Transcoder at
the Main Site (Cont.)
Region: BR
Region: HQ
Region: HQ
Region Configuration:
Remote Site
Within HQ: G.711
Main Site
HQ to BR: G.729
Transcoder
Within BR: G.711
The figure illustrates how the solution described earlier in this topic is implemented in Cisco
Unified Communications Manager.
All headquarters devices (phones, voice-mail system, software conference bridge, and the
transcoder) are in region HQ. Remote site phones are in region BR.
Cisco Unified Communications Manager region configuration allows G.711 to be used within
region HO and within region BR. Calls between regions HQ and BR are limited to G.729.
When a call is placed from a remote site phone to the voice-mail system, Cisco Unified
Communications Manager identifies the need for a transcoder that is based on the capabilities
of the devices(G.711 only at the voice-mail system)and the maximum permittedcodec
(G.729). A device may support only a codec with higher bandwidth requirements than
permitted by the region configuration, for example. If such a device can access a transcoder. the
call is set up and invokes the transcoder resource. The call would otherwise fail.
3-16
Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0
) 2010 Cisco Systems, Inc
Example: Implementing a Transcoder at the Main Site
The figure illustrates an example of implementing a transcoder at the main site.
Example: Implementing a Transcoder at
the Main Site
G.729 audio streams are transcoded at the main site before
being passed on to endpoints supporting only G.711.
Cisco Unified
Communications
Manager
At the main site, there are two devicesthat supportCi.711 only. One device is a Cisco Unified
Communications Manager software conference bridge; theother device is a third-party voicemail application.
Regions are configured in such a way that all voice traffic between the remote site and the main
site has to use the G.729 codec.
When a user at a remote site needs to be added to a conference via the software conference
bridge, the user cannot be added, because G.729 must be used over the IP WAN but only G.711
is supported by the conference bridge.
By adding a transcoder resource at the main site gateway, you enable the remote site user to
send a G.729 voice stream, which is transcoded to G.711 and passed on to the conference
bridge b\ the transcoder that is located at the main site.
The same approach can beused for calls to the voice-mail system from the remote site.
) 2010 Cisco Systems, Inc
Bandwidth Management and CAC Implementation
Number of phones at remote site and number of calls that are placed toG.711-only
phones over the IP WAN: How many phones are located at the remote site? How often do
the phones need to communicate to phones located at the headquarters that support G.711
only and hence require a transcoder when G.729 must be used over the IP WAN? How
many of these calls occur at the same time?
Available bandwidth and cost of additional bandwidth: Is there enough bandwidth (or
can additional bandwidth be provisioned) to allow G.711 for calls to devices that do not
support G.729? How does the cost of adding bandwidth compare to the cost of deploying
local DSPs?
3-14
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Transcoder Implementation
This topic describes how to implement transcoders in order to allow low-bandwidth codecs to
be used when they are not supported by both endpoints.
Considerations When Deploying
Transcoders
Allows low-bandwidth codecs to be used in the IP WAN when
one or both endpoints do not support low-bandwidth codecs:
Affected endpoint uses high-bandwidth codec toward the
transcoder.
Transcoder changes voice stream from high-bandwidth
codec to low-bandwidth codec.
Low-bandwidth codec stream is sent to other device
(or transcoder) over the IP WAN.
Efficiency depends on several factors:
Number of devices at remote site and likelihood of requiring
a transcoder
- Ease of adding required hardware resources (DSPs).
Available bandwidth toward central site.
As mentioned earlier in this lesson, transcoders arc devices that transcode voice streams, fhat
is. they changethe way that the audio payload is encoded(for instance, G.711 audio streams
are changed to G.729 audiostreams). Transcoders arc deployedin order to allow the use of
low-bandwidth codecs over the IP WAN even if one of the endpoints supports only highbandwidth codecs such as G.711.
The transcoder hasto be deployed close to the device thatsupports only G.711. Thatdevice
will send a G.711 stream to the transcoder. which transcodes the audio to a low-bandwidth
codec such as G.729. The G.729 voice stream is then sent from the transcoder to the other
device— a phone that is located at a remotesite—overthe IP WAN.
Note
Itis important forthe MRGL to know that the devicethat is limited to the higher bandwidth
codec is the one that will request the transcoder media resource. If, for example, onlyG 729
is permitted between two IPphones, butone IP phone supports only G.711, the phone that
cannotcomply with the permitted codec (G.729, inthiscase) is the one that will requesta
transcoder. Therefore,the MRGL of this phone has to have access to a transcoder, which
should be physically located close to the requesting device. Regions have to be set up in
such a waythat the requesting phone is allowed to use G.711 to the transcoder(notethat
this call leg is also subject to region configuration).
Refore deploying transcoders. you must consider some factors that are like the factors that must
be considered when you deploy local conference bridges. Here arethe factors:
• Cost of adding DSPs: Is it necessary lo add DSPs to anexisting router only, or docs t le
whole platform have to be replaced?
) 2010 Cisco Systems. Inc
Bandwidth Management and CAC Implementation
3-13
Example: Implementing Local
Conference Bridges at Two Sites (Cont.)
HQ_SWMRG
Main Site
Cisco Unified
Communications
Manager
The figure illustrates how Media Resource Groups (MRGs) and Media Resource Group Lists
(MRGLs) are used to ensure that headquarters phones use the conference resources at the
headquarters and that remote site phones use the remote site conference resource when
establishing a conference.
These three MRGs are created:
•
HQ_HW-MRG: Includes the hardware conference bridge that is provided by the voice
gateway that is located at the headquarters
•
IIQ_SW-MRG: Includes the software conference bridge that is provided by a Cisco
Unified Communications Manager server that is located at the headquarters
•
BR_H\V-MRG: Includes the hardware conference bridge that is provided by the voice
gateway that is located at the remote site
The HQ HW-MRG is the first entry of the MRGL, which is called HQ_MRGL; (he HQ SWMRG is the next entry. Headquartersphones are configured with the HQ_MRGL. Because
MRGs arc used in a prioritized way, headquarters phones that invoke a conference will first use
the available hardware conference resources; when all of them are in use, the software
conference resources are accessed.
At the remote site, all phones refer to the BRMRGL, which includes only the BRJIW-MRG.
This configuration allows remotephonesto use their local conference bridge when they invoke
conferences instead of accessing conference resources that are located across the IP WAN.
3-12
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
J2010 Cisco Systems, Inc.
Example: Implementing Local Conference Bridges at Two Sites
The figure shows a sample scenario for deploying local conference bridge resources at a remote
site.
Example: Implementing Local
Conference Bridges at Two Sites
Phones at the remote site should use a local
conference bridge when creating a conference.
Main Site
Cisco Unified
Communications
Manager
WAN
Remote Site
The figure shows a main site with software and hardware conference resources. At the remote
site, hardware conference resources are added lo the remote site gateway. As a result, the
remotesite phones can set up conferences by using local resources insteadof by always
accessing the conference resources that are located at the main site. For conferencing remote
site members onlv. no traffic has to be sent across the IP WAN.
Note
When an ad hoc conference includes members of separate sites, a separate voice stream
for each remote member has to be sent across the IP WAN. However, if a Meet-Me
conference is set up, the users that are located at the remote site could first establish an ad
hoc conference (by using a media resource that is localto the remote users) and then add a
call to the remote Meet-Me conference to their local ad hoc conference. In this case, there is
onlya single voice stream that is sent across the IP WAN connecting the twoconferences.
©2010 Cisco Systems.
Bandwidth Management and CAC Implementation
3-11
Local Conference Bridge Implementation
"fhis topic describes howyou can implement local conference bridges at remote sites in orderto
keep voice streams local when all members of the conference are located at thesame site.
Considerations When Deploying Local
Media Resources at Remote Sites
• Trafficstays offthe IP WAN when remote site devices require a
media resource—for example, conference bridges.
- Local media resources require appropriate hardware at remote
site (DSPs in gateway).
« Efficiency depends on several factors:
- Number ofdevices at remote site and likelihood of using
a feature or application requiring the media resource
- Ease of adding required hardware resources (DSPs)
- Available bandwidth toward central site
When local conference bridges or Media Termination Points (MTPs) are deployed at each site,
traffic does not have to cross the IP WAN if all endpoints are located at the same site. You can
implement local media resources such as conference bridges and MTPs by providing
appropriate hardware (digital signal processors [DSPs]) at the routers that are located at the
remote sites.
Whether the extra cost for providing the DSP resources will be worthwhile depends on several
factors:
•
Cost of adding DSPs: Is it necessary to add DSPs to an existing router only, or does the
whole platform have to be replaced?
•
Number of devices at remote site and likelihood of using applications or features that
require access to the media resource that is considered to be locally deployed: How
many phones are located at the remote site? How often do the phones use features that
require a media resource that is currently available only over the IP WAN? What is the
maximum number ofdevices that require access to the media resource at the same time?
•
3-10
Available bandwidth and cost of additional bandwidth: Is (here enough bandwidth (or
can additional bandwidth be provisioned) to accommodate the requirements that are
determined by the preceding factors? How does the cost of adding bandwidth compare to
the cost of deploying local DSPs?
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
J2010 Cisco Systems, Inc.
Example: Codec Configu ration (Cont)
Ipffcw OvftfluraJk*
Pwk-ti ctrfifteBfaifon
ftevw JnfonmHIon————
flimt
E4L_0|.2ftai
•teuton RalBilQn4jiw* - - *mli*0
Hm,*mm+****t*
5=_J-
- .bpif^ -si.
•a_)>
s *!>> i s ' : «
M5,si>ii-tt
e "til (S :«;
wq_i-<,-«
Sites ts'Jf,
/\
HQ_phones uses G.711 within
its own region and to region
HQ_gw; to all other regions,
BR_phones uses G.711 within
G.729 is used
G.729 is used.
its own region and to region
BR_gw, to all other regions,
"fhe figure illustrates region configuration in Cisco Unified Communications Manager for the
discussed scenario. The configuration of the HQ_phones and the BR_phones regions is
illustrated. Both regions are configured in such a way that calls within the region and calls to
the local gateway (regions HQ_gw and RR_gw) arc allowed to use (i.7l I. while calls to all
other regions are limited to G.729.
Note
The preceding example is a partial configuration only. It does not show the configuration of
the other regions.
© 2010 Cisco Systems, Inc.
Bandwidth Management and CAC Implementation
Example: Codec Configuration
The figure shows a sample scenario for codec configuration in Cisco Unitied Communications
Manager.
Example: Codec Configuration
Region Configuration
Within MQ_phorie9 G 711
WitntnHQ_gw G711
HQ_ptionestoHQ_gw G.711
Within BR_pnonee G 711
Within BRjuf G 711
BR_phones to BR_gv/ G.711
All others G 729
Remote Site
Region: HQ_phones
Region BR_phones
In the figure, phones that are located in the headquarters are configured with region
HQ_phones. An intercluster trunk that connects to another Cisco Unified Communications
Manager cluster and a Session Initiation Protocol (SIP) trunk connecting to an Internet
telephony service provider (ITSP) are in region HQ_trunks. The public switched telephone
network (PSTN) gateway that is located in the headquarters is configured with region HQ_gw.
At the remote site, phones are in region BR_phones and the PSTN gateway is in region
BR_gw.
Cisco Unified Communications Manager regions are configured in the following way:
Within HQ_phones: G.711
Within IIQ_gw: G.711
HQ_phones to HQ_£w: G.711
Within BR_phones: G.711
Within BR_gw: G.711
BR_phones to BR_gw: G.711
All others: G.729
As a result, of this configuration, all calls that use the IP WAN between the remote site and the
headquarters use G.729. Calls that are sent through the intercluster or SIP trunk use G.729 as
well. These calls use G.711: calls between phones within the headquarters, calls between
phones within the remote site, calls from headquarters phones to the headquarters PS'fN
gateway, and calls from remote site phones to the remote site PSTN gateway.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
>2010 Cisco Systems, Inc.
Review of Cisco Unified Communications Manager Codecs
This subtopic reviews how to control the codec that is used for a call in Cisco Unified
Communications Manager.
Review of Cisco Unified
Communications Manager Codecs
The codec that will be used depends on Cisco Unified
Communications Manager region configuration:
• Each region is configured with the highest permitted codec bandwidth
- Within the configured region.
Toward specific other regions (manually added).
Toward all other regions (that have not been manually added)
• Region is assigned to a device pool.
• Device pool is assigned to a device.
• Codec that is actually used depends on capabilities of the two devices:
Best codec that is supported by both devices and does not exceed
codec bandwidth permitted in region configuration.
If devices cannot agree on a codec, a transcoding device is invoked.
Loss type (configured in region) is also considered.
The codec that will be used for a call depends on the Cisco Unified Communications Manager
region configuration. Eachdevice is assigned with a regionvia the device pool configuration.
For each region, the administrator can configure the highestpermitted codec bandwidth within
a region, to other specifically listed regions, and to all other (not listed) regions.
Whena call is placed between two devices, the codec is determined based on the regionsof the
two devices and on the capabilities of the devices: The devices will use the bestcodec that is
supported by both devices and that doesnotexceed the configured codec bandwidth forthe
region or regions that are involved in the call. If the two devicescannotagree on a codec (for
instance, if region configuration allowsonly 8 kb/s as the maximum codec bandwidth but one
device supports only G.711). a transcoder is invoked, if available. The losstypeof a linkcan
alsobe configured. On links that are configured to be lossy, codecs that are lesssensitive to
packet lossarepreferred overcodecs thatresult in higher-quality degradation. Formore details
about codec selection, refer to the Implementing Cisco Unified Communications Manager,
Part / (CIPTI) course.
)2010 Cisco Systems, Inc.
Bandwidth Management and CAC Implementation
Cisco Unified Communications Manager Codec
Configuration
This topic describes how to configure Cisco Unified Communications Manager in order to limit
the codec that is used for a call.
Codec Configuration Considerations
• Use high-bandwidth codecs in LAN environments,
• Use low-bandwidth codecs for IP WAN.
* Low-bandwidth codecs are designed for human speech:
- Do not work well for other audio streams, such as music.
-
Use alternative methods for MOH:
• Disable MOH for remote sites.
• Use multicast MOH from branch router flash.
To conserve IP WAN bandwidth, you should use low-bandwidth codecs in the IP WAN. For
calls within a LAN environment, you should use high-bandwidth codecs for optimal audio
quality. Whenyou are designing where to use which type of codec, it is important to consider
that low-bandwidth codecs such as G.729are designed for human speech. They do not work
well for other audio streams, such as music.
As stated in the previous topic, other methods exist for limiting the bandwidth that is required
for MOH streams. If you cannot use multicast MOH from branch router flash but MOH streams
are not desiredon the IP WAN. you can disable MOHfor remotesite phones.
3-6
Implemenling CiscoUnified Communications Manager, Part 2 (CIPT2) v8,0
) 2010 Cisco Systems, Inc.
Other bandwidth management solutions includethe use of transcoders or the implementation of
special features such as multicast MOH from branch router flash. Transcoders are devicesthat
can transcode voice streams. That is. they change the way that the audio payload is encoded
(for instance, a G.711 audio stream is changed to a G.729 audio stream). Transcoders allow the
use of low-bandwidth codecs over the IP WAN even if one of the endpoinls is limited to a
high-bandwidth codec such as G.711. Multicast MOII from branch router Hash allows a
multicast MOH stream lo be generated by a Cisco IOS router that is located at the remote site,
instead of being sent over the IP WAN from a centralized MOH server.
©2010 Cisco Systems. Inc.
Bandwidth Management and CAC Imptementation
Bandwidth Management Overview
fhis topic describes bandwidth management options in a Cisco Unified Communications
Manager deployment.
Examples of Bandwidth Management
Options
Bandwidth in the IP WAN can be conserved by:
• Techniques that reduce required bandwidth of voice streams:
- RTP header compression (as part of QoS link efficiency
mechanisms)
-
Low-bandwidth codecs
• Techniques that influence the flow of voice streams:
- Local media resources such as conference bridges
• Other solutions:
- Transcoders
- Multicast MOH from branch router flash
You can use varioustechniques to conserve valuable IP WANbandwidth. One technique is to
reducethe requiredbandwidth of voice streamsby using Real-Time Transport Protocol (RTP)
header compression (which is a qualityof service [QoS]link efficiency mechanism), Another
technique is to use low-bandwidth audio codecs. You can also use a combination of these two
solutions.
Note
Refer to the "Quality of Service" moduleof the Implementing Cisco Voice Communications
and QoS (CVOICE) course for more detailed discussion about QoS.
Otheroptions for managing IP WANbandwidth are techniques that influence where voice
streams are sent. If three phones, all located at a remotesite, establishan ad hoc conference,
there is a greatdifference in bandwidth usage if the conference bridge is located at that remote
site—local to the phones that are members of the conference—or if the conference is located at
the main site and has to be accessed over the IP WAN. Inthe latter case, all three phones are
sending their voice stream tothe conference bridge over the IP WAN. The conference bridge is
mixing thereceived audio and is then streaming it back to all conference members (in three
separate streams). Although the call appears to be local to the remote site—because all
conference members are located atthat site—due tothe remotely located conference bridge, the
IP WAN is occupied by three calls.
3-4
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
© 2010 Cisco Systems, Inc.
Lesson 1
Managing Bandwidth
Overview
When an IP WAN connects various sites in a Cisco Unified Communications network,
bandwidth consumption at the IP WAN should beminimized. Several techniques can help
conserve bandwidth on the IP WAN in a multisite deployment;
•
Reducing the required bandwidth of voice streams
•
•
Keeping some \oice streams (such as local media resources) away from the IP WAN
Employing special features like multicast music on hold (MOH) from branch router flash
(or the use of transcoders).
"fhis lessondescribes all these techniques and features and their implementation.
Objectives
Upon completing this lesson, you will be able to describe techniques to reduce bandwidth
requirements on IP WAN links in Cisco Unified Communications Manager multisite
deployments. This ability includes being able tomeet these objectives:
•
Describe methods to minimize bandwidth requirements for Cisco Unilied Communications
• Configure Cisco Unified Communications Manager in order tocontrol the codec that is
used for a call
• Implement local conference bridges in order to avoid accessing conference bridges over the
IP WAN even if all participants are local
• Implemeni transcoders in order to allow low-bandwidth codecs to be used iflow-bandwidth
codecsare not supported by both endpoints
• Implement multicast MOH from branch router flash to avoid MOH streams over the IP
WAN
3-2
Implementing Cisco Unrfied Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems, inc.
Module 3
Bandwidth Management and
CAC Implementation
Overview
Individual sites of a multisite deployment are usually interconnected by an IP WAN in
situations in which bandwidth is relatively costly and bandwidth consumption should be
minimized. Call Admission Control (CAC) is alsoimportant in a multisite environment in order
to avoid bandwidth oversubscription by too many voice calls.
This module describes methods of reducing bandwidth requirements and explainshow to
design and implement those methods so that the IP WAN bandwidth can be used as efficiently
as possible. It also describes CAC options for intrasile calls and intersite calls and how to
implemeni them in amultisite Cisco Unified Communications Manager deployment. The
module also describes automated alternate routing (AAR), which allows the public switched
telephone network (PSTN) to be used as a backup for calls that CAC denies due to insufficient
bandwidth.
Module Objectives
Upon completing this module, you will be able to implement bandwidth management and CAC
to prevent oversubscription ofthe IP WAN. This ability includes being able to meet these
objectives:
• Describe techniques to reduce bandwidth requirements on IP WAN links in Cisco Unified
Communications Manager multisite deployments
•
Describe and configure CAC mechanisms and AAR in Cisco Unified Communications
Manager and in gatekeepers
Module Self-Check Answer Key
QD
C
02)
B. C
03)
voice service voip
h323
no h225 timeout keepalive
2-102
Q4)
A,D
05)
D
06)
B
Q7)
B
Q8)
D
09)
D
010)
A
Qll)
D
Q12)
B.C
Q13)
A,D
Q14)
B.C
Q15)
A
Impiementing Cisco Unified Communications Manager, Part 2 (CIPT2) vB.O
©2010 Cisco Systems, Inc.
Q12)
Which two statements about Cisco Unified Communications Manager Express are
true? (Choose two.) (Source: Implementing Cisco Unified Communications Manager
Express in SRST Mode)
A)
IP phonesregisterwith Cisco Unified Communications Manager Express in
standalone mode when Cisco Unified Communications Manager Express is
partof the Cisco Unified Communications Manager group (hat is specified in
the device pool of the phone.
B)
C)
During SRST fallback. IP phonesregisterwith Cisco Unified Communications
ManagerExpress in SRSTmode when Cisco Unified Communications
ManagerExpress is configured as the SRST reference for the IP phone.
Cisco Unified Communications Manager Express in SRST modeprovides
more features than standard SRST.
D)
The same platform can serve more phones when running Cisco Unified
Communications Manager Express in SRST mode versus running standard
E)
Standalone Cisco Unified Communications Manager Express routerscan be
SRST,
clustered for redundancy.
013)
Which two features have been added in Cisco Unified Communications Manager
Express Release 8.0? (Choose two.) (Source: implementing Cisco Unitied
Communications Manager Express in SRST Mode)
014)
Q15)
A)
fi\ e additional MOI 1sources
B)
C)
D)
presence with BLE status
\ideo support
demote argument of the dialplan pattern command
E)
local MOII
Whichtwo commands are notconfigured in telephony-service configuration mode?
(Choose two.) (Source: Implementing Cisco Unified Communications Manager
Express in SRST Mode)
A)
create cnf-files
B)
C)
D)
ephone
ephonc-dn
max-ephoncs
E)
max-dn
F)
ip source-address
Which statement about Cisco Unified Communications Manager Express in SRST
mode is not true? (Source: Implementing CiscoUnified Communications Manager
Express in SRST Mode)
A)
B)
C)
D)
If only the ephone is preconfigured, only the ephone-dn is learned by SNAP.
If only the ephone-dn is preconfigured, only theephone is learned bySNAP.
If ephone and ephone-dn arepreconfigured, SNAP is notused.
If neither ephone norephone-dn areprcconfigured, ephone and ephone-dn are
learned bv SNAP.
>2010 Cisco Systems, Inc
Centralized Call-Processing Redundancy Implementalion
2-101
Q6)
When implementing MGCP Fallback and SRSf, which configuration is not performed
at Cisco Unified Communications Manager? (Source: Implementing SRST and MGCP
Fallback)
Q7)
A)
B)
adding SRST references
enabling MGCP fallback at the MGCP gateway configuration page
C)
D)
applying SRST reference to phones
The SRST reference is configured under System > Enterprise Phone Parameters.
(Source: Implementing SRST and MGCP Fallback)
A)
B)
Q8)
configuring CFUR to reach remote-site phones during SRST mode
true
false
Which command is used for SRST configuration at the Cisco IOS router? (Source:
Implementing SRST and MGCP Fallback)
A)
B)
C)
D)
Q9)
QIO)
telephony-server
ccm-manager fallback
service alternate default
call-manager-fallback
Which command is not used for MGCP Fallback configuration? (Source:
Implementing SRST and MGCP Fallback)
A)
B)
C)
D)
ccm-manager fallback-mgcp
application
global
telephony-server
E)
service alternate default
Wheredo you configure the maximumnumberofhops that can be used by CFUR?
(Source: Implementing SRST and MGCP Fallback)
A)
B)
C)
D)
Q1I)
service parameter
enterprise parameter
SRST gateway configuration
phone configuration
How cancalling privileges be implemented for SRST individual phones? (Source:
Implementing SRST and MGCP Fallback)
A)
only when using Cisco Unified Communications ManagerExpress in SRST
mode
B)
C)
D)
2-100
by preconfiguring the phonesthat need callingprivileges assigned
by configuring an ephone-dn template
by configuring COR lists for directory numbers
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Module Self-Check
Use the questions hereto review whatyou learned in this module. Thecorrect answers and
solutions are found in the Module Self-Check Answer Key.
Q1)
Which of these provides redundancy for MGCT-controlledgateways? (Source:
Examining Remote Site Redundancy Options)
A)
B)
C)
D)
Q2)
MGCP SRST
SRST fallback
MGCP fallback
MGCP in SRST mode
Which two types of calls arc notpreserved during switchover of an SRSf gateway?
(Choose two.) (Source: Examining Remote Site Redundancy Options)
A)
B)
calls between IP phones that are located at the remote site
conference calls of remote-site phones using a conference bridge that is located
C)
D)
E)
calls between IP phones that arc located at the remotesite and at the main site
calls between IP phones that are located at the main site
calls from main-site phonesthat wereplaced to a remote-site phone and then
transferred from the remote-site phone to another main-site phone
at the main site
Q3)
Which configuration is required lo allow calls lo bepreserved during switchback from
H.323 to MGCP? (Source: Examining Remote Site Redundancy Options)
Q4)
Q5)
What are the two correct statements of supported phones in SRST for the given
platform? (Choose two.) (Source; Examining Remote Site Redundancy Options)
A)
800: 4
B)
C)
D)
E)
2801:500
2851:350
3825:350
3845:1024
What can you use toconfigure the dial plan ata remote-site gateway insuch a way that
branch users can stillreach the headquarters when dialing internal directory numbers
during fallback? (Source: Examining Remote Site Redundancy Options)
A)
Ihis is not possible. Users have to dial headquarters users by their PSTN
numbers while in fallback mode.
B)
C)
D)
i2010 Cisco Systems. Inc.
Use translation profiles modifying the callingnumber.
Issue the dialplan-pattern command.
Use translation profiles modifying the called number.
Centralized Call-Processing Redundancy Implementation
Cisco Systems. Inc. Number Translation Using Voice Translation Profiles, February 2006,
http://w\\w.cisco.coni,'eii/US/tech/tk652/lky()/technolo«ies configuration__exantpld)9IS6a0
0803f818a.shtml
Cisco Systems. Inc. Voice Translation Rules, November 2006.
http:/.''www,eisco,com/en/IJS/lcch/tk652/lk90/lechnologies_tech_note09l86a0080325e8e.s
html
Cisco Systems. Inc. Cisco Unified Communications Manager Express System
Administrator Guide, November 2007 with updates 2010.
hup://ww w,cisco.com/en/US/docs/voice ip comm/cucme/admiii/coiifiguration/guide/cmea
dm.html
2-9S
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Module Summary
This topic summarizes the key points that were discussed in this module.
Module Summary
Cisco Unified Communications Manager supports features
that provide remote phones with redundancy in case of an IP
WAN failure.
Cisco Unified SRST allows remote site phones to register at
a local router that performs call processing. MGCP gateways
that lose the connection to their call agents can fall back to
H.323 or SIP.
Cisco Unified Communications Manager Express can be
used in SRST mode to provide more features than standard
Cisco Unified SRST in the event of an IP WAN failure.
Ihis module described the available features for providing remote phones with backup in the
event of an IP WAN outage. It explained call survivability. Media Gateway Control Protocol
(MGCP) fallback, and Cisco Unified Survivable Remote Site Telephony (SRS'f). fhe module
also contrasted the differences between standard SRST and Cisco Unified Communications
Manager Express in SRST mode. In addition, itdescribed how to implement standard SRST
and a dial plan to support intersite connectivity through the public switched telephone network
(PSTN), as well as PSTN access during IP WAN failure. Finally, the module showed how to
implement a backup solution using Cisco Unified Communications Manager Express in SRST
mode instead of using standard Cisco Unified SRST.
References
For additional information, refer to these resources:
•
Cisco Systems. Inc. Cisco Unified Communications System 8.x SRND, April 20IC.
hup://\\\\w,cisco.com •'en.TJS/d()cs/voicejp_comni/ciiein/snid/8x/uc8x.hlinl
•
Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide
Release 8.011). February 2010.
http://www.cisco.coni/cn/t'S/docs/v()icejp_coinm/cLicni/admin/8J) l/ecmcfg/bccm-KOIcm.html
•
Cisco Systems. Inc. Cisco Unified Survivable Remote Site Telephony Version 8.0.
November 2009,
lntp://w\\w.cisco.eoni/en/US/prod/collateral/voicesxv/ps6788/vcallcon/ps2l69/daUi .sheet_>
78-570481.html
©2010 Cisco Systems, Inc.
Centralized CalI-Processing Redundancy Implementation
2-97
2-96
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8,0
©2010 Cisco Systems, Inc.
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
Cisco Unified Communications Manager Express cannot be a
member of a Cisco Unified Communications Manager cluster.
Cisco Unified Communications Manager Express supports
Cisco Extension Mobility, phone and media security, busy
lamp field (BLF), and video.
Basic configuration of Cisco Unified Communications
Manager Express includes telephony-service configuration,
ephone configuration, and ephone-dn configuration.
The srst mode command in telephony-service configuration
mode is required to allow Cisco Unified Communications
Manager Express to learn ephones and ephone-dns via
SNAP
References
For additional information, refer to this resource:
•
Cisco Systems. Inc. Cisco Unified Communications Manager Express System
Administrator Guide. November 2007 with updates 2010.
http:'•'www.cisco.coiTt/en.-rS/docs.'Voice ip coiiim-'ciiciiie/adniin/cunliguration/guide/cnica
dm.html
) 2010 Cisco Systems. Inc
Centralized Call-Processing RedundancyImplementation
2-95
Cisco Unified Communications Manager Express in SRST
Mode Configuration Example
The figure shows a configuration example of Cisco Unified Communications Manager Express
in SRST mode.
Cisco Unified CME* in SRST Mode
Configuration Example
Cisco
Unity
Cisco Unified
Communications
Manager
telephony-service
erst mode Auto-provision none
srst
dn
line-mode dual
srst aphone templata 1
srst dn
template 3
srst ephone description CUCME-SRST
ephone-template 1
keep-conference local-o Qly
ephone-dn-template 3
hold-alert 25
idle
In the example. Cisco Unified Communications Manager Express uses ephone template 1 for
newly added phones. This template configures conferences to drop if no internal members are
left in the conference.
Ephone-dns, which are learned using SNAP, are configured to alert the user if a call is on hold
for 25 seconds and the phone is idle.
For easier distinction, the description of learned phones should include an SRS'f string. The
ephone-dns should be dual-mode lines.
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Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) w8.0
) 2010 Cisco Systems, Inc.
Note
Ifsingle-line ephone-dns is used with multiline features likecall waiting, Call Transfer, and
conferencing, there must be more than one single-line directory number on a phone.
Configuring Cisco Unified Communications
Manager Express in SRST Mode (Cont)
CHERouter(confia-telephonv)#
srst dn template template-tag
• Specifies an ephone-dn template to be used in SRST on a Cisco
Unified Communications Manager Express router. Range is 1 to 15.
CHERouter(config-telephony)#
srst ephone template template-tag
• Specifies an ephone template to be used in SRST mode on a Cisco
Unified Communications Manager Express router. Range is 1 to 20.
CHERouter (conf id-telephonv) tt
srst ephone description string
• Specifies a description to be associated with an ephone learned in
SRST mode on a Cisco Unified Communications Manager Express
router Maximum is 100 characters
To specify an ephone-dn template to be used in SRST mode on a Cisco Unified
Communications Manager Express router, use the srst dn template command in telephonyservice configuration mode:
•
Theparameter template-tag is the identifying number of an existing ephone-dn template.
The range is from I to 15.
To specifv anephone template to beused in SRS'f mode on a Cisco Unified Communications
Manager Express router, use the srst ephone template command intelephony-service
configuration mode:
•
The parameter template-tag is the identifying number of anexisting ephone template. The
range is from 1 to 20.
To specifv a description to be associated with anephone in SRST mode on a Cisco Unified
Communications Manager Express router, use the SRST ephone description comma.id in
telephony-service configuration mode:
•
Maximum length of the parameter string is 100characters.
© 2010 Cisco Systems. Inc
Centralized Call-Processing Redundancy Implementation
Configuring Cisco Unified Communications Manager Express
in SRST Mode
The configuration of Cisco Unified Communications Manager Express in SRST mode is
performed in telephony-service configuration mode. When the telephony-service command is
active, the call-manager-fallback command is not accepted by the CI,I, and vice versa.
Configuring Cisco Unified Communications
Manager Express in SRST Mode
CHERouter(config-telephony)#
srst mode auto-provision {all | dn | none}
• Enables SRST mode for a Cisco Unified Communications Manager
Express router
- all writes information for leamed ephones and ephone-dns into
the running configuration.
- dn writes information for learned ephone-dn into the running
configuration.
- none does not include information for learned ephones or
ephone-dns in the running configuration.
CHERouter(confiq-telephony)#
srst dn line-mode {dual | single}
• Specifies the line mode for ephone-dns in SRST mode on a Cisco
Unified Communications Manager Express router. Default is single
line
To enable SRS'f mode for Cisco Unified Communications Manager Express, use the srst mode
auto-provision command in telephony-service configuration mode:
•
The keywordall includes information for leamed ephonesand ephone-dns in the running
configuration.
•
The keyword dn includes information for learnedephone-dns in the runningconfiguration.
•
The keyword none does not include information for learnedephonesor ephone-dns in the
running configuration. Use this keyword when Cisco UnifiedCommunications Manager
Express is providing SRSTfallback services for Cisco Unified Communications Manager,
Note
Ifthe administrator saves the running configuration after learning ephones and ephone-dns,
the fallback IP phones will be treated as locally configured IP phones on the Cisco Unified
Communications Manager Express SRST router,whichcould adversely impact the fallback
behavior of those IP phones.
To specify the line mode for the ephone-dns that are automatically created in SRST mode on a
Cisco Unified Communications ManagerExpress router,use the srst dn line-mode command
in telephony-service configuration mode. The keywords provide these specifications:
2-92
•
The keyword dual specifies dual-line ephone-dns.
•
The keywordsingle specifies single-line ephone-dns (the default).
Implementing Cisco Unified Communications Manager,Part 2 (CIPT2) vB.O
>2010 Cisco Systems, Inc.
N
Phone Registration Process
This section describes the phone registration process.
Phone Registration Process
When an IP phone registers to Cisco Unified
Communications Manager Express in SRST mode during
fallback:
• The router searches for an existing ephone with the MAC address of
the phone.
tf found, tho ephone must be configured with a directory number
(button command), or automatjcassignmentmust be enabled.
If not found, the router searches for an existing ephone-dn that
matches the IP phone directory number (learned by SNAP).
• Iffound, an ephone is added that refers to the matched
ephone-dn
• If not found, an ephone is added that refers to a newly
created ephone-dn autoconfigured by SNAP.
• Whenever an ephone is added by Cisco Unified SRST, the srst
ephone template is applied.
• Whenever an ephone-dn is added by Cisco Unified SRST, the srst
ephone-dn template is applied.
When a phone loses connectivity to the Cisco Unified Communications Manager, itregisters to
its configured SRST reference.
If that SRST reference is Cisco Unified Communications Manager Express in SRS'f mode, the
Cisco Unified Communications Manager Express router firstsearches for anexisting
(preconfigured) ephone with the MAC address ofthe registering phone. Ifthe router finds an
ephone. the stored ephone configuration isused. No phone configuration settings that arc
pro\ ided b> SNAP are applied, and no ephone template is applied. Ifthe configured ephone is
configured with one ormore ephone-dns. the stored configuration isused for the ephone-dn or
ephone-dns ofthe phone. Neither the information that is provided by SNAP nor the ephone
template thai isconfigured under telephony-service isapplied. Ifthe configured ephone isnot
configured with an ephone-dn. automatic assignment has tobe enabled for the phone tobecome
associated with an ephone-dn. SNAPis no option in this case.
Ifno ephone is found for the MAC address ofthe registering phone. Cisco Unified
Communications Manager Express adds the ephone (and applies the ephone template, if
configured), using SNAP. Ifthe directory number exists, it is bound tothe added phone:
otherwise, the directory number islearned using SNAP. Ifconfigured, the ephone-dn template
is applied.
© 2010 Cisco Systems, Inc
Centralized Call-Processing Redundancy Implementation
2-91
Advantages of Cisco Unified Communications Manager
Express in SRST Mode
UsingCisco Unified Communications ManagerExpress in SRST modehas several advantages,
when compared with using standard SRST.
Advantages of Cisco Unified
Communications Manager Express in
SRST Mode
• Provides many more features than standard SRST
- Allfeatures of Cisco Unified Communications Manager
Express can be used.
* Allows a mix of preconfigured phones and directory numbers,
and phones and directory numbers learned by SNAP
- Only those phones and directory numbers that require
individual configuration have to be preconfigured.
- Pre configuration can be based on MAC address or on
directory number.
- Phones that do not require individual settings do not have
to be preconfigured.
- Additional settings can be applied to phone and directory
number configuration leamed by SNAP, using ephone and
ephone-dn templates.
These are the main advantages:
•
Cisco Unified Communications Manager Express provides more features than standard
SRST (in general).
•
Cisco Unified Communications Manager Express in SRST mode allows a mix of
preconfigured phones and directory numbers for those phones, and directory numbers that
require individual settings and phones that are not configured but are leamed by SNAP.
•
You can combine the advantages of Cisco Unified Communications Manager Express
(more features) and standard SRST (dynamic provisioning of phone configuration during
fallback, using SNAP) by preconfiguring the required individual settings in Cisco Unified
Communications Manager Express.
•
Ephone configuration is based on MAC addresses; ephone-dn configuration is based on the
directory number.
•
All phones or directory numbers that collectively require identical configuration that is not
provided by SNAP do not have to be preconfigured, but the additional configuration can be
applied by templates.
These advantages allow flexible configuration of any Cisco Unified Communications Manager
Express features in a scalable way, since only those phones and directory numbers that require
additional features (or individual settings) have to be manually preconfigured.
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ImplementingCisco Unified Communications Manager, Part 2 (CIPT2)v8.0
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Phone Provisioning Options
The table summarizes the phone provisioning options and shows the relevant configuration
parts.
Phone Provisioning Options
gpftone and
Ephone-dri
ephone 1
Ephoneand
*f„„„„„
ephone-dn 6
ephone 1
mac-address.
mac-address...
Cisco IOS
type 7960
Configuration
button 1:6
type 7960
(no button...)
ephone-dn 6
number 3001
telephony- service
auto assign..
United CME"
Resulting
Existing
Directory Number
United CME'
Configuration
con fig u rat ion
ephone-dn
configuration
referenced by
automatic
assignment
Resulting Phone
Configuration
Ephone.dr, and Complete5RST
SRST Provisioning Provisioning
Existing
Existing
United CME*
united CME'
configuration
ccrifiguration
number 3001
lelephony-seryice
telephory-sarmce
srst mode
srst mode
auto-
auto-
provision. •
provieion...
United CME*
Phone directory
number
ephone-dn
configuration of
matching phone
directory number
plus Unified CME'
united CME'
United CME*
configuration
SRST ephone-dn
template (If used)
ephone template (ft ephone template (if
used)
used)
-Urified CME = Cisco Unified Communications Manager Express
As shown in the table, if an ephone and ephone-dn are configured in Cisco Unified
Communications Manager Express, a phone that registers with the configured MAC address
will get the complete configuration (phone and director;' number) appliedas configured in
Cisco Unified Communications Manager Express. Cisco Unilied Communications Manager
Express does not use SNAP at all to configure the phone.
If an ephone is configured butis notassociated with an ephone-dn, automatic assignment hasto
be enabled. Otherwise, the phone will not have a line and cannotplace or receive calls. The
ephone-dn configuration is determined based on the arguments of theauto-assign command.
SNAP is not used (for learning phone settings or director,' number configuration parameters).
If only ephone-dns are configured, the ephone configuration is learned by SNAP, while the
ephone-dn configuration that is configured in Cisco Unified Communications Manager Express
is used instead of the phone directory number configuration that is provided by SNAP. Ephone
templates (if configured) areapplied to the learned ephone configuration.
If neitheran ephone (MAC address) nor a directory numberexists for the registering phone.
Cisco Unified Communications Manager Express will learneverything(ephone and ephonc-dn
configuration) from SNAP. Ephone and ephone-dn templates are applied, if configured.
i 2010 Cisco Syslems. Inc
Centralized Call-Processing Redundancy Implementation
Note
2-88
This combination is not very common because it combines the need for specific phone
configuration parameters with the dynamic assignment of directory numbers.
•
Manually configured ephone-dn: An ephone-dn is not associated with an ephone. The
reason to configure the ephone-dn but not the ephone is that only individual ephone-dn
configuration is required, but default settings or a single template can be used for the
ephone (uhich is added once the phone is registered).
•
No manual configuration: In this case, the ephone-dn and the ephone are learned by
SNAP. You can apply configuration settings that are not supported by SNAP to such newly
added phones and directory numbers by configuring the appropriate templates.
Implementing Cisco Unrfied Communications Manager, Part 2 (CIPT2) v8.0
© 2010 Cisco Systems. Inc
Configuration of Cisco Unified Communications
Manager Express in SRST Mode
This topic describes how to configure Cisco Unified Communications Manager Express in
SRST mode.
Cisco Unified Communications Manager
Express SRST Configuration Overview
• An SRST reference configured in Cisco Unified
Communications Manager can be a Cisco Unified
Communications Manager Express router.
• Phone configuration is not required for Cisco Unified
Communications Manager Express in SRST mode.
* Any combination of the following can be used:
- Manually configured ephone with associated ephone-dn
Manually configured ephone with no associated ephonedn. and automatic assignment enabled
- Manually configured ephone-dn (not associated with an
ephone)
- No manual configuration
• If configured,ephone and ephone-dn templates are applied
to leamed ephones and ephone-dns
An SRST reference in Cisco Unified Communications Manager can be a standard Cisco
Unified SRSTgateway or a Cisco Unified Communications Manager Express router. Unlike
the standalone Cisco Unified Communications Manager Express, when you are configuring
Cisco Unified Communications ManagerExpress in SRST mode, no phones have to be
configured, since they can be learned by Simple Network-Enabled Auto Provision (SNAP).
Houc\ cr. Cisco Unified Communications Manager Express in SRST mode allows an>
combination of these configurations:
•
Manually configured ephone with associated cphone-dn: In thiscase, thephone is
completely configured: both the ephone and anephone-dn, which is associated with the
ephone. exist, fhis configuration isused for phones that require additional configuration
settings that cannot be learned from the phone via SNAP. These settings should be applied
only tothis phone (or to few phones); an ephone template orephone-dn template cann.il be
used, because these templates apply to all learned phonesor directory numbers.
•
Manually configured ephone with no associated ephone-dn: This configuration is useful
ifspecific phone configuration parameters are required (parameters that cannot be assigned
from a template) but no specific directory number isrequired. Ifan ephone ispreconfigured
in Cisco UnifiedCommunications Manager Express and it is not associated with a
directory number, thedirectory number is notlearned viaSNAP. Therefore, the phone will
not have a directory number unless automatic assignment (which is equivalent to
autoresistralion in Cisco Unified Communications Manager) is enabled.
)2010Cisco Systems, Inc.
Centralized Call-Processing Redundancy Implementation
2-87
Additional Music on Hold Sources—Configuration Example
The figure shows an example of a router that is configured with a default MOH audio source
and two additional MOH audio sources for different departments.
Additional Music on Hold Sources-
Configuration Example
voice moh-group 1
moh Elashimohl.au
description HOH: customer services
multicast moh 239.1.1.1 port 16381
extension-range 1000 Co 1099
extension-range 1300 to 1399
I
voice moh-group 2
moh flashimoh2.au
•r
description HOB: marketing
multicast moh 239.1.1.2 port 16384
extension-range 3000 to 3099
I
telephony - service
moh-file-buffer
5000
moh flashidefault.wav
multicast moh 239.1.1.3 port 163 64
For each department, an MOH group is configured. Within each group, the location of the
MOH audio file and the extensions that should utilize the group have to be configured. In
addition, an optional description can be configured and multicast MOH can be enabled for each
MOH group.
You configure RAM caching under call-manager-fallback (in the case of Cisco Unified SRST)
or under telephony-service (in the case of Cisco Unified Communications Manager Express).
You use the moh-file-buffer size-in-kb command for this configuration, and it specifies the
maximum size of the MOH RAM cache, 'fhe configured limit applies to each audio source file.
You cannot enable or disable audio source caching on a per-file basis. The total amount that is
used for audio source caching, therefore, dependson the numberof configured MOI 1groups. If
all five possible MOH groups and a default audio source are configured, the file buffer size that
is allocated will be six times the specifiedamount. Ifa configured audiosource file is larger
than the configured moh-file-buffer, it will not be cached but will be read from flash instead.
Note
2-86
You can use the show flash command to see the size of the MOH files.
Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0
>201u Cisco Systems, Inc.
*m»
Additional MOH Sources
•Mr
Cisco Unified Communications Manager Express Version 8 introduces the capability of
configuring up to five MOH sources in addition to the default MOH source.
Additional Music on Hold Sources
• Up to five additional MOH sources can be configured.
- Used oniy by SCCP phones that put call on hold.
- RAM caching of MOH file can be enabled.
• Less CPU utilization because of reduced flash reads
• Higher memory consumption
• Supported by Cisco Unified Communications Manager Express,
Cisco Unified Communications Manager Express in SRST mode,
and Cisco Unrfied SRST.
• Supported on Cisco Unified Communications 500 series and Cisco
1800, 2800. 2900. 3800, and 3900 Series Routers,
• Only au or .wavfiles in G 711 8-bit mono format are supported.
Minimum size 100 Kb.
• Default flash dnve should be used to store MOH files.
_*^^_
• Configuration is based on MOH groups.
• Phones not configured to use one of these groups use the default
MOH source.
These additional MOH sources can be utilized by SCCP phones that put calls on hold. Any
other entities thai putcallson hold (such as basicautomatic calldistribution [B-ACD] or SIP
phones) will use the default MOH source. If live audio feed is used, it can beconfigured only
as the default MOH source.
Cisco Unified Communications Manager Express can be configured to cache files in RAM.
This configuration reduces CPU utilization because flash reads areessentially eliminated after
the audio fileshave been loadedto RAM. However, cachingaudio files in RAM can drastically
increase memory consumption. Memory requirements depend onthenumber of MOI I files and
their size (there is no limitation on the maximum size of an audio file).
Multiple MOH sources arc supported by Cisco Unified Communications Manager Express.
Cisco Unified Communications Manager Express in SRSTmode, and Cisco Unified SRST.
Multiple MOH sources arc supported onthese platforms: Cisco Unified Communications 500
Series and Cisco 1800. 2800. 2900, 3800. and 3900 Series Integrated Services Routers.
The audio files have to be .au or .wav files in G.711 8-bit mono format, and their minimum size
is 100 kb. Ifmultiple flash devices are present inthe router, the default flash drive should be
utilized.
The configuration of multiple MOH sources isbased on MOH groups. Endpoints that do not
support MOH groups orthat are not configured to use an MOH group will use the default MOH
source.
Note
The MOH source is selected based on the configuration of the holder (thatis, the phonethat
puts the call on hold).
^^hr
) 2010 Cisco Systems, Inc
Centralized CalI-Processing Redundancy Implementation
2-85
Cisco Unified Communications Manager Express: MOH
MOH is an audio stream that isplayed to PSTN and VoIP callers who are placed on hold by
phones in aCisco Unified Communications Manager Express system. This audio stream is
intended to reassure callers that they are stillconnected to theircalls.
Default Music on Hold Source
The default MOH source is
enabled by the moh command
under telephony service.
For multicast MOH, add the
multicast moh command.
Cisco Unified Communications
Manager Express supports only
G.711 for MOH.
Transcoders are required to allow
G.729 to be used for MOH.
telephony-service
moh moh-file.au
multicast moh
239.1.1.1 port 16384
When thephone thatis receiving MOH is part of a system thatuses a G.729 codec, transcoding
is required between G.711 and G.729. TheG.711 MOH must betranslated to G.729. Note that,
because of compression. MOH that is using G.729 is of significantly lower fidelity than MOII
that is using G.711.
If the MOH audio stream is also identified as a multicast source, the Cisco Unified
Communications ManagerExpress router additionally transmits the streamon the physical IP
interfaces of the Cisco Unified Communications Manager Express router that you specify
during configuration. This transmission permitsexternal devices lo have accessto the router.
Certain IP phones do notsupport IP multicast and, therefore, do notsupport multicast MOI I.
You can disable multicast MOII to individual phones that do not support multicast. Callers hear
a repeating tone when they are placed on hold.
In Cisco Unified Communicalions Manager Express, the MOH feature is supported when a call
is put on hold from a SIP phone and when the user of a SIP phone is put on hold by a SIP,
SCCP. or plain old telephone service (POTS) endpoint. The holder (the party who pressed the
Hold key) or holdee (the party who is put on hold) can be on the same Cisco Unified
Communications Manager Express group or on a different Cisco Unified Communications
Manager Express group that is connected through a SIP trunk. MOH is also supported for Call
Transfer and conferencing, with or without a transcoding device.
Configuring MOH for SIP phones is the same as configuring MOH for SCCP phones.
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Implemenling Cisco Unified Communications Manager, Pan 2 (CIPT2) v8.0
) 2010 Cisco Systems, Inc
-km*
Providing Phone Loads
You can configure Cisco Unified Communications Manager Express to provide specific phone
loads to IP phones for each type of phone.
Providing Phone Loads
Cisco Unified
IP Phones 7945 and
7S65witti SCCP
Copy phone load
Firmware v9.0(2)SR1
files to flash.
Make them
accessible by TFTP.
Specify the phone
load to be used per
phone type under
telephony service.
If no phone load is
specified, the current
phone load of the
phone is used.
tftp-aerver flash; apps4S.9-Q-2ES2.sbn
tftp-server flash; cnu 4 5.9 - 0 - 2 ES2.sbn
tftp-server flash; cmv45sccp.9-0-2ES2.sbn
tftp-server flash: dsp45.9-0-2ES2.sbn
tftp-server flash: jar45sccp.9-0-2BS2.sbn
tftp-server flash: SCCP4 5.9-0-2SR1S.loads
tftp-server flash: term45.default.loads
tftp-aerver flash: termS5.default.loads
I
telephony-service
load 7965
SCCP45. 9-0-2SR1S
load
SCCP45. 9-0-2SR1S
7945
Cisco Unified Communications Manager Express must be configured so that IP phone
firmware files are available through the TFTP server. The command tftp-server flash:
filename allows the specified tile that resides in fiash memory to be downloaded via TFTP.
The figure shows an example with finnuare files that support Cisco Unified IP Phone 7945 and
7965 with SCCP finnuare version 9.0(2)SR1).
Note
You can view a list of IP phone models that are supported by the Cisco Unified
Communications Manager Express router by entering the load ? command in telephonyservice configuration mode
Tip
© 2010 Cisco Systems, Inc.
Firmware filenames are case-sensitive.
Centralized Call-Processing Redundancy Implementation
Cisco Unified Communications Manager Express: Basic
Configuration Example
The figure shows an example of a basic Cisco Unified Communications Manager Express
configuration.
Cisco Unified Communications Manager
Express: Basic Configuration Example
telephony-service
max-e phone s 5
max-dn 10
lp source-address 10.1.250.102 port 2000
create cnf-files
I
Cisco Unified
Communications
Manager Express
ephone-dn
6
number 3001
ephone-dn
7
number 3002
1
ephone
3
mac-address
0012.0154.5D98
type 7 96 0
button
ephone
1:6
4
mac-address
0007.0E5 7.6F43
type 7961
button
1:7
1
dlalplan-pattern 3 5215553... extension-length 4
The exampleshows a Cisco Unified Communications ManagerExpress configuration of two
ephones—one ephone withdirectory number3001 and one ephone withdirectorynumber
3002. The four-digit extensions are expanded to a 10-digit E.164 PSTN address (521 5553xxx).
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Implementing CiscoUnrfied Communicalions Manager, Part 2 (CIPT2) v8.0
J2010 Cisco Systems, Inc.
number: Use this command lo define a directory number for an ephone-dn (extension),
which then can be assigned to an IP phone.
ephone: Use thiscommand in global configuration mode to create a phone in Cisco
Unified Communications Manager Express.
mac-address: Use this command in ephone configuration mode to specify the device ID of
an ephone. When a phone registers with Cisco Unified Communications ManagerExpress,
it has to providea device ID (which is based on the MAC addressof the phone) that is
configured in Cisco Unified Communications ManagerExpress.
type: Use this command in ephone configuration mode lo specify the phone type of this
ephone.
button: Use this command in ephone configuration mode to assign one or moreephor>
dnsto an ephone.
dialpian-pattem: Use thiscommand in telephony-serv ice configuration mode to map
E. 164 PSTN numbers to internal extension numbers.
©2010 Cisco Systems. Inc
Centralized Call-Processing Redundancy Implementation
2-81
General Configuration of Cisco Unified
Communications Manager Express
-#
This topic describes general Cisco Unified Communications Manager Express coniiguration
parameters and their functions.
Basic Cisco Unified Communications
Manager Express Commands
• telephony-service: Enters Cisco Unified Communications Manager
Express configuration mode
- max-ephones: Defines maximum number of IP phones
- max-dn: Defines maximum number of directory numbers
- ip source-address: IP address used by Cisco Unified
Communications Manager Express
- create cnf-files: Generates XML configuration files for phones
• ephone-dn: Enters ephone-dn configuration mode
- number: Sets the extension of the ephone-dn
• ephone: Entersephoneconfiguration mode
- mac-address: Specifies the MAC address of the IP phone
- type: Sets the IP phone model type
- button: Applies an ephone-dn to an IP phone line
• dialplan-pattern: Maps E.164 numbers to extensions
To deploj a Cisco Unified Communications Manager Express system, youmust configure
these commands:'
•
telephony-service: Usethis command to enter the telephony-service configuration mode,
where you can configure the global settings of Cisco Unified Communications Manager
Express
•
max-ephones: Usethis command in telephony-service configuration mode to configure
Cisco Unified Communications Manager Express with the maximum number of ephones.
•
max-dn: Usethis command in telephony-service configuration mode to configure the
maximum numberof extension numbers (ephone-dns) in Cisco Unified Communications
Manager Express.
Note
The default values of max-ephones and max-dn are 0. These defaults have to be modified
inorder foryouto configure ephones and ephone-dns. The maximum numberof supported
ephones and ephone-dns is version-specific and platform-specific. The number that is
displayed in Cisco IPS Software Help files does not always reflect the actual limit.
•
ip source-address: Usethis command in telephony-service configuration mode to define
the IP address to which Cisco Unified Communications Manager Express is bound.
•
create cnf-files: Use this command intelephony-service configuration mode to generate
XML configuration files for phones.
•
2-80
ephone-dn: Use this command, a global configuration command, to create a directory
number. After you enter this command, the router is inephone-dn subconfiguration mode.
Implementing Cisco Unrfied Communications Manager, Part 2 (CIPT2) v8.0
>2010 Cisco Systems, Inc.
Important Cisco Unified Communications Manager Express
Features
Cisco Unilied Communications Manager Express also includes features that are like legacy
low-end PBXand key system features, creatinga cost-effective, highly reliable, feature-rich
communications solution for the small office.
Important Cisco Unified Communications
Manager Express Features
Call Transfer, call transfer blocking, paging, intercom, call
coverage
Call Park, park call recall, dedicated park slot per extension,
MOH, multicast MOH
Hunt groups, basic automatic call distribution (B-ACD)
Ad-hoc conferencing, retain conference call when conference
initiator drops
Night bell, night service call forwarding
Extension mobility, presence
New in Cisco Unified Communications Manager Express 8.0:
Five additional MOH sources
Support for E. 164 numbers and + prefixes
- Enhancement of dialplan pattern command (new demote
argument)
- Voice translation profiles enhancement (new callback
number)
Ihe figure lists important features of Cisco Unified Communications Manager Express.
fhese new features are introduced with Cisco Unified Communications Manager Express
Version 8.0:
•
Kive additional music on hold (MOH) sources: SkinnyClient Control Protocol (SCCP)
phones can beconfigured to use oneof five additional MOII source files thatarc
configured by MOH groups.
•
Support for fc.164 numbers and + prefixes: IP phones can use E.164 format with a +
prefix for their directory numbers.
•
Enhancement of the dialplan pattern command: You can use the dialplan pattern
command to allow internal devices to call each other by an internally used shorter number
that is derived from a longer directory number of the phone {typically in E.164 format with
+ prefix).
•
Enhancement of voice translation profiles: Whena cal! is sent to an IP phone,an
additional number (that is. a callback number) is sent to thephone. The phone shows the
calling-party number on its display but uses the callback number for callbacks from call
lists.
© 2010 Cisco Systems. Inc
Centralized Call-Processing Redundancy Implementation
2-79
Cisco Unified Communications Manager Express
Features
This topic describes Cisco Unified Communications Manager Express versions, their protocol
support and features, and the required Cisco IOS Software releases.
Cisco Unified Communications
Manager Express Features
Delivers capabilities to the small office that were previously
available only to larger enterprises
Reduces the total cost of ownership by delivering voice, video, and
data over a consolidated infrastructure
Runs on Cisco IOS Software; protects customer investment
Supports converged applications
Administered by GUI or CLI
Cisco Unified Communications
Manager Express
Cisco Unified Communications Manager Express delivers to the small or medium-sized
business the capabilities that were previously available only to largerenterprises.
Cisco Unified Communications ManagerExpressintegrates with voice-mail systemssuch as
Cisco Unity, Cisco Unity Connection, Cisco Unity Express, and third-party voice-mail systems.
GUI and command-line interface (CLI) are available for administering Cisco Unified
Communications Manager Express.
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Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0
J2010 Cisco Systems, Inc.
In summary, use Cisco Unified Communications Managerwhen Cisco Unified
Communications Manager Express does not scale to the number of endpoints or does not
provide all therequired features. If youuse Cisco Unified Communications Manager and the
standard Cisco Unified SRST features do not meet the requirements for backup scenarios, you
should use Cisco Unified Communications Manager Express in SRST mode.
©2010 Cisco Systems Inc.
Centralized Call-Processing Redundancy Implementation
2-77
Standalone Unified CME* vs. Unified CM*
and Unified CME in SRST Mode (Cont)
Standalone Cisco Unified Communications Manager Express:
• Small and medium sites (up to 450phones per Cisco Unified
Communications Manager Express).
• Trunks used to interconnect multiple systems provide fewer features.
Cisco Unified Communications Manager with Cisco Unified
Communications Manager Express in SRST mode for backup:
• Medium and large deployments (up to 30,000 phones per Cisco Unified
Communications Managerclusler).
* No limits of features withn a cluster; larger number of phones have access
to all features.
• Cisco Unitied Communications Manager Express in SRST mode used only
as backup for remote sites with centralized cal processing.
• More features in SRST mode than standard Cisco Unified SRST.
"Unified CME = Cisco Unified Co mmuneat ions Manager Express
'"Unified CM = Cisco Unified Communications Manager
The Cisco Unified Communications Manager call-processing solution offers feature-rich
telephony services to medium or largeenterprises. Cisco Unified Communications Manager
Express can serve small deployments on its own or is used as a backup for a centralized callprocessing Cisco Unified Communications Manager deployment (Cisco Unified
Communications Manager Express in SRST mode).
The Cisco Unified Communications Manager Express solution is based on the Cisco access
router and Cisco IOS Software. Cisco Unified Communications Manager Express is simple to
deploy and manage, especially for customers who already use Cisco IOSSoftware products.
This simplicity allows customers to take advantage of the benefits of IP communication without
the higher costs and complexity of deploying a server-based solution. However, the number of
supported phones is relatively lowand depends heavilyon the routerplatform. It ranges from
15 phones at the Cisco 1861 routermodelto as many as 450 on the Cisco 3945E Integrated
Services Router model. Refer to http://www.cisco.com/en/US/prod/collatcral/voicesw/ps6788/
vca!lcon/ps4625/dataj>heet_c78-567246.html for detailedcapacity information per router
platform.
Although multiple Cisco Unified Communications Manager Express systems canbe
interconnected using trunks, the features that are supported across trunks arc limited.
Cisco Unified Communications Manager Express cannot beused if certain features arerequired
to operate across multiple sites. These features include Cisco Unified Communications
Manager Fxtension Mobility and Device Mobility, locations-based Call Admission Control
{CAC) (including Resource Reservation Protocol [RSVPJ-enabled locations), call hunting, Call
Pickup, presence, and many others. In this case, or simplybecauseof the size of the
deployment Cisco Unified Communications Manageris the better choice.
Cisco Unified Communications Manager iscommonly used ascentralized call processing for
somesites. In such an environment, IP phonesregister to a Cisco UnifiedCommunications
Manager across the IP WAN. Inthis case. Cisco Unified Communications Manager Express in
SRST mode is a better choice than standard Cisco Unified SRST functionality, because Cisco
Unified Communications Manager Express in SRST mode offers more features than standard
Cisco Unified SRST.
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Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0
>2010 Cisco Systems, Inc.
Because of the centralized architecture of Cisco Unified Communications Manager, remote site
survivability isextremely important. As discussed earlier. Cisco Unified SRST can be used to
provide survivability. However, it is quite limited in terms of telephony features.
To provide a richer feature that issetto IPphones that are in fallback mode, you can use Cisco
Unified Communications Manager Express in SRST mode. Such a deployment combines the
advantages ofCisco Unified Communications Manager—centralized configuration and the
availability offeatures to all phones, with the better feature support dial isprovided by Cisco
Unified Communications Manager Express versus standard Cisco Unified SRST in casethe site
is disconnected from the centralized Cisco Unilied Communications Manager cluster.
)2010 Cisco Systems, Inc
Centralized Call-Processing Redundancy Implementation
2-75
When to Use Cisco Unified Communications Manager Express
in SRST Mode
To choose between the standalone Cisco Unified Communications Manager Express and Cisco
Unified Communications Manager Express in SRST mode, means to choose between the
features of Cisco Unified Communications Manager Express and the features of Cisco Unified
Communications Manager.
Standalone Unified CME* vs. Unified
CM* and Unified CME in SRST Mode
Unilied CM with Unified CME for SRST
Standalone Cisco'
Unified
Feature
Enterprise size
Clustering
Centralized call
processfig
Features
Normal Operation SRST Mode (United
(Unified CM)
Medium to large
Manager Express
Multiple smal lo
Smal
medium sites
Yes
No
No
Yes
No
Only within local site
All features
AI features
supported by Unite
supported fay Unified si
CM*
Feature limitations
Communications
CME foi SRST)
None
CME*
CME'
Features are
Features are
available only within available only within
local site
local site
•Unified CME = Cisco Unified Communications Manager Express
"Unrfied CM = Cisco Unified Communications Manager
The server-based Cisco Unified Communications Manager telephony solutionprovides
scalability for largeenterprises. Cisco Unified Communications Manager servers can be
groupedin a cluster to provide fault-tolerant telephony for up to 30,000IP phones. Customers
can make use of extensive server-based application programming interfaces (APIs) with Cisco
Unified Communications Manager.
Cisco Unified Communications Manageris a centralized architecture, which allowsendpoint
call controlonly from serverswithin a cluster.Cisco Unified Communicalions Manager
Express is a distributed architecture where eachCiscoUnified Communications Manager
Express router provides call processing to an individual small site or a group of small sites.
Cisco Unified Communications Manager offers a greater choice of voice codecs and video
product selection. Cisco Unified Communications Manager Express doesnot support all Cisco
Unified Communications Manager features.
2-74
Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0
© 2010 Cisco Systems, Inc.
-:-«*
Cisco Unified Communications Manager Express in SRST
Mode
SRSTfallback supportusing Cisco Unified Communications ManagerExpress is a leaturethat
enablesrouters to providecall-processing support for Cisco Unified IP phones if they lose
connection to remote primary, secondary-, or tertiary Cisco Unified Communications Manager
installations or if the WAN connection is down.
Cisco Unified Communications
Manager Express in SRST Mode
Cisco Unified Communications Manager Express in SRST mode
is used only as a backup for Cisco Unified Communications
Manager. In case of WAN failure;
• Cisco Unified Communications Manager Express provides Cisco
CallManager service to IP phones at the remote site,
• Cisco Unified Communications Manager Express controls voice gateway
functions of trie Cisco IOS router at the remote site.
Cisco
Unity
When Cisco Unified SRST functionality is provided by Cisco Unilied Communications
Manager Express, provisioning of phones isautomatic and most Cisco Unified
Communications Manager Express features arcavailable to the phones during periodr of
fallback. The benefit is that Cisco Unified Communications Manager users will gain access to
more features during fallback withoutany additional licensing costs.
) 2010 Cisco Systems, Inc
Centralized Call-Processing RedundancyImplementation
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*9
Cisco Unified Communications Manager Express
Overview
This topic describes Cisco Unified Communications Manager Express and the modes in which
it can be used.
Standalone Cisco Unified
Communications Manager Express
Standalone Cisco Unified Communications Manager Express
operates independently:
• Remote site IP phones register with Cisco Unified Communications Manager
Express.
• Remote site IP phones send signaling messages to Cisco Unified
Communications Manager Express,
• Calls between main site cluster and remote site use standard PSTN
connections or VoIP trunks.
H 323 or SIP
Trunk Connection
Regislrafoon
Cisco Unified
Communications
and Signaling,' J
Manager
The figure shows a deployment of a Cisco Unified Communications Manager Express router
with several phones and devices that are connected to it. The Cisco Unified Communications
Manager Express router is connected to thepublic switched telephone network (PSTN) and
WAN.
Cisco Unified Communications Manager Express is a feature-rich, entry-level IPtelephonysolution that is integrated directly into Cisco IOS Software. Cisco Unified Communications
Manager Express allows small-business customers and autonomous small-enterprise branch
offices to deploy voice, data, andIP telephony on a single platform for smalloffices, which
streamlines operations and reduces network costs.
Cisco UnifiedCommunications ManagerExpress is ideal for customers who have data
connectivity requirements and need a telephony solution in the same office. Whether offered
through a managed serv ices offering of a service provider or purchased directly by a
corporation. Cisco Unified Communications Manager Express provides mostof the core
telephony features thatarerequired in a small office. It also provides many advanced features
that are not available with traditional telephony solutions. Being able todeliver IP telephony
and data routing using a single, converged solution allows customers to optimize their
operations and maintenance costs, resultingin a very cost-effective solution that meetsoffice
needs.
ACisco Unified Communications Manager Express system isextremely flexible because it is
modular. It comprises a router thatserves asa PSTN gateway and supports oneor more
VLANs that connectIP phones and phone devices,as well as PCs,to the router.
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Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
>2010Cisco Systems, Inc
Lesson 3
Implementing Cisco Unified
Communications Manager
Express in SRST Mode
Overview
This lesson begins with a description of key Cisco Unified Communications Manager Express
features, benefits, and limitations. It also explains the processof registering IP phoneswith
Cisco Unified Communications Manager Express, fhis lesson discusses the basic telephonv
sen.ice commands for configuring ephones and ephone-dns.
Objectives
Upon completing this lesson, you will be able to configure Cisco Unified Communications
Manager Express to provide telephony services lo IP phones ifthe connection to the centralized
call agent islost. This ability includes being able tomeet these objectives:
•
Describe Cisco Unified Communications Manager Express and the modes in which it can
be used
•
Describe Cisco Unified Communications Manager Express versions, their protocol support,
their features, and the required Cisco IOS Software releases
•
Describe general Cisco Unified Communications Manager Express configuration
•
parameters and their functions
Describe how to configure Cisco Unified Communications Manager Express to support
SRSI fallback
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
A simplified Cisco Unified SRST dial plan has to be
implemented on the remote site gateways to ensure
connectivity for remote sites in SRST mode.
SRST reference is assigned via device pool membership.
Basic Cisco IOS gateway SRST configuration requires only
six configuration steps.
Cisco IOS gateway MGCPfallback configuration requires
only two configuration steps.
CFUR forwards calls made to unregistered directory numbers
to the defined destination.
ISDN overlap dialing has to be enabled in countries with
open numbering plans.
References
For additional information, refer to these resources:
•
Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0(1), February 2010.
hftp://vvA\w.CMscii.coni/cn/lJS/does/voiee_ip_comm/cucin/admin/8_0_i/ecmefg,'lKem-80Icm.html
• Cisco Systems. Inc. Number Translation Using Voice Translation Profiles, February 2006.
http://\vw u.cisco.com,'en/US/teclL/tk652/lk90/teehnoloaies configuration e\ample()9186a0
0803f818a.shtmi
•
Cisco Systems. Inc. Voice Translation Rules, November 2006.
http://ww\v.cisco.com/cn/[i.S/tcch/tk652/tk90/techiK)logies_tech_note09186a008()325e8e.s
html
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Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0
) 2010 Cisco Systems, Inc.
Cisco Unified SRST Dial Plan Example
(Cont)
dial-peer voice
1 pots
dial-peer voice 9011 pots
corlist outgoing intl
destination-pattern 9011T
port 1/0:23
incoming called-number
direct-in»ard-dial
]
prefix Oil
dial-peer voice 911 pots
destination-pattern 911
I
port 1/3:33
dial-peer voice 2000 pots
translation-profile outgoing to-HQ
1
dial-peer voice
destination-pattern 2...
9 pots
corlist outgoing local-ntl
destination-pattern 9
port 1/0:23
prefix Oil
port 1/0:23
[
voice translation-rule 1
rule 1 /*2/ /151155S2/
1
1
dial-peer voice 91 pota
corlist outgoing local-ntl
destination-pattern 91
voice translation-profile to-HQ
translate
called 1
port 1/0:23
Outgoing COR lists are applied to the outbound dial peers. Note that all dial peers that should
be available to all phones (that is. dial peer 911 for emergency and dial peer 2000 for intersite
calls) are nol configured with an outgoingCOR list.
Local and national PSI'N destinations areprotected by outgoing COR listlocal-ntl. This COR
has one member, pstn-local-ntl. and this member islisted only in the incoming COR list of
Phonc2. not in the incoming COR listof Phone3. Dial peer 9011, which is used for
international calls, is configured with an outgoing COR list intl, and the only member ofthai
list, pstn-intl. is not included in the incoming COR lists ofPhone2 and Phone3.
One dial peer is configured with ihe incoming ealled-numbcr . command, "fhis dial peer is
used as an incoming POTS dial peer, fhe dial peer isconfigured tosupport direct inward
dialing.
The called numbers of inbound PSTN calls (521 555-3xxx) are mapped to four-digit extensions
because ofthe dialplan-pattern command that is configured in call-manager-fallback
configuration mode (see earlier in this subtopic). As aresult, incoming PSTN calls are sent to
the four-digit extensions.
Outgoing calls to phones that are located at the main site (calls to 2...) match the destination
pattern in dial peer 2000. That dial peer sends calls to port 1/0:23 after performing digit
manipulation using the to-HQ voice translation profile, 'fhis profile translates the four-digit
calling number to an Il-digit E.164 PSTN number, which means that during SRST fallback,
users can still dial 4-digit extensions to reach the headquarters.
km*
)20:0 Cisco Systems.
Centralized Call-Processing Redundancy Implementation
2-69
Cisco Unified SRST Dial Plan Example
(Cont.)
application
dial-pe
global
service a l t e r n a t e
mentoet
sr
cor
list
local-ntl
pstn- local n t l
defa u l t
dial-pe
call-nanager- fallback
member
ip source-address 10.1 .250.101
port 2000
man-ephonss 3
r
cor
dial-peer cor
member
list intl
pstn- i n t l
list
phone 2
pfltn- local n t l
mai-dn 3
cor incoming phone! 2
3002
cor incoming phone3 3
3003
dialplan-pattern 1
dial-peer cor l i s t phone 3
member
no-pstn
5215553.,.
extension-length \
disl-peer cor custom
name
no-pstn
name pstn-local-ntl
name
pstn-intl
Thefigure shows the first part of the SRST configuration. Itincludes a dialplan-pattern
command (configured in call-manager-fallback configuration mode) thatmaps the internal
four-digit directory numbers to the E.164 PSTN number.
Based on the scenario, one phone (Phonel) should have unlimited access. No incoming COR
list is required at that phone because, in the absence of an incoming CORlist, all outbound dial
peers are available regardless of a configured outgoing COR listat theoutbound dial peer.
The other two phpnes should have difTerent classes; therefore, anincoming COR listis
configured for each of them (COR lists Phone2 and Phone3). Phone3 should not be allowed to
dial the PSTN at all (exceptfor emergency calls to 911), while Phone2 should not be allowedto
dial international PSTN destinations.
The dial peer that will be used for emergency calls will not be configured with an outgoing
COR list, and hence will beavailable to all callers. Thesame principle applies to all internal
directory numbers. Because they are not configured with anoutgoing COR list, they all are
reachable by everyone. The dial peer for international calls will be protected with outgoing
COR list intl. The member ofthis outgoing COR list (pstn-intl) isnot listed in the incoming
COR list ofeither Phone2 or Phone3. This way, neither ofthese phones can place international
calls. Asmentioned earlier. Phonel does nothave an incoming COR list, and therefore, the
outgoing COR list at the international dial peer isignored for calls from Phone 1, Finally, all
other PSTN dial peers (local and national calls) are protected with outgoing COR list local-ntl.
The incoming COR list ofPhone2 includes the member ofoutgoing COR list local-ntl (pstnlocal-ntl) and therefore can dial local and national PSTN destinations but is not able to dial
internationally. The incoming COR list ofPhone3 includes a member that iscalled no-pstn that
isnot listed in any outgoing COR list. Ilence, this incoming COR list does not provide access
to any protected pattern. Its only use is to change from the default behaviorthat in the absence
ofan incoming COR list all outgoing COR lists are ignored and hence all outbound dial peers
are available. You could also configure COR list Phone3 with no member, and itwould have
the same effect. Ilow ever, it isrecommended that you always include at least one member ner
COR list.
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
>2010 Cisco Systems, Inc.
Cisco Unified SRST Dial Plan Example
The figure shows an example ofa standalone dial plan configuration for a Cisco IOS router that
is enabled with Cisco Unified SRST.
Cisco Unified SRST Dial Plan Example
Requirements for external calls during SRST mode:
• Phonel (3001) should have unlimited access.
• Phone2 (3002) should not be allowedto place international calls.
• Phone3 (3003) should be allowed to dial only headquartersphones
(2 .) and emergency (911)
Remote Site
Mam Site
Cisco
Unified
Phonel
Communications
Manager
Phone2
Phone3
The example shows a headquarters site with a PSI'N number of511 555-2xxx, and a remote
site witha PSTN number of 521 555-3xxx. four digits are used for internal calls(including
calls between the main site and remote site).
There are three phones atthe remote site. During SRS'f fallback. Phonel (using directory
number 3001) should have unlimited access. Phone2 (directory number 3002) should not be
allowed toplace international calls, and Phonc3 (directory number 3003) should be allowed to
place only internal calls. Four-digit dialing to the headquarters should work: the calls should be
sent to the main site over the PSTN.
) 20'0 Cisco Systems, Inc
Centralized Call-Processing RedundancyImplementation
2-67
Combinations of COR Lists and Results
COR List on
COR List on
Incoming Dial Peer
Outgoing Dial Peer
No COR
No COR
Call will succeed.
No COR
COR list that is
Call will succeed. By default, the incoming dial peer
applied for outgoing
has the highest COR priority when no COR is
applied. Ifyou apply no COR for an incoming call
leg to a dial peer, the dial peer can make a call out
of any other dial peer, regardless of the COR
configuration on the outgoing dial peer.
calls
COR list that is applied
for incoming calls
No COR
Result
Call will succeed By default, the outgoing dial peer
has the lowest priority. Because there are some
COR configurations for incoming calls on the
incoming or originating dial peer, it is a superset of
the outgoing-call COR configuration for the
outgoing or terminating dial peer.
COR list that is applied
for incoming calls
(superset of COR list
that is applied for
outgoing calls on the
outgoing dial peer)
COR list that is applied
for incoming calls
(subset of COR list
that is applied for
outgoing calls on the
outgoing dial peer)
2-66
COR list that Is
applied for outgoing
calls (subsets of COR
list that is applied for
incoming calls on the
incoming dial peer)
Call will succeed. The COR list for incoming calls
on the incoming dial peer is a superset of the COR
list for outgoing calls on the outgoing dial peer
COR list that is
Call will not succeed. The COR list for incoming
applied for outgoing
calls (supersets of
calls on the incoming dial peer is not a superset of
COR list that is
peer.
the COR list for outgoing calls on the outgoing dial
applied for incoming
calls on the incoming
dial peer)
Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0
)2010 Cisco Systems, Inc.
Cisco Unified SRST Dial Plan Commands: COR
You can assign calling privileges to IP phones when they are in SRS'f mode by using COR
commands.
Cisco Unified SRST Dial Plan
Commands: Class of Restriction
router(config-cm-fallback)#
cor {incoming I outgoing} cor-list-name [cor-list-number
starting-number - ending-number | default]
• Configures a COR on dial peers associated with directory
numbers (ephone-dn)
The command cor configures a COR ondial peers that are associated with directory numbers.
• The keyword, incoming specifies that the COR list is tobe used by incoming dial peers.
• The keyword outgoing specifies that the COR list is tobe used by outgoing dial pers.
•
The parameter cor-list-name is the COR list name.
• The parameter cor-list-number is a COR list identifier. The maximum number ofCOR lists
thatcan be created is 20.and the listsconsist of incoming or outgoing dial peers. The first
six COR lists are applied to a range ofdirectory numbers, 'fhedirectory numbers that do
not ha\e a COR configuration arc assigned lo the default COR list, as long asa default
COR list has been defined.
• The parameters starting-number - ending-number define the director- number range—for
example. 2000 to 2025.
•
fhe ke\ word default instructs the routerto use an existingdefault COR list.
fhe complete configuration ofCOR is discussed in the Implementing Cisco Voice
Communications and QoS (CVOICF) course. The table presents an overview only.
i 2010 Cisco Systems. Inc
Centralized CalI-Processing RedundancyImplementation
2-65
Cisco Unified SRST Dial Plan Commands: Number Modification
(Profile Activation)
Voice translation profiles can be bound to dial peers, source groups, trunk groups, voice ports,
and the voice service POTS.
SRST Dial Plan Commands: Number
Modification (Profile Activation)
router(config-volcaport)#
translation-profile {incoming | outgoing} name
* Assigns a translation profile to a voice port
router(config-cm-fallback>#
I translation-profile {incoming | outgoing} name
' Assigns a translation profile to the call-manager-fallback
Cisco IOS service
In this example, the voice translation profile is bound to a voice port.
To assign a translation profileto a voice port, you use the translation-profile command in
voice-port configuration mode.
•
The keyword incoming specifies that this translation profileprocessesincoming calls.
•
The keywordoutgoing specifies that this translation profileprocessesoutgoingcalls.
•
The parameter name is the name of the translation profile.
The voice translation profiles can also be bound to call-manager-fallback Cisco IOS service.
The structure of the command is identical.
Note
The incoming direction ofthe voice translation profile that is boundto the CiscoCallManager
fallback Cisco IOS service processes the calls comingfrom IP phones that are registered
with the router.
For more information about voice translation profiles, refer to Cisco TechNotes Number
Translation Using I oice Translation Profilesat
http:/^vww,cisco.coni/en/[;S/tech/tk652/tk90/technologies^ci)nfiguration_examplc()9l86a()08().5
fS18a.shtml and TechNotes Voice Translation Rules at
http://wwu.cisco.com/en/L;S/tech/tk652/lk90/iechnologies_tech..nole09186a0080325e8c.shlm!.
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Cisco Unified SRST Dial Plan Commands: Number Modification
(Voice Translation Rules)
In voice translation rules, sets of number modification rules are defined.
SRST Diaf Plan Commands: Number
Modification (Voice Translation Rules)
router (confi gl #
voice-translation-rule
number
• Defines a translation rule for voice calls
router(cfq-translation-rulel#
rule precedence /match-pattern/ /replace-pattern/
[type {match-type replace-type} [plan {match-type replacetype}]]
* Defines a translation rule
To define a translation rule for voice calls, use the voice-translalion-rulc command in global
configuration mode.
•
Number: The number that identifies the translation rule. The range is from 1 lo
2147483647.
To define a translation rule, use the rule command in voice translation-rule configuration
mode.
•
The parameter precedence defines thepriority of the translation rule, fhe range is from I to
•
The parameter Imatch-patternl is a stream editor (SED) expression thatis used to match
incoming call information, fhe slash (/) is a delimiter in thepattern.
15.
• The parameter Ireplace-patternl isa SED expression that isused toreplace the match
pattern in the callinfonnation. Theslash is a delimiter in the pattern.
• The optional construct type match-type replace-type allows for modification ofthe number
type ofthe call. Valid values for the match-type argument are these: abbreviated, any,
international, national, network, reserved, subscriber, unknown. Valid values for the
replace-type argument are abbreviated, international, national, network, reserved,
subscriber, unknown.
• The optional construct plan match-type replace-type allows for modification ofthe
numbering plan of the call. Valid values for the match-type argument are any, data, ermes.
isdn, national, private, resen-ed, telex, unknown. Valid values for the replace-type
argument are data, ermes. isdn. national, private, reserved, telex, unknown.
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Cisco Unified SRST Dial Plan Commands: Number Modification
(Voice Translation Profiles)
The combination of voice translation profiles and voice translation rules creates a powerful tool
for modifying numbers so that they match dial plan needs.
SRST Dial Plan Commands: Number
Modification (Voice Translation Profiles)
router(config)#
I voice-translation-profile name
Defines a translation profile for voice calls
router(cfq-translation-profile)#
translate {called | calling | redirect-called | redirecttarget | callback} translation-rule-number
Associates a translation rule with a voice translation profile
To define a translation profile for voice calls, you use the voice-translation-profile command
in global configuration mode.
The parametername definesthe name of the translation profile. The maximum lengthof the
voice translation profile name is 31 alphanumeric characters.
To associate a translation rule with a voice translation profile, you use the translate command
in voice translation-profile configuration mode:
•
called: Associates the translation rule with called numbers
•
calling: Associates the translation rule with calling numbers
•
redirect-called: Associates the translation rule with redirected called numbers
•
redirect-target: Associates the translation rule with transfer-to numbers and callforwarding final destination numbers
•
callback: Associates the translation rule with the numberto be used by IP phones for
callbacks
Note
While ona call, IPphones display the calling-party number. When callbacks are placed from
call lists, the callback number {if present) is utilized for the outbound call, and notthe callingparty number that was shown while the call was active.
•
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translation-rule-number: The number of the translation rule to use for the call translation.
The valid range is from 1 to 2147483647. There is no default value.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) u8.0
>2010 Cisco Systems, Inc.
•
The keyword extension-length sets the number of extension digits that will appear as a
caller ID followed by the parameter length, which is the numberof extension digits. The
extension length must match the setting for IP phones in Cisco Unified Communications
Manager mode, 'fhe range is from I to 32.
•
The optional keyword evtension-pattern sets the leadingdigit patternof an extension
number when the patternis difTerent from the leadingdigits that are defined in the pat.jm
variableof the F.164 telephone number, such as when site codesare used. The parameter
extension-pattern that follows defines the leading digit patternof the extension number. It
comprises one or more digits and wildcard markers or dots (.). For example, 5.. would
include extensions 500 to 599: and 5... would include extensions 5000 to 5999. The
extension pattern configuration should matchthe mappingof internal to external numbers
in Cisco Unified Communications Manager.
•
Theoptional keyword no-reg prevents the I7..164 numbers in the dial peerfrom registering
with the gatekeeper.
fhe example for the dialplan-pattcrn command shows how lo create a dial plan pattern for
directory numbers 500 lo 599that is mapped lo a DID range of 408 555-5000 to 5099. Ifthe
router receives an inbound call to 408 555-5044, then the dial plan pattern command is matched
and the extension of the called I7,. 164 number. 408 555-5000, is changed lo directory number
544. If an outbound calling-party extension number (544) matches thedial plan pattern, the
calling-part) extension will beconverted to the appropriate L.I64 number (408 555-5044). The
F..I64 calling-party number will appear as the caller ID:
Router (config)# call-manager-fallback
Router(config-cm-fallback)# dialplan-pattern 1 40855550..
extension-length 3 extension-pattern 5..
Since Cisco Unified SRST 8.0.the dialplan-pattcrn command has been used in the opposite
way with the addition of the keyword demote tothe end ofthe command. In this case it
demotes IF phone directory numbers thatarespecified in F.I64 format with a + prefix to
shorter extensions, which are to be used internally. Extemal callers place calls to the phones,
using E.164 format with a-f prefix. Ifthe calls are not natively received in this format from the
PSTN (which they rarely are), you have to transform the called number accordingly. Internal
users.howe\er. can dial each other by usingshorterextensions, which arc set up by the
dialplan-pattern command with the demote argument:
Router(config)# call-manager-fallback
Router(config-cm-fallback)# dialplan-pattern 1 +415526....
extension-length 5 demote
In this example, phones are configured with directory numbers+415526.... and have tobe
called that way from the outside. Internal users, however, can call each other by using the last
five digits (6....).
Note
Thedialplan pattern command with thedemote argument isalso available in Cisco Unified
Communications Manager Expressand hence can also be used forCisco Unified SRST
when Cisco Unified Communications Manager Express is used in SRST mode.
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Cisco Unified SRST Dial Plan Commands: Open Numbering
Plans
To activate overlap receiving on ISDN interfaces, you use the isdn overlap-receiving
command in interface configuration mode.
Cisco Unified SRST Dial Plan
Commands: Open Numbering Plans
router(conflq-if)#
isdn overlap-receiving [T302 ms]
• Enables overlap receiving on ISDN interfaces
router(conflq-cm-fallback)#
timeouts interdigit aec
• Configures the interdigit timeout value for all Cisco IP phones
attached to a router
router(config-cm-fallback)#
dialplan-pattern tag pattern extension-length length
[extension-pattern extension-pattern]
[no-reg]
[demote]
• Creates a global prefix that can be used to expand or demote the
intemally used directory numbers
The isdn overlap-receiving command is applicable on BRl interfaces or on the ISDN interface
ofTl/El controllers in PRI mode.
The optional parameter T302 defines the number of milliseconds that the T302 timer should
wait beforeexpiring. Validvalues for the milliseconds argumentrange from 500 to 20000.The
default value is 10000 (lOseconds).
Caution
Modification of the T302parameter, whenconnectedto public networks, might disablethe
function.
To configure the timeout value to waitbetween dialed digits for all Cisco IP phones that are
attached to a router, usethe timeouts interdigit command in call-manager-fallback
configuration mode.
•
The parameter sec defines the interdigit timeout duration, in seconds, for all Cisco IP
phones. Valid entries are integers from 2 to 120.
To create a global prefix that can be used to expand the extension numbers of inbound and
outbound calls into fully qualified E.164 numbers, youuse thedialplan-pattern command in
call-manager-fallback configuration mode.
•
The parameter tag is the unique identifier that is used before the telephone number. The tag
number is any number from I to 5.
•
The parameter pattern is the dial plan pattern, such as the area code, theprefix, and the first
oneor twodigitsof theextension number, plus wildcard markers or dots (.) for the
remainder of the extension number digits.
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dial-peer voice 5 pots
description PSTN-emergency
destination-pattern 911
port 0/1/0
forward-digits all
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Cisco Unified SRST Dial Plan
Commands: Dial Peer (Cont)
This table contains common classes ofPSTN calls in the NANP
and lists the pattern that is used for each class. An access code
*f
of 9 should be used to indicate a PSTN call.These patterns
have to be reachable in SRST mode:
Call Type
Emergency
911
Services
[2-8)11
Local
[2~9]xx-xxxx
Long distance or national
1[2-9]xx [2-9]xx-xxxx
International
011+countrycode+number
Toll-free
1[800,866,877,8 88]xxx-xxxx
1 900xxx-xxxx
Premium
976-xxxx
The table represents the most common classes ofPSTN calls in the Nortli American
Numbering Plan (NANP) and lists the pattern that is used for each class.
An access codeof 9 should be usedto indicate a PSTN call.
The patterns in the table are the minimum patterns that must be reachable in SRST mode. This
example lists the configuration of dial peers that would be needed to reach all the numbers that
are indicated.
dial-peer voice 1 pots
description PSTN-LD
destination-pattern 91 [2-9] .. [2-9]
port 0/1/0
forward-digits 11
dial-peer voice 2 pots
description PSTN-local
destination-pattern 9 [2-9]
port 0/1/0
dial-peer voice 3 pots
description PSTN-international
destination-pattern 9011T
port 0/1/0
prefix 011
i
dial-peer voice 4 pots
description PSTN-emergency
destination-pattern 9911
port 0/1/0
forward-digits 3
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>2010 Cisco Systems, Inc.
*^**
Note
Details about the meanings of these special characters and about Cisco IOS dial peer
configuration in general are provided in theImplementing Cisco Voice Communications and
Ju
QoS (CVOICE) course.
• The optional control character T indicates that the dcslination-pattcrn value is avariablelength dial string. Using this control character enables the router to wait until all digits arc
received before routing the call.
To associate adial peer with aspecific voice port, use the port command in dial peer
configuration mode.
•
The parameter slot-number defines the number of iheslot in the router inwhich the voice
interface card (VIC) is installed. Valid entries depend onthe number ofslots that the router
platform has.
•
The parameter/jo/7 defines the voice port number. Validentries are 0 and I.
mm
m»
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Cisco Unified SRST Dial Plan Commands: Dial Peer
The dial-peer command is the main component for configuring dial plans on Cisco IOS
routers.
Cisco Unified SRST Dial Plan
Commands: Dial Peer
router(config)M
| dial-peer voice tag [pots | voipj
- Defines the dial peer
router(config-dial-peer) #
|destination-pattern [+]Btring[T]
Specifies either the prefix or the complete E.164 telephone
3
numbertobeused tbradial peer
router(conflg-dlal-peer)B
I port slot-number/port
• Associates a dial peer with a specific voice port
To define a particular dial peer, specify a voice encapsulation method, and enter dial peer
configuration mode, you use the dial-peer voice command in global configuration mode.
• The parameter tag specifies digits that define a particular dial peer. The range is from 1to
2147483647.
• The keyword pots indicates that this peer isa plain old telephone service (POTS) peer;
voip indicates that this peer is a VoIPpeer.
To specif; either the prelix orthe complete E. 164 telephone number tobe used for a dial peer,
you use the destination-pattern command in dial peer configuration mode.
•
The optional character+ indicates that an E.164 standard numberfollows.
• The parameter string defines aseries ofdigits that specify apattern for the E. 164 orprivate
dialing plan telephone number. Valid entries are the digits 0 through 9, the letters A
through D. and these special characters:
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>2010 Cisco Systems, Inc.
Additional Cisco Unified SRST Dial Plar
Requirements (Cont.)
• Preservation of calling privileges
Cisco Unified Communications Manager CoS should be
taken over into SRST dial plan.
- Requires appropriate COR configuration at SRST
gateway.
» Preservation ofdialing behavioron variable-length numbers
Tuning of interdigit timeout,
Use of # key to terminate dialing.
Use of overlap sending if supported by PSTN.
Ifthe calling privileges (which in normal mode are controlled by Cisco Unified
Communications Manager) have tobe preserved in SRS'f mode, class ofrestriction (COR)
configuration has to be used.
fhe handling ofvariable-length numbers should also be preserved in SRST mode, fhis
includes tuning ofthe interdigit timeout, the possibility to use the # key toterminate dialing:
and the implementation of overlapsending.
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Additional SRST Dial Plan Requirements
The goal of Cisco Unified SRS'f dial plan design should be to reach the largest degree of
analogy between usability in normal mode and in SRST mode. The telephony service should
have the same design for the user, regardless of the mode that the system is in.
Additional Cisco Unified SRST Dial Plan
Requirements
Preservation of calling-party number presentation on inbound
PSTN calls
-
Includes international and national access codes
-
Includes PSTN access codes to enable one-click callback
from lists
Preservation of on-net dial plan
- Internally used numbers for remote sites have to be
transformed to their PSTN format.
- Applies to calling-party numbers of inbound calls and
called-party numbers of outbound calls.
Ideally, the numbers in call lists (such as missed calls) have the correct format (PSI'N access
code plus PSTN phone number) that is required for callback so that users do not have to edit
the number manually. In this case, the calling party ID of incoming calls from the PSTN needs
to be modified by voice translation profiles and voice translation rules.
Abbreviated dialing between sites of the site code plus the extension number is possible in
SRST mode. Voice translation profiles have to be used to expand the called numbers to PSTN
format for intersite dialing.
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>2010 Cisco Systems, Inc.
Cisco IOS Gateway MGCP Fallback and Cisco
Unified SRST Dial Plan Configuration
This topic describes the minimum dial pian configuration steps that are needed for
communication between phones in SRST mode and the PSTN.
Minimum SRST Dial Plan to Enabl
Calls Between PSTN and Remote
Dial peer configuration must have destination patterns that
correspond to the PSTN access code.
ISDN overlap receiving has to be enabled if PSTN uses
overlap sending.
Transformation between internally used directory numbers
and externally used PSTN numbers has to be configured.
Dial plan pattern command allows automatic
transformation.
Voice translation profiles can be used when more features
and granularity are required.
• Use of ISDN type of number
• Use of different internal and external DID ranges
The minimum requirement for a dial plan in SRST mode is that it must enable the remote site
users to place and receive calls from the PSTN.
At least one dial peer must be configured lo enable calls to the PSTN. The destination pattern
of that dial peer has to correspond to the PS IN access code (for example, 9T). 'fhe more
elegant wa\ is to configure several dedicated dial peers with destination patterns that match the
number patterns in a closed numbering plan, such as 91
(91 followed by 10 dots).
In countries that have open numbering plans, the only destination pattern that is needed is 9T.
Because of the variablelength of dialednumbers, the router is waitingfor the interdigittimeout
(1302) or for a hash (#) sign to indicate the end of the dial string.Cisco Unified SRSTversion
4.1 and Cisco Unified Communications Manager F.xpress Version 4.1 do not support the
overlap sending feature to the PSI'N. fhe receiving of ISDN overlap dialing from PSTN is
supported but has lo be enabledon the interlaces. To shorten the wail time for users afterthe;
complete the dial string, it is possible to reduce the inlerdigit timeout from thedefault of 15
seconds.
Dial plan pattern configuration is a powerful tool forthe modification of incoming called
numbers to match remote site extensions.
© 2010 Cisco Systems. Inc.
Centralized Call-Processing Redundancy Implementation
2-!
If different sites requiredifferent dialing patterns (for example, an international deployment
where each country has different PSTN access codes and international access codes), it is
recommended that you specify the PSTN number to be used for CFUR in H.I64 format with a
+ prefix. The CFUR CSS should match a \+! router pattern and refer to a route list, which is
configured to use the Standard Local RouteGroupof the callingdevice. At the egress gateway,
after path selection has been performed using the local route group, the called number can be
modified by global transformations (via a called-party transformation CSS configured at the
egress gateway), based on the individual requirements of the selected egress gateway.
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems, Inc
Step 3: Configuring CFUR
"fheCFUR feature is a way to reroutecalls placed to a temporarily unregistered destination
phone. Theconfiguration of CFUR consists of twomain elements: destination selection and
CSS.
Step 3: Configuring CFUR
Setting CFUR on phone lines:
• Can pointloaPSTNdesttnation
• Can point to \oice mail
Cisco Unified Communications ManagerAdministration > Call Routing :
Directory Number
When the directory number isunregistered, calls can be rerouted to the voice mail that is
associated with theextension or to a directory number that is used to reach the phone through
the PSTN. The latter approach ispreferable when a phone islocated within a site whose VAN
link is down. Ifthe site is equipped with SRST. thephone (and itseolocated PS'fN gateway)
will reregister with the eolocated SRST router. The phone isthen able to receive calls placed to
its PSI'N direct inward dialing (DID) number.
In this case, the appropriate CFUR destination isthe corresponding PSTN DID number ofthe
original destination directory number. Configure this PSTN DID in the destination field, along
with applicable access codes and prefixes (for example. 9 1415 555-1234).
Cisco Unified Communications Manager attempts toroute the call tothe configured destination
number b\ using the CFUR CSS ofthe called directory number. The CFUR CSS is configured
on the target phone and is used by all devices that are calling the unregistered phone.
As a result, all calling devices will use the same route pattern, route list, and route group to
place the call. Ifaspecific route group is defined in the route list, all CFUR calls to agiven
unregistered device will be routed through the same unique gateway, regardless ofthe location
ofthe calling phone. In this case, it is often recommended that you select a centralized gateway
as the egress point to the PSTN for CFUR calls and configure the CFUR CSS to route calls to
the CFUR destination through this centralized gateway.
Abetter solution istouse the local route group feature. When you use this feature, the route list
does not refer to a specific route group, but "Standard Local Route Group" is added to the route
list instead, fhe route group that isto be used for the calls is then determined by the local route
group that is configured at the device pool ofthe calling device. In this case, phones at different
sitescan refer to different route groups viatheirdevice pool configuration.
>2010 Cisco Systems Inc
Centralized Call-Processing RedundancyImplementation
2-51
Step 2: Setting Max Forward Unregistered Hops to DN
The Cisco CallManager service parameter Max Forward Unregistered Hops to DN in Cisco
Unified Communications Manager reduces the impact thaiis caused by CFUR routing loops.
Step 2: Setting Max Forward
Unregistered Hops to DN
Max Forward Unregistered Hops to DN service parameter:
• Default is 0 (counter disabled).
• A reasonable value is 2.
Cisco Unified Communications ManagerAdministration > System >
Service Parameter > Cisco CallManager
; r,-.^l!J-i-E'™'
,;
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.-,*«• >-V*i-« >*lrT W.I-.- r.V
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This parameter specifics the maximum number offorward unregistered hops that are allowed
for a director, numberat one time. It limitsthe numberof times that the call can be forwarded
because of the unregistered directory number when a forwarding loop occurs. Use this count to
stop forward loops for external calls that have been forwarded by CFUR, such asintercluster IP
phone calls and IP phonc-to-PSTN phone calls that are forwarded toeach other. Cisco Unified
Communications Manager terminates thecall when the value that is specified in thisparameter
is exceeded, fhe default 0 disables the counter but not the CFUR feature. The allowed range is
from 0 to 60.
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Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc
Cisco Unified Communications Manager Dial
Plan Configuration for SRST Support
This topic describes theconfiguration that is necessary for adjusting the Cisco Unified
Communications Manager dial plan to work with Cisco Unilied SRST.
SRST Dial Plan Configuration Procedure in
Cisco Unified Communications
Define a CSS for CFUR.
Set the Max Forward Unregistered Hops to DN parameter.
Configure CFUR at directory numbers of remote phones.
Step 1: Define a CSS for CFUR
You must adjust the Cisco Unified Communications Manager dial plan to ensure the
reachability of remote site phones by their extensions even if the remote site runs in SRS'f
mode. The parameter that enables this adjustment is the CFUR destination setting, which has to
be defined on even' line of an SRS'f-enabled remote site phone. This parameter was introduced
in Cisco Unified Communications Manager Version 4.2.
I he CFUR feature forwards calls to unregistered (disconnected or logged out) director;'
numbers for the defined destination, "fhe destination might be the PSTN number of a phone at a
remote site or the voice mail for a user in a Cisco Unified Communications Manager Extension
Mobility setting.
Cisco Unified Communications Manager attempts to route the call to the configured destination
number by using the CFUR calling search space (CSS) of the directory number that was called.
CFUR causes routing loops whenever there is a single disconnected SRST phone in which the
remote location is not in SRST mode. Internal calls to that directory number will be forwarded
lo the CFUR (PSTN) destination and will be received by the remote site gateway in normal
mode. This gateway will process the call as usual, sending the signaling lo its Cisco Unified
Communications Manager subscriber. Then Cisco Unified Communications Manager will
again fonvard the call to the PSTN, causing an inevitable routing loop.
To limit the impact of these routing loops, Cisco introduced a Cisco CallManager service
parameter: Max Fonvard Unregistered I lops to DN. When activated, this counter limits the
calls thai arc fon\arded lo one CFUR destination.
©2010 Cisco Systems, Inc.
Centralized Call-Processing Redundancy Implementation
MGCP Fallback Configuration Example
The figure shows an example of an MGCP gateway fallback configuration for a Cisco IOS
router that is enabled with Cisco Unified SRST.
MGCP Fallback Configuration Example
Main Site
Remote Site
SRST# configure terminal
SRST(config)# ccm-manager fallback-mgcp
SRST(config)# application
SRST(config-app)# global
SRSTfconfig-app-global)# service alternate Default
SRSTIconfig-app-global)# end
SRST#
TheCisco Unified SRST router is installed at a small branch officesitewith three IP phones.
Here is the configuration that is necessary forthe Cisco Unified SRST routerto perform the
MGCP gateway fallback in this environment:
SRST# configure terminal
SRST(config)#ccm-manager fallback-mgcp
SRST(config)#application
SRST(config-app)#global
SRST(config-app-global)#service alternate Default
SRST(config-app-global)#end
SRST#
Note
2-48
More commaindsmight
be necessary, depending on the complexity of the deployment.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
) 2010 Cisco Systems, Inc
Steps 1 and 2: Enabling MGCP Fallback and Setting Fallback
Service
The Cisco IOS command ccm-manager fallback-mgcp enables the gateway fallback feature
and allows an MGCP voice gateway to provide call-processing services through Cisco Unified
SRST or other configured applications when Cisco Unified Communications Manager is
unavailable.
Steps 1 and 2: Enabling MGCP Fallback
and Setting Fallback Service
router (config)#
ccm-manager fallback-mgcp
Enables the MGCP gateway fallback feature.
router (config) #
call application alternate Default
or
router Iconfig-app-global)W
service alternate Default
Specifies that the voice application named Defaulttakes over
ifthe MGCP call agent is not available; this action allows a
fallback to H.323 or SIP. Enter the service alternate
command in global application configuration mode.
Use either of these commands, depending on Cisco IOS
Software release.
The call application alternate Default command specifies that the default voice application
takes o\er ifthe MGCP call agent is not available, fhiscommand allows a fallback to H.323 or
SIP. which means that local dial peers are considered for call routing.
You enter the sen ice alternate Default command in the global application configuration
mode. To na\igate to this location, perfonn these steps:
Step 1
Step 2
To enterapplication configuration mode to configure applications, use the
application command in global configuration mode.
To enter global application configuration mode, use the global command in
application configuration mode.
Enter eitherof the two commands, depending on the Cisco IOSSoftware release. The newer
configuration method is the servicecommand.
As discussed in the previous lesson, analog calls are preserved in the event of MGCP fallback.
In order toprovide call preservation during switchback, you must enable call preservation for
H.323 using the following commands:
voice service voip
h323
no h225 timeout keepalive
© 2010 Cisco Syslems. Inc.
Centralized Call-Processing Redundancy Implementation
Cisco IOS Gateway MGCP Gateway Fallback
Configuration
This topic describes the configuration steps that are needed to activate the MGCP gateway
fallback feature on a Cisco IOS router.
MGCP Gateway Fallback Configuration
Procedure in Cisco IOS Router
1
Activate MGCP gateway fallback.
2
Define the service to fall back to.
Configuration of the MGCP gateway fallback on a Cisco IOS router to support the MGCP
fallback function requires these two steps:
•
Activation of MGCP gateway fallback
•
Definition of the service to fall back to
Toenable outbound calls while in SRST mode on an MGCP gateway, youmust configure two
fallback commands on the MGCP gateway. These two commands allow SRST to assume
control overthevoice portandovercall processing on the MGCP gateway. With CiscoIOS
Software releases earlier than 12.3(14)T, configuration of MGCP gateway fallback requires the
use of the ccm-manager fallback-mgcp andcall application alternatecommands. With
Cisco IOS Software releases later than I2.3(I4)T, configuration of MGCP gateway fallback
requires the use of the ccm-manager fallback-mgcp and service commands.
Note
Both commands have to beconfigured. Configurations will not be reliable if only theccmmanager fallback-mgcp command is configured.
To use CiscoUnified SRST on an MGCP gateway, you mustconfigure CiscoUnified SRST
and MGCP gateway fallback on the same gateway.
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>20IOCisco Systems, Inc.
Cisco Unified SRST Configuration Example
fhe figure shows anexample of a Cisco Unified SRST configuration to support SCCPcontrolled IP phones.
Cisco Unified SRST Configuration Example
Remote Site
Main Site
SRST# configure terminal
SRST(config)# call-manager-fallback
SRST(config-cm-fallback)# ip source-address 172.47.2.1 port 2000
SRST(config-an-fallback! tt max-ephones 3 dual-line
SRST(config-cm-fallback]# max-dn 6
SRST(config-cm-fallback)# limit-dn 7960 2
SRST(config-cm-fallback)# keepalive 20
SRSTlconfig-cm-fallback)# end
SRST#
"fhe SRST router is installed at a small branch office site with three IP phones, each having two
lines (six lines intotal). The IP address 172.47.2.1 isconfigured onthe Fthemct interface where
the IP phones are connected. Here isthe configuration thai you must perform for the Cisco
Unified SRST router to operate in this environment:
SRSTtt configure terminal
SRST(config)# call-manager-fallback
SRST(config-cm-fallback)# ip source-address 172.47.2.1 port
2C00
SRST(config-cm-fallback)# max-ephones 3 dual-line
SRST(config-cm-fallback)# max-dn 6
SRST(config-cm-fallback)# limit-dn 7960 2
SRST(config-cm-fallback)# keepalive 20
SRST(config-cm-fallback)# end
SRST#
Note
) 20'0 Cisco Systems. Inc
More commands might be necessary, depending on the complexity ofthe deployment
Centralized CalI-Processing RedundancyImplementation
2-46
Steps 5 and 6: Setting Maximum Directory Numbers Per Phone
and Keepalive Timer
To optimize perfonnance of the system, it is recommended that you use the two commands that
are shown in the figure.
Steps 5 and 6: Setting Maximum Directory
Numbers Per Phone and Keepalive Timer
router(config-cm-fallback)#
|limic-dn (7910 / 7335 / 7940 / 7960} max-lines
* Limitsthe directory number lines on Cisco IP phones during
SRST mode.
* Default is 6-line maximum.
router(config-cm-fallback} #
Ikeepalive oecondg
* Sets the time interval, in seconds, between keepalive
messages that are sent to the router by Cisco IP phones.
• Default is 30.
The optional Cisco IOS command limit-dn limitsthe directory number lineson Cisco IP
phones during SRST mode, depending on device types.
Note
This command mustbe configured during initial Cisco Unified SRST router configuration,
before any phone actually registers with the Cisco Unified SRST router. However, the
number of lines can be modified at a later time.
The setting for maximum lines is from 1to 6.The default number of maximum directory lines
is setto 6. Ifthere is any active phone with the lastline number greater than this limit, warning
infonnation is displayed for phone reset.
The optional Cisco IOS command keepalive sets the time interval, in seconds, between
keepalive messages thataresent to therouter by Cisco IPphones. Therange is 10 to 65535.
Default is 30.
The keepalive interval is the lime between keepalive messages that aresentby a network
device.
The default keepalive timer is 30 seconds.
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Implementing Cisco Unified Communicalions Manager, Part2 {CIPT2) v8.0
12010 Cisco Systems, Inc.
Steps 3 and 4: Setting Maximum Directory Numbers and
Telephones
"fhe ne\t two commands, max-dn and max-ephone. are mandatory because the default value?
for both commands are defined as 0.
Steps 3 and 4: Setting Maximum
Directory Numbers and Telephones
rouCer(conflg-cm-fallback)W
max-dn max-directory-numbers [dual-line]
preference-order]
[preference
• Sets the maximum number of directory numbers or virtual
voice ports that can be supported by the Cisco Unified SRST
router and activates the dual-line mode
router (config-cm-fallback) #
__^_
Imax-ephones max-phones
Configures the maximum number ofCisco IP phones that can
be supported by the Cisco Unified SRST router
The Cisco IOS command max-dn sets the maximum numberof directory numbers or virtual
voice ports that can be supported by the router, and activates the dual-line mode. The maximum
numberis platform-dependent. The default is 0.
The dual-line keyword isoptional. Itallows IP phones in SRS'f mode tohave a virtual voice
port with two channels.
Note
Thedual-line keyword facilitates call waiting. Call Transfer, and conference functions by
allowing two calls tooccur onone line simultaneously. In dual-line mode, all IP phones on
the Cisco Unified SRST router support two channels per virtual voice port.
The optional parameter preference sets the global preference for creating the dial peers for all
director) numbers that are associated with the primary number. The range is from 0to 10. The
default is 0. which is the highest preference.
Note
The router must berebooted in order to reduce thelimit ofthedirectory numbers orvirtual
voice ports after the maximum allowable number is configured,
To configure the maximum number ofCisco IP phones that can be supported by aCisco
Unified SRS'f router, use the max-ephones command incall-manager-fallback configuration
mode. The default is 0. and the maximum configurable number is platform-dependent.
Note
The router must be rebooted in order to reduce thelimit ofCisco IP phones after the
maximum allowable number is configured.
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Centralized Call-Processing RedundancyImplementation
2-43
Steps 1 and 2: Enabling Cisco Unified SRST and Setting Cisco
Unified SRST IP Address
The figure describes the commands for the firsttwo Cisco Unified SRST configuration steps.
Steps 1 and 2: Enabling SRST and
Setting Cisco Unified SRST iP Address
router (confiq)#
call-manager-fallback
• Enables the Cisco Unified SRST feature and enters Cisco
call-manager-fallback configuration mode.
router (config-cm-fallback)#
ip source-address ip-address [port port]
[any-match
strict-match]
• Enables the router to receive messages from the Cisco IP
phones through the specified IP addresses and provides for
strict IP address verification.
• The default SCCP port number is 2000.
The Cisco IOS command call-manager-fallback enters call-manager-fallback configuration
mode.
The Cisco IOS command ip source-address enables therouter to receive messages from the
Cisco IP phones through the specified IP addresses and provides for strictIP address
verification. The default port number is 2000. This IPaddress will besupplied later asan SRST
reference IP address in Cisco Unified Communications ManagerAdministration.
The ip source-address command is a mandatory command. Thefallback subsystem does not
start if the IP address ofthe Ethernet port towhich the IPphones are connected (typically the
Ethemet interface of the local Cisco Unified SRST gateway) isnot provided. Ifthe port number
is not provided, the default value (2000) is used.
The any-match keyword should beused to instruct the router topermit Cisco IP phone
registration even when the IP server address that is used by the phone doesnot match the IP
source address. You can use this option to allow registration of Cisco IPphones on different
subnetsor on subnetswithdifferent default DHCProutersor different TFTP server addresses.
The strict-match keyword should beused to instruct the router to reject Cisco IPphone
registration attempts ifthe IP server address that isused by the phone does not exactly match
the source address. By dividing the Cisco IP phones into groups on different subnets and giving
each group difTerent DHCP default-router orTFTP server addresses, this option can beused to
restrict thenumber of Cisco IP phones that areallowed to register.
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Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8.0
>2010 Cisco Systems, Jnc.
Cisco IOS Gateway SRST Configuration
This topic describes the configuration steps for enablingCisco Unified SRSTon the Cisco IOS
router.
Cisco Unified SRST Configuration
Procedure in Cisco IOS Router
:
Enable Cisco Unified SRST
2
Define IP address and port to which the SRST service
;;
Define maximum number of directory numbers to support.
binds.
a
Define maximum number of IP phones to support.
•::
Define maximum numbers allowed per phone type.
6
Define phone keepalive interval.
To configure Cisco Unified SRST on a Cisco IOS router tosupport the Cisco IP phone
functions, follow these steps:
Step 1
Fnable Cisco Unified SRST. sothat itenters call-manager-fallback configuration
mode.
Step 2
Define the IP address and port to which the SRS'f service binds.
Step3
Define the maximum number of directory numbers to support.
Step 4
Define the maximum number of IP phones to support.
Step 5
Define the maximum numbers thatare allowed per phone tjpe.
Step 6
Definethe phone keepalive interval.
Note
Steps 5 and 6 are optional.
Tip
When Cisco Unified SRST is enabled, Cisco IP phones do not have to be reconfigured while
in catl-manager-fallback configuration mode, because phones retain the same configuration
that was used with Cisco Unified Communications Manager.
© 2010 Cisco Systems. Inc.
Centralized Call-Processing Redundancy Implementation
2-41
Step 2: Device Pool
«&
The SRST reference is assigned to IP phones by using device pools.
Step 2: Device Pool
SRST reference to device pool assignment:
• SRST reference definition is selected from the drop-down
menu in the device-pool configuration.
Cisco Unified Communications ManagerAdministration > System > Device Pool
— Roaming Sai»rttv« bettings -
Dftte/Ti-nc Grouo'
Pe jion *
Media Resource Grcup Us*
Locator
Nefwort Locale
BkST Reference*
Connection Monitor Duration***
Physical Location
Ctv^e MabiJjty Group
Administrators select the configured SRST reference from the drop-down menu in the device
pool configuration.
Note
Ifdevices are associated with this SRST reference, a message is displayed, saying that
devices must be reset for the update to take effect.
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implemenbng Cisco Unified Communications Manager, Part 2 (CJPT2) v8.0
>201DCisco Systems, inc.
Step 1: SRST Reference
An SRST reference comprises the gateway, which can provide limited Cisco Unified
Communications Manager functionality when all other Cisco Unilied Communications
Manager servers for an IP phone are unreachable.
mm
Step 1: SRST Reference
Create a new SRST reference in the system menu:
• It must have a unique name.
• The IP address and port numbers of the Cisco Unified SRST
gateway must be defined.
Cisco Unified Communications ManagerAdministration > System > SRST
i— SftSI Reference Information
"amf *
SRST-Hemcltl
IF i^ddr-5G"
]•;;•;!
SIP NtVAC'k IP ASd-csi
|
5H5T CtrafiMte P'wJf Pjri' ;-i35
_j [s SSFT Sccurr'
SRST references detemiine the gateways that IP phones will search when (hey attempt to
complete a call if the Cisco Unified Communications Manager is unavailable.
Administrators must configure Cisco Unified Communications Manager with a uniqueSRST
reference name that specifies the IP address of (he CiscoUnilied SRSf gateway. The default
TCP port number 2000 is normally used.
The SIP network and IP address applies to Cisco Unified SIP SRST. If Cisco Unified SIP
SRST is used, the IP address and port that are used by the SIP protocol of the Cisco Unified
SRSf gateway have lo be specified: thedefault port number is 5060. Theconfigured address
and port will be used by SIP phones to register with theCisco Unified SIP SRST gateway.
For Cisco Unified SRSTgateways that supportSCCPphones with defaultport number2000
with secure SRST disabled, it is not necessary to add an SRST rclcrcncc if the IP address of the
Cisco Unified SRST gateway is the default gateway of the IP phone. In this case,you can use
the option Use Default Gateway at the device pool of the affected IPphones.
>2010 Cisco Systems, Inc.
Centralized Call-Processing Redundancy Implementation
2-39
Cisco Unified Communications Manager SRST
Configuration
This topic describes the configuration stepsfor enabling CiscoUnified SRST in the Cisco
Unified Communications Manager environment.
SRST Configuration Procedure in Cisco
Unified Communications Manager
1.
Create a new SRST reference.
2
Assign an SRST reference to device pool.
The role of Cisco Unified Communications Manager regarding the SRST feature is to provide
the phones with the needed information for finding the relevant SRST gateway to register with
when they lose contact with Cisco Unified Communications Manager subscribers.
The first step is to define an SRST reference. This reference contains information about IP
addresses and ports of SRST gateways for SCCP and Session Initiation Protocol (SIP) phones.
Because the SRST functions are different for SIP and for SCCP, the addresses and ports are
also different.
The second step is to provide a group of phones with this information by assigning the SRST
reference to a proper device pool, which is then assigned to the phones.
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MGCP Fallback and Cisco Unified SRST Configuration
Requirements
When configuring MGCP fallback and Cisco Unified SRST,you must take several steps at
different locations.
MGCP Fallback and Cisco Unified SRST
Configuration Requirements
MGCP fallback and Cisco Unified SRST configuration
consist of the following components:
• SRST references for phones have to be configured in Cisco
Unified Communications Manager.
• MGCP fallback and Cisco Unified SRST has to be enabled
and configured on the Cisco IOS gateway.
• The CFUR feature has to be configured on the Cisco Unified
Communications Manager to reach remote sites in SRST
mode.
• An SRST dial plan has to be implemented on the remote site
gateways to ensure connectivity for remote sites in SRST
mode.
You need to use the Cisco Unified Communications Manager Administration to define the
SRST references for phones. You must also configure (he Call Forward Unregistered (CFUR)
feature and set the CFUR destination of lines on remote site phones to the correct public
switched telephone network (PSTN) number on the Cisco Unified Communications Manager
Administration to enable reachable remote sites in SRS'f mode.
On Cisco IOS gateways, you must enable and configure the MGCP fallback and Cisco Unified
SRSf features. In addition, you must implement a simplified SRST dial plan on the remote site
gateways to ensure connectivity for remote sites in SRST mode.
© 2010 Cisco Systems, Inc.
Centralized Call-Processing Redundancy Implementation
2-37
MGCP Fallback and Cisco Unified SRST
Configuration Overview
This topic describes the high-level configuration of Cisco Unified SRSTand of MGCP gateway
fallback on Cisco IOS routers.
MGCP Fallback and Cisco Unified SRST
Configuration
Main Site
Remote Site
Cisco Unified
Communications
Manager
In Cisco IOS
Software, MGCP
fallback and Crsco
Unified SRST must
be enabled and
configured.
Activate and configure the MGCP gateway fallback feature on the Cisco IOS router.
Cisco Unified SRST must be configured on the side of the Cisco Unified Communications
Manager and on the side of the Cisco IOS router.
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Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0
>2010 Cisco Systems, Inc.
Lesson 2
-**»
Implementing SRST and
MGCP Fallback
Overview
fhis lesson describes how to configure CiscoUnified Survivable Remote SiteTelephony
(SRST) on Cisco IOS routers to provide redundancy to Cisco Skinny Client Control Protocol
(SCCP) phones. It also describes how to configure the Media Gateway Control Protocol
(MGCP) gateway fallback feature. In addition, this lesson illustrates how to configure features
like musicon hold (MOH)and voice-mail integration for Cisco Unified SRST.
Objectives
Upon completing this lesson. \ouwill be able to configure Cisco Unified SRST to provide call
survivability for IPphones, and MGCP fallback for gateway survivability, fhis ability includes
m+
beingable to meet theseobjectives:
• Describe the configuration requirements at Cisco Unified Communications Manager and at
the SRST gateway
•
Describe how lo configure Cisco Unified Communicalions Manager to enable SRSI for
remote phones
•
Describe how to configure a Cisco IOS router for SRS'f
a
Describe how to configure a Cisco IOS router to support MGCP fallback
•
Describe how to configure Cisco Unified Communications Manager to route calls to
unregistered devices via the PSTN
•
•mm
Describe how to configure a Cisco IOS router with a dial plan for SRST operation
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
MGCP fallback works in conjunction with Cisco Unified SRST to
provide telephony service to remote IP phones during WAN
failure.
The Cisco Unified SRST switchover time depends on the
keepalive timers and on the number of Cisco Unified
Communications Manager servers in the Cisco Unified
Communications Manager group applied to the remote phones.
The MGCP gateway fallback default application is H.323 or SIP.
The Cisco Unified SRST version is linked with Cisco IOS
Software release.
When Cisco Unified SRST is active, you must take several
measures to ensure connectivity from remote sites to PSTN
destinations, between different sites, and inside the site itself.
References
For additional information, refer to these resources:
•
Cisco Svstems. Inc. Cisco UnifiedCommunications System8.x SRND, April 2010.
http://'\vw\v.eiseo.com/cn/L^S/docs/voiee_ip_comm/ciiem/srnd/8x/uc8x.html
•
Cisco Systems. Inc. Cisco UnifiedCommunications ManagerAdministration Guide
Release 8.0(1). February 2010.
hltp://«ww.cisco.com/en/US/docs/voice ip comin/cucm/admin/8 0 l/ccmcfg/'bccm-801cm.html
•
Cisco Systems. Inc. Cisco UnifiedSurvivable RemoteSite Telephony Version 8.0,
November 2009.
http:.;V\\w'w,eisc().com/en/l!S/pmd/'collateral/voiccsvv/ps6788/Vcallcon/ps2l6Wdata^heet e
78-57048l.html
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Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
© 2010 Cisco Systems, Inc.
Cisco Unified SRST Dial Plan Requirements Example
Call-routing components on Cisco IOS routers and Cisco Unified Communications Manager
arc necessary before a dial plan will work in SRS'f mode.
Cisco Unified SRST Diaf Plan
Requirements Example
The dial plan enabled by Cisco Unified SRST has to
include the following components:
• CFUR on the Cisco Unified Communications Manager side
• Dial peers, COR, dial plan pattern, and voice translation
profiles on the Cisco IOS router site
Dial peer with destination pattern
Remole Site
.?.T (MIntsto_PSTN
Cisco
Unified
COR lists deny access for
900-services dial jjeer
Communications
Manager
Configuration includes
dial plan pattern and voice
translation profiles to allow
extension-only dialing.
CFUR must be defined on the Cisco Unified Communications Manager side. Configuring the
Cisco IOS router is a little more complex when vou use dial peers, COR. dial plan pattern, and
\oice translation profiles to define the simplified Cisco Unified SRST dial plan.
© 2010 Cisco Systems. Inc
Centralized Call-Processing Redundancy Implementation
Keeping Calling Privileges Active in SRST Mode
Undernormal conditions in multisite deployments with centralized call processing, you
implement calling privileges by using partitions and CSSs within Cisco Unified
Communications Manager.
Keeping Calling Privileges Active in SRST
Mode
Calling privilegesare implemented in Cisco Unified
Communications Manager by using partitions and CSSs.
In the case of a WAN failure, all centralized routing
information is lost, including mappings of calling privileges.
COR is the mechanism to use in implementing calling
privileges on SRST routers:
- Tags have to be defined for each type of call.
- Outgoing COR lists containing a single member tag
correspond to partitions.
- Incoming COR lists containing subsets ofthe member
tags correspond to CSS.
However, when IP WAN connectivity is lost betweena branch site and the central site, Cisco
Unified SRSTtakes controlof the branch IP phones, and the entire configuration that is related
to partitions andCSSs is unavailable until IP WAN connectivity is restored. Therefore, it is
desirable to implement classes of service within the branch router when running in SRST mode.
Forthis application, you mustdefine classes of service in Cisco IOSrouters by using the class
of restriction (COR) functionality. You can adapt the COR functionality to replicate the Cisco
Unified Communications Manager concepts of partitionsand CSSs by following these main
guidelines:
•
•
Define named tags for each type of call that you want to distinguish.
Assign basic outgoing COR listscontaining a single tag each to the outgoing dial peersthat
should not be available to all users. These outgoing COR lists are equivalent to partitions in
Cisco Unified Communications Manager.
•
Assigncomplexincoming COR lists containing one or more tags to the directory numbers
that belong to the various classes of service.
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>2010 Cisco Systems, Inc.
CFUR Example with Globalized Call Routing
The figure shows the same scenario, but thistimewith globalized call routing.
CFUR Example with Globalized Cal
Routing
Route Pattern
Partition. System
Device Pool
LRG Site2-GW
3001
Phone
CFUR *152155530O1
CFUR CSS System
There is only a single \+! route pattern, the referenced route list has local route groups enabled.
All phones use the same CFUR CSS. which provides access to the partition of the global route
pattern, fhe egress gatewa; is selected by the local route group feature. Localization of the
called number occurs at the egress gateway by global transformations.
Ifa called is placed to an unregistered phone of Site l, die CFUR destination+152I5553001 is
called using the single off-net route pattern, which is configured to use the local route group (in
the referenced route list). Consequently, like with any other PSTN call, CFUR calls use the
local gateway instead of the HQ gateway, regardless of the location of the caller. There is no
need for all callers to use the same gateway for CFUR calls. In addition, all CFUR destination
numbers are specified in global format (E. 164 with +- prefix).
'2010 Cisco Systems, Inc.
Centralized Call-Processing Redundancy Impfementation
2-31
CFUR Example Without Globalized Call Routing
"Hie example illustrates the call flow with CFUR without local route groups.
CFUR Example Without Globalized Call
Routing
CFUR caii from HQ
phone to Stie 1 phone
uses MO PSTN
galeway
Route List
Route Group: HQ-GW
Ptione
CFUR 915215553001
JP ;.,)!: K)ir> hiic;2 ph:*^; tc '~Wt ' pSn">si<; t;;K-.s HG
CFUR CSS HQ
There are three sites: HQ, Site I, and Site 2. The remotesites are backed up by SRS'f gateways.
If IP connectivity between site 1 andthe HQfails, Site 1 phones will failover to SRST mode.
They can still call the HQ and Site 2 viathe PSTN. When an HQ phone attempts to call a phone
at Site 1—which is unregistered in Cisco Unified Communications Manager—the call is placed
to the CFUR destination configured at the Site 1phone (915215553001 in thisexample). The
CFUR CSS of the Site 1 phoneensuresthat a route pattern9.@—which refers to the HQ
gateway—can be accessed. Therefore, the call is redirected to the PSTN number of the called
phone and sent to the HQ gateway.
When a user at Site 2 attempts to call a phone at Site 1, the same thing happens. The CFUR
destination 915215553001 is called using the CFUR CSS configured at the Site 1 phone and
therefore matches the 9.@ route pattern that is referring to the HQgateway and not to a 9.fy
route pattern referring to a Site2 gateway. Therefore, the call will utilize the IP WAN to get
from Site 2 to the HQ and from there it will break out to the PSTN towardsSite 1.
If there were more sites, they would all use the HQ gateway for CFUR calls to Site 1. This can
lead to suboptimal routing. In addition, differentroute patterns may be needed depending on
the destination of the CFUR call. In an international deployments, the CFUR destination
numbermay be a mix of national and international numbers. Each destination numberhas to be
specified in a way that it can be routed by the CFUR CSS. There is no common format for all
CFUR destinations—some may be specified in national format, others in international format.
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Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
© 2010 Cisco Systems, Inc.
CFUR Interaction with Globalized Call
Routing (Cont.)
Optimized gateway selection with multiple sites:
• Without using local route groups the CFUR CSS determines
the gateway used for the CFUR call
* The CFUR CSS is set at the phone that is unregistered {in
SRST mode)
- All callers use the same CFUR CSS to reach such an
unregistered phone
• Ifcallers are at different sites, they all have to use the same
gateway (typically the main site gateway is used)
- With local route groups each spoke site can use its local
gateway for CFUR calls
• CFUR number is the same for all callers regardless of the
originating site
Without using local routegroups, the CFUR CSS determines the gateway that is used for the
CFUR call. The CFUR CSS of the phonethat is unregistered is used not the one of the phone
that tries to reach the unregistered phone. This means that all callers use the same CFUR CSS
when calling an unregistered phone (the CFUR CSS configured at the destination phone).
Consequently, if callers arc located at different sites, they will all use the same gateway for the
CFUR call. Usually the main site gateway is used for that purpose; that means that the CFUR
CSS (applied to all phones) provides access lo PSTN route patterns that use the main site
gateway (via the referenced route list and route group).
With local route groups, each caller can use its local gateway for CFUR calls; there is no need
to use the IP WAN touards the main site and then break out to the PS'FN with the CFUR call at
the main site gateway. Depending on the deployment this can be a huge improvement for
reaching sites that lost IP connectivity to Cisco Unified Communications Manager.
© 2010 Cisco Systems. Inc
Centralized Call-Processing Redundancy Implementation
2-29
CFUR Interaction with Globalized Call Routing
CFUR canbenefit from globalized call routing in multisite deployments, especially when a
Cisco Unified Communications Managercluster serves multiplecountries.
CFUR Interaction with Globalized Cal!
Routing
CFUR can benefit from globalized call routing:
- Ifglobalized number is used as CFUR destination number
- CFUR calls are placed to global number
- Single route pattern (\+!) sufficient for all CFUR calls
- Same route pattern can be used for AARand PSTN
access
- Route pattern refers to single route list
- Route list includes only "Standard Local Route Group"
- CFUR CSS can be the same for all phones
If globalized numbers arc used as CFURdestinations, callsto unregistered phones (for
example, phones that lost IP connectivity to Cisco Unified Communications Manager and
which are in SRST mode) are using the only configured off-net route pattern \+! for CFUR. All
calling devices will use the same route pattern, route list, and route group to place the call, "fhis
route pattern is a general off-net route pattern and is used for PSTN calls, AAR calls, as well as
by CFUR calls, 'fhe CFUR CSS can be the same for all phones and the local gateway will be
used for the CFUR call because local route groups are configured.
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Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
>2010 Cisco Systems, Inc.
a director;' numberwhose CFUR setting is configured for voice mail. At the same time, this
configuration would alsolimit potential loops to two fordirectory numbers whose CFUR
configuration sends calls through the PSTN.
Note
Cisco Unified Communications Manager Extension Mobility directory numbers should not be
configured to send CFUR calls to the PSTN DID that is associated withthe directory
number. The directory numbers of Cisco Unified Communications Manager Extension
Mobility profiles in the logged-out state are deemed to be unregistered; therefore, any calls
to the PSTN DID number of a logged-out directory number would trigger a routing loop To
ensure that calls made to Cisco Unified Communications Manager Extension Mobility
directory numbers in the logged-out state are sent to voice mail, you must configure their
corresponding CFUR parameters to send calls to voice mail.
) 2010 Cisco Systems, Inc
Centralized Call-Processing Redundancy Implementation
2-27
CFUR Considerations
CFUR was first implemented in Cisco Unified Communications Manager Version 4.2.
CFUR Considerations
CFUR was introduced in Cisco Unified Communications
Manager v4.2.
CFUR points to PSTN destinations or voice mail.
Using CFUR to forward calls to PSTN number for
disconnected phones in remote sites causes routing loops.
To reduce the impact of routing loops, service parameter
limits the number of CFUR hops per call.
CFUR for Cisco Extension Mobility lines should always point
to voice mail to avoid routing loops.
As mentioned earlier, the CFUR feature allows calls that are placed to a temporarily
unregistered phone to be rerouted to a configurable number.The configuration of CFUR
comprises two main elements;
•
Destination selection: When the directory number is unregistered, calls can be rerouted to
voice mail or to the directory numberthat was used to reach the phone through the PSTN.
•
Calling search space (CSS): Cisco Unified Communications Manager attempts to route
the call to the configured destination number using the CFUR CSS of the directory number
that was called. The CFUR CSS is configured on the target phone and is used by all devices
that are calling the unregistered phone.
If a phone is unregistered while the gateway that is associated with the direct inward dialing
(DID) number of that phone is still under the control of Cisco Unified Communications
Manager, CFUR functionality can result in telephony routing loops. For example, if a phone is
simply disconnected from the network, the initial call to the phone would prompt the system to
attempt a CFUR call to the DID of the phone through the PSTN. The resulting incoming PSTN
call would, in turn, trigger another CFUR attempt to reach the directory number of the same
phone, triggering yet another CFUR call from the central PSTN gateway through the PSTN.
This cycle could repeat i'sclf until system resources are exhausted.
'fhe Cisco CallManager service parameter Max Forward UnRegistered Hops to DN in the
Clusterwide Parameters (Feature—Forward) section in Cisco Unified Communications
Manager Administration controls the maximum number of CFUR calls that are allowed for a
directory number at one time. The default value of 0 means that the counter is disabled. If any
directory numbers are configured to reroute CFUR calls through the PSTN, loop prevention is
required. Configuring this service parameter to a value of 1 would stop CFUR attempts as soon
as a single call is placed through ihe CFUR mechanism. This setting would also allow only one
call to be forwarded to voice mail, if CFUR is so configured. Configuring this service
parameter to a value of 2 would allow up to two simultaneous callers to reach the voice mail of
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Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
) 2010 Cisco Systems, Inc.
Ensure Connectivity from Main Site Using CFUR
During fallback, main site users still should be able locall remote site users by using their
extension numbers.
Ensure Connectivity from
Using CFUR
in
Remote site has lost connectivity to main site. Phones are
registered to remote gateway.
• Cisco Unified Communications Manager at the main site does not roule calls
to the directory numbers of affected IP phones, which are now unregistered
in Cisco Unified Communications Manager.
• CFUR allows routing to alternate numbers for affecled (unregistered) IP
phones.
Main Site
Remote Site
1001 status is
unregistered
Configured CFUR
9 1 408 555-1001
Communications
Manage
DID' 408 555-
Remote
100110 1003
Gateway
Cisco Unified Communications Manager considers the remote site phones as unregistered and
cannot route calls to the affected IP phone directorv' numbers. Therefore, if main site users dial
internal extensions during the IP WAN outage, the calls will fail (or go to voice mail).
To allow remote IP phones to be reached from other sites, you can configure Call Forward
Unregistered (CFUR) at the remote site phones. You should contigurc the CFUR destination at
each remote IP phone with the PSTN number of the IP phone so that internal calls from other
sites get forwarded to the PSTN number of an IP phone that is currently unregistered and is
therefore not reachable over the IP network.
©2010 Cisco Systems. Inc
Centralized Cal I-Processing Redundancy Implementation
2-25
Ensure Connectivity for Remote Sites
When Cisco Unified SRST is active, you musttakeseveral measures to ensure connectivity
from remotesites to PSTN destinations, betweendifferentsites, and insidethe site itself.
Ensure Connectivity for Remote Sites
PSTN connectivity:
* You must implement dial peers with destination patterns
corresponding to the PSTN access code.
• Voice translation profiles modify the calling-party number to
enable callback.
* Interdigittimeout adopts open numbering plans that do not
have a fixed number of digits.
Intrasite and intersite connectivity:
• Voice translation profiles expand the called number to PSTN
format for site code dialing.
* The command dialplan pattern modifies incoming called
PSTN numbers to match internally used extensions.
To guarantee PSTN connectivity, you must implement dial peers with destination patterns
corresponding to the PSTN access code. In H.323or SIP gateways, these dial peers must be
presentfor normal operation. When MGCP gateways are used, dial peers are activated by the
MGCP gateway fallback mechanism. Interdigit timeout adopts open numbering plans that do
not have a fixed number of digits.
Voicetranslation profilesthat are applied to dial peers, the voice interface, or the voice port
modify the calling party ID to enable callback from call lists.
For intrasite and intersite connectivity, voice translation profiles are configured to expand
called numbers to PSTN format during fallback.
The Cisco IOS command dialplan-pattern in call-manager-fallback configuration mode
modifies incoming called numbers to match the remote site extensions, 'fhis command also
ensures that internal extensions can be dialed even though the lines are configured with the site
code and extension, "fhe Line Text Label settings that are defined in Cisco Unified
Communications Manager will not be applied to the Cisco Unified SRST phones, so the
complete directory number that is applied to the line will be visible to the user.
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Dial Plan Requirements for MGCP Fallback and
Cisco Unified SRST Scenarios
Thistopic describes therequirements of standalone dial plans for working with MGCP fallback
and Cisco Unified SRST.
Cisco Unified SRST Dial Plan Requirements far
Calls from the Remote Site
Cisco Unified SRST-mode dial plan should be as close to
normal mode as possible:
• Remote site users need to reach one another by extension.
• Remote site users need to reach the internal extensions of main site users.
* Remote site PSTN access should workas usual, includingclass of service.
isco Unified Communicalions
Calls to main sile extensions
Manager applies the
need to be changed to PSTN
dial plan for all sites
numbers; class ol service
during normal operation
must be enforced
Unified
Communicalions
Manager
Cisco Unified SRST
needs a simplified dial
plan for remote sites in
SRST mode
SRST failo\er lea\es the remote site independent from the complex dial plan that is
implemented in Cisco Unified Communications Manager in the main site. The Cisco Unified
SRST router needs to have a minimal dial plan that is implemented lo allow for all remote site
phones, all main site phones, and all PSTN destinations to be reached with the same numbers as
in standard mode.
During fallback, users should be able to dial main site directorv' numbers as usual. Because
these calls have to be routed over the PSTN during fallback, main site extensions have to be
translated to E.164 PSTN numbers at the PSTN gateway.
Most enterprises limit the range of destinations that are reachable from specific extensions by
applying a class of service to the extensions, fhis limitation should still be valid during times in
SRST mode.
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Support for Multiple MOH Sources
Cisco Unified SRST v8.x also introduces support for multiple music on hold (MOH) sources.
Support for Multiple MOH Sources
Cisco Unified SRST v8.0 supports up to five MOH groups:
• Supports SCCP phones only.
• Each group can be configured with individual MOH file.
• Multicast MOH can be enabled or disabled per MOH group.
« Each group is configured with directory numbers that should be
applied to it.
• Supported by Cisco Unified Communications Manager Express in
SRST mode, as well.
• Files can be cached in router RAM.
• Traditional MOH configuration still supported,
- Configured under Cisco Unified SRST (standalone or Cisco
Unified Communications Manager Express)
- Applies toall phones thathave no MOH group assigned
* All SIP phones
• SCCP phones if directory numbers are not specified in MOH
group
Before Cisco Unified SRST v8.x. only asingle MOH file was supported by Cisco Unified
SRST. Cisco Unified Communications Manager Express in SRST mode, and Cisco Unified
Cisco Communications Manager Express in standalone mode.
Cisco Unitied SRST v8.x allows you toconfigure up tofive additional MOH sources by
configuring MOH groups.
Only SCCP IP phones support these newly introduced MOH groups. You can configure each
MOH group with an individual MOH file that is located in the flash memory of the router, and
you can enable multicast MOH for each MOH group. Each MOH group is configured with the
directory' number ranges that should utilize the corresponding MOH group when callers are put
on hold.
The traditional MOH configuration for Cisco Unified SRST and Cisco Unified
Communications Manager Express is still supported. Itisused by all phones that do not have a
MOH group assigned. All of these phones are SIP and SCCP phones whose directory numbers
have not been specified in any MOH group.
MOH files can becached inrouter RAM. This process isuseful toreduce the amount of read
operations in flash, but it requires enough available RAM at the router. You can specify a
maximum size per MOH file in order to limit RAM usage for MOH file caching.
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Plus (+) Prefix and E.164 Support in Cisco Unified SRST
Cisco Unified SRST version 8.0 introduces support for directory numbers in E. 164 format with
a plus (-t-) prefix.
Plus (+) Prefix and E.164 Support in
Cisco Unified SRST
Cisco Unified SRST8.0supports directory numbers with +
prefix:
• IP phones can usedirectory numbers in E.164 format with +prefix.
* SIP and SCCP phones supported
• Special dial plan pattern command can be used to demote E.164
number to internal extension.
- Allows internal extensions to be used to call phones with E 164
number
Reversefunctionality from standarddial planpattern command
• Calls from outside are placed to E 164 numberwith + prefix.
« Voicetranslation profiles are extended by translationof callback
number
- Allows separate translation of calling-party numberand callback
number
Enables different number to be used for callback rather than
numbershown on phone display
SIP and SCCP IP phones can fallback to SRST and register with adirectory number in E. 164
format with a+prefix. Assigning directory numbers in E. 164 format ensures globally unique
numbers: the + sign is prefixed in orderto indicate thatthe number is in E. ]64 format.
Note
The E 164standard describes telephone numbers in international format. E 164numbers
are globally unique numbers within the PSTN and startwith the country code.
Cisco Unified SRST and Cisco Unified Communications Manager Express in SRST mode
allow internal callers to use internal extensions for calling IPphones that have numbers in
E.164 format. A new dial plan pattern command hasbeen introduced with Cisco Unified SRST
v8.x to achieve the demotion of the I:. 164 number to the internally used shorter numbers. While
the standard dial plan pattern command expands to a longer PSTN format any directory
numbers that are applied tophone lines, the new dial plan pattern command has the opposite
function: Itallows internal callers to dial shorter, internally used extensions, which are
expanded to the applied directory numbers in E.164 format.
Outside callers dial the IP phone directory numbers as configured—with a 4 prefix and the
complete E.164 number.
At the IPphones, the calling-party number that isshown on the phone display can be
transformed independently from the number that will be used for callback. This transformation
is possible because ofa newly introduced translation type in voice translation profiles—a
translation rule of the callback number.
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Lesson 4
Implementing a Dial Plan for
International Multisite
Deployments
Overview
Multisite dial plans have to addressspecial issues,such as overlapping and nonconsecutive
director;'numbers, publicswitchedtelephone network(PSTN)access, PSTN backup,and tailend hop-off(TEHO). This lessondescribes how to build multisitedial plans using Cisco
Unified Communications Managerand Cisco IOS gatewayconfiguration. The lessonalso
describes the conceptof globalized call routing—a new way of buildingdial plans in
international multisite deployments.
Objectives
Uponcompleting this lesson,you will be able to implement a dial plan to supportinbound and
outbound PSTN dialing, site-code dialing, and TEHO in an international environment. This
ability includes being able to meet these objectives:
•
•
List dial plan issues and possible solutions
Describe how site codes and transformation masks solve issues that are caused by
overlapping directory numbers
•
Describe how to implement PSTN access in a multisite deployment
•
Describe how to implement selective PSTN breakout
•
Describe how to use the PSTN as a backup for calls to other VoIP domains
•
Describe how to implement TEHO
•
Describe the concept of globalized call routing and how it simplifies dial plans in
international multisite deployments with centralized call processing
•
Explain special considerations for implementing globalized call routing
Multisite Dial Plan Overview
This topic lists dial plan requirements for multisite deployments with centralized call
processing.
Dial Plan Requirements for Multisite Deployments
with Centralized Call Processing
Implementing access and site codes;
- Allows routing independent of directory numbers
Solves overlapping and nonconsecutive directory-number
ranges
Implementing PSTN access:
- Simple, prioritized list of gateways forall PSTN access
TEHO (gateway selection based on PSTN destination)
Implementing PSTN backup:
MGCP fallback in case of MGCP gateways
- SRST for IP phone fallback at remote site
- Call Forward Unregistered at central site
In multisite environments with centralized call processing, you use these dial plan solutions:
•
Access and site codes: By adding an access code and a site code lo director},' numbers of
remote locations, you can provide call routing that is based on the site code instead of on
directory numbers. As a result, directory numbers do nol have to be globally unique,
although they must be unique within a site. Configuration requires route patterns,
translation patterns, partitions, and calling search spaces (CSSs).
•
Implementing PSTN access: You implement PSTN access within a Cisco Unified
Communications Manager cluster by using route patterns, route lists, route groups,
partitions, and CSSs. When implementing TEHO. you use the same dial plan configuration
elements: however. \ou have lo configure more entities, which makes the configuration
more complex.
•
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Implementing PSTN backup: The IP WAN that is used in a multisite deployment with
centralized call processing is backed up by Media Gateway Control Protocol (MGCP)
fallback. Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST) or Cisco
Unified Communications Manager Express in SRS'f mode, and Call Forward
Unregistered(CFUR).
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Dial Plan Requirements for Multisite Deployments with
Distributed Call Processing
Dial plan requirements for multisite deployments with distributed call processing are like the
dial plan requirements ofmultisite deployments with centralized call processing.
Dial Plan Requirements for Multisite Deployments
with Distributed Call Processing
Implementing access and site codes:
- Allows routing independent of directory numbers
- Solves overlapping and nonconsecutive directory number
ranges
Implementing PSTN access:
- Simple, prioritized listofgateways forall PSTN access
- TEHO (gateway selection based on PSTN destination)
Implementing PSTN backup:
- Route lists and route groups for path selection:
• First choice: on-net
• Second choice: off-net
In multisite environments with distributed call processing, you use these dial plan solutions:
•
Access and site codes: Byadding an access codeanda site codeto directory numbers of
remote locations, you can provide call routing thatis based on thesite code instead of on
directory' numbers. As a result, directory numbers do nothave to be globally unique,
although they must be unique within a site. Configuration elements include route patterns
and translation patterns.
•
Implementing PSTN access: You implement PSTN access within a Cisco Unified
Communications Manager cluster by usingroute patterns, route lists, routegroups,
partitions, and CSSs. When implementing TEHO, you use thesame dial plan configuration
elements; however, you have to configure more entities, which makes the configuration
more complex.
•
ImplementingPSTN backup: Backup of the IP WAN is provided by route lists androute
groups with on-net (prioritized) and off-net (PSTN)paths.
) 2010 Cisco Systems, Inc
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Dial Plan Scalability Solutions
Dial plan requirements for multisite deployments with distributed call processing are like the
dial plan requirements of multisite deployments with centralized call processing.
Dial Plan Scalability Solutions
aH Agent
Ca IASeni Call Agent
Call Agent
Call Agent CallAgent
Dial plan scalability issues can be solved by Call Control Discovery (CCD)
CCD allows dynamic exchange of call routing information utilizinga SAF-enabled
network
In large networks with severalcall agents—such as Cisco Unified Communications Manager
Express. Cisco Unified Communications Manager, Cisco Unified Border Element, Cisco
Unified SRST.and Cisco IOS gateways—the implementation and maintenance of dial plans
can be very complex.
Without centralized services (such as H.323 gatekeepers or SIP network services), a full-mesh
configuration is required. In other words, each call control domain has to be configured with
call-routing information toward all other call-routing domains, fhis implementationmodel does
not scale at all and therefore is suitable only for smaller deployments.
In a hub-and-spoke deployment model, call-routing infonnation for each call-routing domain is
configured onh once at the centralized call-routing entity. This centralized call-routing entity
can be a SIP network service or an H.323 gatekeeper. Such a solution scales better than lullmesh topologies: however, it introduces a single point of failure and therefore requires
redundant deployment of the centralized service. In addition, the centralized call routing still
has to be manually configured. For example, if telephone number ranges or prefixes are
changed at one of the call-routing domains, these changes also have to be manually perfonned
at the centralized call-routing service. Further. PSTN backup has to be implemented
independent!; ai each call-routing domain.
With Call Control Discovery (CCD). a new feature that was introduced with Cisco Unitied
Communications Manager Version 8. each call-routing domain advertises locally known
telephone numbers or number ranges. Because local numbers are typically used by internal
patterns (using VoIP) as well as via the PSTN, each call-routing domain advertises both the
internally used numbers and the corresponding external PSTN numbers.
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© 2010 Cisco Systems. Inc.
CCD solvesdial plan scalability issues by allowingCisco Unified Border Element, Cisco
Unified SRST.Cisco Unified Communications Manager, Cisco Unified Communications
Manager Express, andCisco IOS gateways to advertise and learn call-routing information in
the form of internal directory numbers and PSTN numbers or prefixes.
CCD utilizes the Cisco ServiceAdvertisement Framework (SAF).SAF is a network-based,
scalable, bandwidth-efficient, real-time approach to service advertisement and discovery.
Note
CCD is described in more detail in a later lesson of this course.
r
mm
>2010 Cisco Systems. Inc
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Implementing Site Codes for On-Net Calls
This topic describes how to use site codes in multisite environments.
Access and Site Codes
Users dial access and site code, followed by directory number;
directory numbers do not have to be unique across sites because of
site identification
Cisco Unified
Communication;]
Manager
Nonconsecutive
£000-2157
2365-2999 "♦"
Numbers
Inthe example, two sites have overlapping and nonconsecutive directory' numbers. To
accommodate unique addressing of all endpoinls. site-code dialing is used. Users dial anaccess
code (8 in this example), followed by a three-digit site code. When calling thephone with
directory number 1001 at the remote site, a user who islocated atthe main site has to dial
8222.1001. For calls in the other direction, remote usersdial 8111-1001. Whendistributed call
processing is used, each Cisco Unified Communications Manager cluster isaware ofonly its
own director) numbers indetail. For all directory numbers that are located at the other site, the
call is routed to a Cisco linified Communications Manager server at the other site that is based
on the dialed site code.
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Digit Manipulation Requirements for Using Access and Site
Codes
The figure shows digit manipulation requirements for site code implementation.
Digit Manipulation Requirements When
Using Access and Site Codes
Remote Site
Main Site
Cisco Unified
Communications
Mm
Manager
Outgoing: Cisco Unified
Incoming: Cisco Unified
Communications Manager
strips access code and site code
Communicalions Manager
adds access code and
from called number.
site code to calling number.
When you are using site codes in mullisite environments with distributed call processing, call
processors must strip off the access and site code from the called number on outgoing calls. If
access and site codes are configured before the "." (dot) in the route pattern, you can easily strip
them offusing the discard digits instruction (DDI) on the route pattern orroute list. For
incoming calls, you need to use translation patterns to add the access code and appropriate site
tow
code that are used to get to ihe callersite.
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Centralized Call-Processing Deployments: Access and Site
Codes
Ifoverlapping director) numbers exist in acentralized call-processing deployment, access and
site codes are implemented in adifTerent way. as shown in the figure.
Centralized Call-Processing
Deployments: Access and Site Codes
Cisco Unified
Communications
Manager
The example shows two sites with centralized call processing. Directory numbers in the main
site("headquarters." or "HO" in the figure) and the remote site(-'branch." or "BR'" in the
figure) partially overlap. Again, access and site codes are used losolve the problem of
overlapping directory numbers.
However, in this case, partitions and CSSs need lobedeployed ina way that phones at the
remote site do notsee director) numbers of main-site phones, and vice versa. Then a translation
pattern is added per site.
The translation pattern of each site includes the access and site code of the respective site.
Phones ateach site have a CSS assigned, which provides access tothe directory numbers of the
local site and the translation pattern for the other site orsites, fhetranslation patterns are
configured witha transformation mask that strips off the accesscode and site code. Further,
each translation pattern must have a CSS. which provides access to only those director)
numbers that are located at the target site ofthe respective translation pattern, fhis way., all
phones can dial local directory numbers and site-code translation patterns for accessing other
sites. After a user dials an intersite number(composed of the accesscode,site code, and
director)' number), the corresponding translation pattern is matched. The translation patlern
strips the site code and access code sothat only the directory number remains. This directory
number is matched again in the call-routing tableusing a CSS that hasaccess only to the
directory numbers of the site, which was identified bv the site code.
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Implementing PSTN Access in Cisco IOS
Gateways
This topic describes how to implement PSTN access in a multisiteenvironment.
Dial Plan Requirements for PSTN
Access
Perform this digit manipulation at Cisco Unified Communications
Manager (in the case of MGCP gateways) or at the H.323 gateway:
• Outgoing calls (to PSTN).
- Calling number transfix ma Iion
• If no DD range, transform all directory numbers to (single)PSTN number.
•
-
If DID range, add prefix
Called number t"an storm at ion
*
Strip access code
• Incoming calls (from PSTN).
-
-
Calling number transformation
•
Transform to complete PSTN number and add access code.
•
ConsiderTON.
Called number transformation
• If DID range, stnp PSTN prefix to directory number.
• If no DO, route call to attendant or IVR application.
When implementing PSTN access, the following digit manipulation has tobe performed b:fore
the call is sent outto thePSTN. Digit manipulation has to bedone in Cisco Unified
Communications Manager when anMGCP gateway isbeing used, and it can beperformed
either in Cisco Unified Communications Manager orat an H.323 gateway.
•
Outgoing calls to the PSTN:
—
Calling number transformation: Ifno direct inward dialing (DID) range isused at
the PSTN, transform all directory numbers to the same, single PSTN number in the
calling number. If DID is used, extend the directory numbers to a complete PSTN
number.
—
•
Called number transformation: Strip the access code.
Incoming calls from the PSTN:
—
Calling number transformation: Transform calling number loa complete number
{considering the ISDN typeof number [TON]), andadd the access code.
—
Called number transformation: If DID isused, strip offthe office code, area code,
and country- code (if present) to getto thedirectory number. If DID is notused, route
the call tothe attendant ortothe interactive voice response (IVR) application.
>2010 Cisco Systems, Inc
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PSTN Access Example
The figure illustrates an example ofdigit manipulation that isperformed for incoming and
outgoing PSTN calls.
PSTN Access Example
408 555DID XXXX
714 555-2222
PSTN
Outgoing Cisco Unified
Call from 1001 to
9 1 714 555-2222
Call from 408 5551001 to 1 714 555-
Communications Manager or H 323
gateway strips access code from
called number.
2222
Cisco Unified Communications
Manager or H.323 gateway adds
PSTN prefix to called number.
Call from PSTN
Incoming Cisco Unifed Communications
Manager or gateway adds access code and
longdistance 1 to calling number. Cisco
Call from 9.1-
Call from 714
714 555-2222
to 1001
555-2222 to 408
555-1001
Unified Communications Manager or gateway
stnps PSTN prefixfrom called number.
As shown in the example, internal numbers have to be represented as valid PSTN numbers, and
PS'fN numbers should be shown with the accesscode 9 internally.
Note
Adding the access code (and changing 10-digit PSTN numbers to 11-digit PSTN numbers,
including the long-distance "1" digit) tothe calling number ofincoming calls isnot required.
Adding it. however allows users to call back the number from call lists (such as received
calls or missed calls) without having toedit thenumber by adding therequired accesscode
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ISDN TON
fhis subtopicdescribes how to manage PSTN numbers that are based on their TON.
ISDN TON
U.S. TON in ISDN provides infonnation about number format:
• Subscriber
-
7-digit subscriber number
- 3-dig* exchange code
• 4-digit station code
-
National
10-dgrt number
• 3-digt area code
• 7-digit subscriber number
• International
Variablelength (11 digitsfor U.S. numbers)
• Country code (1 digitfor U.S. country code 1)
* Area coders digits for U.S. area code)
• Subscribernumber (7 digitsfor U.S.subscriber number)
| Number has lobe transformed differently depending onthe TON.
The TON isused tospecify in which format a number (such as calling number orcalled
number) isrepresented. Tohave a unique, standardized way to represent PSTN numbers in
Cisco UnifiedCommunications Manager, the numbers have to be transformed based on the
TON.
For example, ifthecalling number of an incoming PSTN call is received with a TON
subscriber, the PSTN access code can be prefixed so that the user can place a callback without
editing thenumber. Ifthe calling number is in national format, then the PSTN access code and
the national access code areprefixed. Ifa calling number is received with an international
TON. the PSTN access code and the international access code are prefixed.
In countries with fixed-length numbering plans, transforming the numbers is not required,
because users can identify the type ofcalling number that isbased on the length. In this case,
users can manually prefix the necessary access codes. In countries with variable-length
numbering plans, however, it can be impossible to identify whether thecallwas received from
the local area code, from another area code ofthe same country, orfrom another country by just
looking atthe number itself. In such cases, the calling numbers ofincoming PSTN calls have to
be transformed based on the TON.
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Example: ISDN TON—Calling Number Transformation of
Incoming Call
The figure illustrates an example ofperforming TON-based digit manipulation on the calling
number of the incoming call.
Example: ISDN TON—Caliing Number
Transformation of Incoming Gall
Site 1
Mam Site
403 555-1111
Site 2
14 555-2222
408 555-
Cisco Unified
Communications
DIDXXXX
Manager
PSTW
^^1^4132673333
Incoming Calls
with Different
TONS
1001-1099
Site
1
2
~3
TON
I Subscr&er
National
International
Calling Number
5551111
Required Catling Number Transformation/
9.5551111
I7145552222
19.1714555 2222
j49404132673333
}9.011464O4132673333
In the example, the main site gateway receives three separate calls, and callbacks should be
possible without requiring the user to edit the number. The first call is received from the local
area with a subscriber "ION and a seven-digit number. This number needs only to beprefixed
with access code9. Thesecond call,which is received with national TON and 10digits, is
modified by the addition ofaccess code 9and the long distance I,all ofwhich are required for
placing calls back to the source ofthe call. The third call is received from another country
(Germany, in this case) with an international TON. For this calk the access codes 9and 011
ha\c to be added to the received number, which begins with the country code.
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>2010 Cisco Systems, Inc.
Implementing Selective PSTN Breakout
This topic describes how to implement selective PSTN breakout in a multisite environment.
Selective PSTN Breakout Configuration
Options
• Gateway selection by CSS
- Each site uses a device-specific CSS.
- Site-specific route pattern is matched.
- Route pattern refers to site-specific route list.
- Route list includes site-specific route group,
- Route group refers to site-specific gateways.
• Gateway selection by local route group feature
- Matched route pattern refers to systemwide route list.
- Route list refers to Standard Local Route Group.
- Device pool of calling device is configured with route group
to be used.
- Eliminates the need for multiple (site-specific) route lists,
route patterns, partitions, and CSS.
- Preferred option since the introduction of local route groups
(Cisco UnifiedCommunications Manager \fersion 7).
"ITiere aretwo ways to select the local gateway forPSTN calls. One way is to configure a sitespecific set of route pattern, partition, CSS, route list, androutegroup. If you apply a sitespecific CSS at theend. a site-specific routegroup is used. This implementation model was the
only oneavailable before Cisco Unilied Communications Manager version 7.
With Cisco Unified Communications Manager version 7, thelocal route group feature was
introduced. With local route groups, all sites that share the same PSTN dial rules can use one
and the same route pattern (orsetof route patterns). Theroute pattern (orset of route patterns)
isput into a systemwide route list, and this route list includes the local route group. Atthe
device pool of thecalling device, oneof theconfigured route groups is configured to bethe
Standard Local Route Group for thiscaller. Inthismodel, the routegroupthat is used is
determined bythedevice pool of thecalling device and notby its CSS. Thelocal route group
feature simplifies dial plans because iteliminates the need for duplicate CSS, partitions, route
patterns, and route lists. Since local route groups have been introduced, they arethepreferred
method for local gateway selection.
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Configuring IP Phones to Use Local PSTN Gateway
Ina multisite deployment, there typically aremultiple PS'fN gateways (usually one persite).
Selective PSTN breakout ensures that local gateways are used to access the PSI'N.
Configuring IP Phones to Use Local
PSTN Gateway
IP phones located in the main site use the main-site gateway for
PSTN access. Remote phones use their local gateway:
• Single 9.@ route pattern.
• Route list configured to use local route group
• Phone device pool configured with route group to be used by phone.
- Mamsite device pool and remote site device pool refer to differentroute groups.
PSTN
Main Site
470 555Cisco Unified
1234
Remote Site
Communications
Manager
From a dial plan perspecti\ c.you create one 9Ar route pattern (assuming that the North
American Numbering Plan [NAN PI is used). This route pattern is in a partition that is part of a
global CSS that is used by all phones, fhe route pattern refers lo asystemwide route list that is
configured to use the local route group. At the site-specific device pools, the standard local
route group isset to the route group thai includes the site-specific gateway.
In the example, there would be a device pool for the main site and a device pool for the remotesite. There would bea main site route group, including themain site gateway, and a remote site
route group, including the remote site gateway. IP phones at the main site and at the remote site
can now be configured with the same CSS. They all will match the same route pattern and
hence will use the same roule list. Based on the local route group feature, however, they will
alwavs use their local PSTN gateway for PSTN breakout.
Note
The local route group is configured with NANP PreDot digit stripping, by default. If the H.323
gateway expects calls that arereceived from Cisco Unified Communications Manager and
that would be routed to the PSTN to includethe PSTN prefix 9, appropriate digit
manipulation has to beconfigured in Cisco Unified Communications Manager. In this case,
the best solutionwould be to configure the called-partytransformation patterns and apply
gateway-specific called-party transformation CSS atthe gateways.
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Implementing PSTN Backup for On-Net Intersite
Calls
"Ihis topic describes how to use the PSTN as a backup for on-net intersite calls.
PSTN Backup
If the IP WAN (ICT) fails, calls are rerouted over the PSTN:
• 8 (site code).XXXX route pattern per site.
• Route pattern points to route lists (first option: ICT; second option:
Local Route Group).
Remote Site
Main Site
Cisco Unified
Communications
Manager
1001-1099
1001-1099
The figure shows a multisite deployment with two sites. Each site has its own Cisco Unified
Communications Manager cluster. Intersite calls should use the intercluster trunk (ICT) over
the IP WAN. However, what if the IP WAN is down? Since both sites have access to the
PSTN, the PS'fN should be used as a backup for intersite calls.
To ensure that phones at different sites always use their local gateway for PSTN backup, a
route list is configured that includes the ICT as the first option and the local route group as the
second option. This way. there is no need to have multiple, site-specific route lists with a
difTerent. site-specific route group as second entry.
) 2010 Cisco Systems. Inc.
Implementing Multisite Deployments
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Digit Manipulation Requirements for PSTN Backup of On-Net
Intersite Calls
PSTN backup for on-net calls can be easily provided by route lists and route groups that give
priority lo the intercluster trunk instead of the PSTN gateway.
Digit Manipulation Requirements for
PSTN Backup of On-Net intersite Cal
Digit manipulation is implemented differently
depending on the selected path.
Incoming Cisco Unifed Communications Manager
or gateway translates calling number of PSTN
gateway of other site to internal number (access
Outgoing Cisco Unified Communications
Manager transforms internal called number
and site code followed by directory number of
caller |if DID used] or attendant);
(access and site code plus directory number)
Id PSTN number Cisco Unified
Cisco Unified Communcations Manager or
Communications Manager Iransterms calling
gateway strips PSTN prefix from called number
number to PSTN number.
ICT First Cho.ce
Call from
Call from
1001 to 1001
8111 1001 to
1001
Incoming Cisco Unified
Outgoing Cisco Unified
Communications Manager
sSips access code and sue
aofls access code and site
code from called number
code to calling number
Communications Manager
When you areusing PSTN backup for on-net calls,you must address internal versus external
dialing. While on-netcalls usual!) use site codes and directory numbers, calls that arc sent
through the PS'fN haveto use PSI'N fonnal. Digit manipulation requirements vary depending
on the path that i> taken for the call:
•
Digit manipulation requirements when you use the ICT{first choice in route listand route
group):
•
—
At the calling site: The accessand site codes are removed from called number.
—
At the receiving site: The access and site codes are added to the calling number.
Digit manipulation requirements when you use the PS'fN (secondary choice in roule list
and route group):
—
At the calling site: The internal called number, which comprises an accesscode,
site code, and director} number, is transformed to the PSI'N numberof the called
phone. The calling number is transformed to the PSTN number of the calling phone.
Note
IfDID is not supported, the PSTN number of the site, rather than the PSTN number of the IP
phone, is used in called number and calling number.
When difTerent digit manipulation configuration is required depending on the
selected path, thedigit manipulation settings areeither eonligured at a path-specific
route group or by using global transformations.
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At the receiving site: The PSTN callingnumberis recognized as a PSTN numberof
an on-net connected site and transformed to the internal number: access and site
code, followed by the directory numberof the callingphone (if DID is used at the
calling site)or of the attendant of the calling site (if DID is not used at the calling
site), The callednumberis transformed to an internal directory numberand routed to
the IP phone (if DID is used at the receivingsite) or to an attendant(if DID is not
used at the receiving site).
)2010 Cisco Systems, Inc
ImplementingMultisite Deployments
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Implementing TEHO
This topic describes how to implement TKMO in a multisite environment.
Implementing TEHO
PSTN breakout occurs at gateway located closest to the PSTN
destination.
* Route partem for each area that can be reached at different costs, one per
site, indifferent partition
• Route patterns point to route lists (with different priorities of gateways;
cheapest gateway first, local gateway next).
• Phone CSS or local route group for correct route-pattern selection.
.470
406 555-
I'.
. RSIN„
DID XXXX L'
,-
""t470 555-1234
Remote
Cisco Unified
Phones
Communications
Manager
Local Path
When you implement 11:HO. PSTN breakout occurs at the gateway that is closest to the dialed
PS 1N destination. Basically, this action occurs because you create a route pattern for each
destination area that can be reached at dilTerent costs, 'fhese route patterns refer to route lists
that include a route group for the TF.HO gateway first and the local route group as the second
entry so that the local gateway can be used as a backupwhen the IP WAN cannot be used.
Note
The use of TEHO might not be permitted in your country or by your provider. There can also
be issues with emergency calls Therefore, ensure that your planned deployment complies
with legal requirements
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Considerations for Using Remote PSTN Gateways
When using backup TEHO. you have to consider several potential issues.
Considerations When Using Remote
PSTN Gateways
What should the calling number look like?
• Using PSTN number of originating site at TEHO gateway:
- Keeps standard numbering to the outside.
- Callbacks from the PSTN are possible.
- Legal issue in some countries or not permitted by provider.
- There are issues with emergency dialing.
• Replacing PSTN number of originating site by PSTN number of
TEHO site;
- Confuses called party.
- Called party might always use signaled calling number in the
future to reach calling party.
- If DID is supported on only one gateway, IVR script at the
gateway without DID has to know how to route calls to directory
numbers of all sites.
- There are issues with emergency dialing.
The first thing to consider when you are using TEi 10 is what number you want to use for the
calling number of the outgoing call. Basically, there are two options for configuring the calling
number for the outgoing call:
•
Use the PSTN number of the originating site at the TEHO gateway: When using the
PSTN number of the originating device for the caller ID of a TEHO call, the called party is
not aware that TEHO has been used. Standard numbering is maintained tor all PSTN calls,
regardless of the egress gateway; callbacks to the calling number are possible. Ilowcver,
sending calls lo the PSTN with PSTN caller IDs of other sites may not be permitted, or the
receiving PSTN provider may remove caller IDs from the signaling messages.
Caution
•
Sending calls out of a gateway with the calling number of another site might not be permitted
in your country or by your provider. There can also be issues with emergency calls.
Therefore, ensure that your planned deployment complies with legal requirements.
Replace the PSTN number of the originating site by the PSTN number of the TEHO
site: Whenusing the callingnumberof the backup gateway, called partiesmay get
confused about the number that should be used when calling back. For instance, they may
updatetheir address books with the differentnumberand inadvertently end up sendingcalls
to the TEHO site every lime ihey call. Further, DID ranges would have to include remote
phones or IVR scripts (automated attendants) to be able to route calls to phones located in
any site, regardless of where the PSTN call was received.
Caution
) 2010 Cisco Systems, Inc.
Using a remote gateway for PSTN access mightnot be permittedin your country or by your
provider. There can also be issues with emergency calls. Therefore, ensure that your
planned deployment complies with legal requirements.
Implementing Multisite Deployments
1-125
In general, it is highly recommended that you use the local route group feature when
implementing TEHO. In order to provide a local backup for TEHO calls, call processing must
route all calls ditTerentlv. based on the source (physical location) and on the dialed number,
when the TEHO path cannot be used. When you are not using local route groups, this approach
can require a huge amount of route patterns, partitions. CSS. and route lists, resulting in
complex dial plans. Suchdial plans are difficult to maintain and troubleshoot.
Note
You must also consider Call Admission Control (CAC) when implementing TEHO. When the
primary (TEHO) path is not admitted, the local gateway should be used instead. More
information about CAC is provided in a separate module of this course.
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TEHO Example Without Local Route Groups
This subtopic illustrates the dial plan requirements when you are implementing TEHO without
local route groups.
TEHO Example Without Local Route
Groups
Eacfi site has a different mute pattern and
route ist tor each ol the other sites.
Each site has one additional route pattern
ana route list for generic PSTN access (to
T1T
non-TEHO PSTN destnations|.
LI
=j_
In the example, there are five sites in a centralized call-processing deployment. Each site uses
identical call-routing policies and numbering plans, but the site-specific details of those policies
prevent customers from provisioning a single set of route pattern and route list that works for
all sites. This principle applies when no local route groups are used (as it was the case before
Cisco Unified Communications Manager Version 7).
Although the primary path for a given TEHO PSTN destination is always the same (the
appropriate TEHO gateway), the backup path is different for each site (the local gateway of the
site where the call has been placed). Without a backup path, TEI10 would require only one
route pattern per TEHO destination number and would refer to only the corresponding TEHO
gateway from its route list and route group. However, as the IP WAN is used for TEHO calls, it
is not recommended that you configure a single path only. Therefore, TEHO configurations
easily end up in huge dial plans: Each site requires a different route pattern and route list for
each of the other sites. In addition, each site has one generic route pattern for non-TEHO PSTN
destinations (using the local gateway).
Note
Some route patterns in the figure include the character"." multiple times (for example,
9.1 703.XXX.XXXX). In this case, theV character is used to illustrate the different
components of the number patterns in order to make it easier to interpret the patterns. In
reality, the"." in route patterns is used only once when being referenced by a corresponding
DDI, for example the PreDot DDI.
i 2010 Cisco Systems, Inc.
Implementing Multisite Deployments
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TEHO Example Without Local Route
Groups (Cont.)
• For each TEHO route pattern, the first entry in the route list is the
TEHO location: the second entry is the local gateway.
• For non-TEHO destinations, there is only one entry (local gateway)
in the route list
* Forn sites, n ' n route patterns and route lists are required.
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In the example, the configuration for one site (Boulder) is illustrated. There is a TEHO route
pattern for area code 703 (Hemdon) that refers to the route list RE-Bldr-I Irdn. This route lists
uses the Hemdon gateway first and the (local) Boulder gateway as a backup. There is also a
route pattern for area code 972 (Richardson), again using a dedicated route list for calls from
Boulder to Richardson (with the Richardson gateway preferred over the local Boulder
gateway). 'fhere are tuo more such constructs for the other two sites. Finally, there is a generic
PSI'N route pattern (9/d) for all other PSTN (that is. non-TEHO) calls. The generic PSTN
route pattern refers to a route list that contains only the local gateway. All five route patterns
are in the Boulder partition (P-Bldr) so that they can be accessed only by Boulder phones
(using the Boulder CSS "CSS-Bldr").
In summan. for each TEHO destination there is a route pattern per originating site that refers to
a dedicated route list utilizing the appropriate TEHO gateway before the local gateway. For n
sites, there are n * (n - \) of these patterns. In addition, each site has a generic route pattern
referring lo a dedicated route list containing the local gateway only. This generic route pattern
increases the total number of route patterns and route lists to n * n. In large TEHO
deployments, this approach does not scale.
Note
Some route patterns in the figure include the character"" multiple times (for example,
9.1.703 XXX XXXX) In this case, the "." character is used to illustrate the different
components of the number patterns in order to make it easier to interpret the patterns In
reality, the "." m route patterns is used only once when being referenced by a corresponding
DDI, for example the PreDot DDI
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) 2010 Cisco Systems. Inc.
TEHO Example with Local Route Groups
When implementing TEHO with local route groups, youcanreduce thenumber of route
patterns and route lists from n * n lo n + I.
TEHO Example with Local Route
Groups
Caling
Search
Spaces
Bouklo.
Route
Roule
Patterns
Lists
>.i*<a.xxxjatxi{'*i'Hn,
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This reduction is possible because, for each TEHO destination, one route pattern is sufficient.
The route pattern refers to a destination-specific route list, which lists the route group
containing the TEHO gateway first, followed by the entry "Default Local Route Group."
Because the backup path is now determined by the device pool of the calling device instead of
being explicitly listed in the route list, the route list has a generic format and can be used by all
sites.
For every TEHO destination, one route pattern and one route list is required. In addition, for
non-TFHO destinations, again, a single route pattern and route list can be utilized by all sites.
This route pattern (9.@) refers to a route list, which includes the "Default Local Route Group"
entry.
Note
Some route patterns in the figure include the character V multiple times (for example,
9.1.703.XXX.XXXX). In this case, the "." character is used to illustrate the different
components of the number patterns in order to make it easier to interpret the patterns. In
reality, the "." in route patterns is used only once when being referenced by a corresponding
DDI, for example the PreDot DDI.
) 2010 Cisco Systems. Inc.
implementing Multisite Deployments
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TEHO Example with Local Route
Groups (Cont.)
Dial plan simplification with local route groups:
• The number of route patterns and route lists for TEHO
destinations is reduced from n* (n- 1) ton (20to5 in this
example with 5 sites).
* The number of route patterns and route lists for non-TEHO
destinations is reduced from n to 1 (5 to 1 in ihis example).
* The total number of route patterns and route lists is reduced
from n " n to n + 1 (25 to 6 in this example).
* The number of partitions and CSS is reduced from n to 1 (5
to 1 in this example).
The number of gateways, route groups, and device pools
remains the same (n).
In the example (five sites), using local roule groups simpliiies the dial plan that is described
here:
•
•
The number of route patterns and route lists for TEHO destinations is reduced from
n * In - 1) to n. In the example, the reduction is from 20 to 5.
fhe number of route patterns and route lists for non-TEHO destinations is reduced from
n to 1 (5 to 1 in this example).
•
Thus, the total number of route patterns and route lists is reduced from n * n to n + 1
(25 to 6).
•
The number of partitions and CSS is reduced from n to I (5 to 1).
The number of gateways. route groups, and de\ ice pools remains the same: ti.
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Implementing Globalized Call Routing
This topic describes how globalized call routing is implemented and how it simplifies
international multisite dial plans.
Globalized Call-Routing Characteristics
Allows one format to be used for routing to PSTN
destinations.
E.164 format with + prefix is used.
Users can still type PSTN numbers in local format.
Localized input is globalized at call ingress.
Routing is based on numbers in + format.
At call egress, numbers are localized depending on the
egress device.
International dial plans are substantially simplified.
With globalized call routing, all calls that involve extemal parties are based on one format. All
numbers are normalized as follows:
•
Normalized called-party numbers
E.164 format with the + prefix is used for external destinations. Therefore, called-number
normalization is the resultof globalization. Internal directory numbers are used for internal
destinations. Normalization is achievedby strippingor translating the callednumberin
internally used directory numbers.
•
Normalized calling-party numbers
E.164 global format is used for all calling-party numbers, except calls that are from an
internal number to another internal number. Such purely internal calls use the internal
directory' number for the calling party number.
If sources of calls (users at phones, incoming PSTN calls at gateways, calls received through
trunks, and so on) do not use the normalized format, the localized call ingress must be
normalized before being routed, fhis requirement applies to all received calls(coming from
gateways and trunks, as well as from phones), and it appliesto both the calling-and calledparty numbers.
Note
>2010 Cisco Systems, Inc.
Except for the internal calls that were mentioned (where the destination is a directory
number and, inthe case of an internal source, the source is a directorynumber), all
numbers are normalized to the E.164 global format. Therefore, call routing that is based on
the normalized numbers is referred to as globalized call routing.
Implementing Multisite Deployments
1-131
After the call has been routed and path selection (if applicable) has been perfonned. the egress
device typically must change the nomialized numbers lo the local fonnat. fhis situation is
referred to as localized call egress.
focalized call egress applies to these situations:
•
Calling- and called-party numbers for calls that arc routed to gateways and trunks: If
the PSTN or the telephony system on the other side of a trunk docs not support globalized
call routing, the called- and calling-party numbers must be localized from the global
format. For example, the called-party number +494012345 would have to be changed to
011494012345 before the call could be sent out lo the PSTN in (he United States.
•
Calling-party numbers for calls that are routed from gateways or trunks to phones:
This situation applies lo the phone user who does not want to sec caller IDs in a global
format. For example, ifa user at a U.S. phone wants to seethe numbers of PSTN callers
who arc in the same area code, that user may want to see each number as a seven-digit
number and not in the+1 XXXXXXXXXX fonnat.
Localized call egress is not needed for the called-party number of calls that arc routed to
phones, because internal directory numbers are the standard (normalized) fonnat for internal
destinations (regardless of the source of the call), fhese numbers might have been dialed
differenth initialK. however: in that case, this localized call ingress was nomialized before call
routing.
Localized call egress is also not required for the calling-party numberof internal calls (internal
to internal) because, again, the standard for the calling-party number of such calls is to use
internal director, numbers.
Globalized call routing simplifies international dial plans because the corecall-routing decision
is always based on the same fonnat. regardless of how the number was initially dialed and
regardless of how the number looks at the egress device.
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Globalized Call Routing: Number Formats
This subtopicdescribes the numberformats that are used by globalized call routingand
explains some commonly used expressions.
Globalized Call Routing: Number
Formats
Call Ingress
Call Routing
Call Egress
Calling Localized E 164
Called Localized E.164
To External
Calling. Directory Number
Called According lo Loca
From Extemal:
Dal Rules (PSTN Access
Calling: Localized E.164
Codes. International
Called: Directory Number
Access Codes, Etc ) or
E 164 (Click to Dial, Call
Lists, Etc
From Internal: ^^Sg
Calling Directory Number
Called: Directory Number
To Interna
Calling Directory
Called. Directory
The table explains expressions that are commonly used to describe globalized call routing. The
table refers to the figure.
Term
Number
normalization
Description
The process of changing numbers to a well-defined, standardized
(normalized) format. In this case, all external phone numbers are
changed to global E.164 format.
Number
globalization
The process of changing numbers to global E.164 format.
Exampte: Because the normalized format is global E.164 (see
"Number Normalization"), you normalize a called number (for
example, 4085551234) by globalizing the number—that is, by
changing the number to global format (for example, +14085551234).
Number
The process of changing from normalized format (in this case, global
localization
format) to local format. Usually, the local format is the shortest
possible format that does not conceal relevant information. An
example of local format is 555-1234 instead of +1 408 555-1234, or
972 333-4444 instead of +1 972 333-4444 (assuming that the device
where localization occurs is located in +1408 area).
Incoming PSTN
Call from PSTN to internal phone. Like all calls, such a call consists
call
of two call legs (incoming and outgoing). See also "Call ingress" and
"Call egress." On an incoming PSTN call, the incoming call leg (call
ingress) is PSTN gateway to Cisco Unified Communications
Manager; the outgoing call leg (call egress) is Cisco Unified
Communications Manager to internal phone.
) 2010 Cisco Systems, Inc
Implementing Multisite Deployments
1-133
Term
Description
Outgoing PSTN
Call from internal phone to PSTN Like all calls, such a call consists
of two call legs (incoming and outgoing). See also "Call ingress" and
call
"Call egress." On an outgoing PSTN call, the incoming call leg (call
ingress) is internal phone to Cisco Unified Communications Manager;
the outgoing call leg (call egress) is Cisco Unified Communications
Manager to PSTN gateway.
Call ingress
Incoming call leg—call received by Cisco Unified Communications
Manager
Call egress
Outgoing call leg—call routed to destination by Cisco Unified
Communications Manager.
Localized
PSTN number in partial (subscriber, national, international) E.164
E 164 (number)
format. See "Number localization."
E.164 (number]
PSTN number in complete E.164 format with + prefix.
On the left side of the figure, call ingress is illustrated by two types of call sources:
•
External callers: Their calls are received by Cisco Unified Communications Manager
through a gateway or trunk. In the case of a PSTN gateway, calling- and called-party
numbers are usually provided in localized E.164 format.
•
Internal callers: Their calls are received from internal phones, in the case of calls to
internal destinations (for example, phone to phone), calling- and called-party numbers are
typically provided as internal directory numbers. In the case of calls to external destinations
(for example, phone to PS'fN). the calling number is the directory number (at call ingress
time) and the called number depends on the local dial rules for PS 1N access. These dial
rules can differ significantly for each location.
The center of the figure illustrates the standards that are detlned for normalized call routing. As
mentioned earlier, because most calls use global E.164 format, this type of call routing is also
referred to as globalized call routing. Ilere are the defined standards:
•
•
•
External to internal:
—
Calling-party number: h. 164
—
Called-party number: directorv' number
External to external (if applicable):
—
Calling-party number: F.164
—
Called-party number: E.164
Internal to internal:
-
Calling-party number: directorv number
Called-party number: directory number
•
1-134
Internal to external:
—
Calling-party number: E.I64
—
Called-party number: E.164
ImplementingCisco Unified Communications Manager, Part 2 (CIPT2)v8.0
© 2010 Cisco Systems. Inc
At the right side of the figure, call egress is illustrated by two types of call targets:
•
Gateways: When sending calls to the PSTN, localized E.164 format is used for both the
calling- and called-party numbers. The format of these numbers (especially of the calledparty number) can significantly differ based on the location of the gateway. For example,
the international access code in the United States is 011, and in most European countries, it
is 00.
•
Phones: When a call from an internal phone is sent to another internal phone, the call
should be received at the phone with both the calling and called number using internal
directory numbers. Because this format is the same format that is used by globalized call
routing, there is no need for localized call egress in this case. When a call from an external
caller is sent to an internal phone,most users (especially users in the United States)prefer
to see the calling number in localized format (for example, national and local calls should
be displayedwith 10digits). The called numberis the directorynumberand usually is not
displayed on the phone.
It is evident from the figure that there are several situations where the numbers that are
provided at call ingress do not conform to the normalized format to be used forcall routing.
These situations applyalso to call egress, wherethe normalized format is not always used when
the call is delivered. Therefore, localized call ingresshas to be normalized (that is. globalized)
and globalized fonnat has to be localizedat call egress.
)2010 Cisco Systems, Inc.
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Normalization of Localized Call Ingress on Gateways
The figure illustrates how localized call ingress on gateways getsnormalized.
Normalization of Localized Call Ing
on Gateways
Cisco Unified
Communicalions
Called Number
Manager
Globalized
Call Routing
GW gateway DP device pool SP service parameter
Ilere are the requirements for normalizing localized call ingress on gateways:
•
Changing the calling number from localized E.164 fonnat to global E.164 fonnat
•
Changing the called number from localized E.164 fonnat to directory numbers for calls to
•
Changing the called number from localized E.164 formal to global E.164 fonnat for calls to
internal destinations
external destinations (if applicable)
As shovsn in the figure, the calling number canbe normalized by incoming calling-party
settings. Thev are configured at the gateway oralthe device pool, orthey can be configured as
CiscoUnified Communications Manager serviceparameters. The figure provides an example
for a gateway in San Jose:
•
Prefix for incoming called-party numbers with number typesubscriber: +1408
•
Prefix for incoming called-party numbers with number type national: +1
•
Prefix for incoming called-party numbers with number type international: +
The called numbercan be normalized bv significant digits that are configured al the gatewav
(applicable onlv ifno calls lo other external destinations are permitted and a fixed-length
number plan is used), or bv translation patterns, orby incoming called-party settings (if
available at the ingress dev ice). In the example, the gateway is configured with four significant
digits.
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Normalization of Localized Call Ingress from Phones
The figure illustrates how localized call ingress on phones gets normalized.
Normalization of Localized Call
from Phones
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Caled According iq Local
Call)ng:E.l64
Dial Rules (PSTN Access
Called: E.164
Globalized
CallRoutng
Codes. International Access
Codes. Etc
£,<3niple if'hooes ir= HarriDu'c,}
To Internal.
C0O'-> DDI Pr~DM Prefi* ->•
Calling. Directory Number
DO •-:> DDIPieDoi Prefi* +49
Called Directory Nimber
Here are the requirements for normalizing localized call ingress on phones:
•
For calls to external destinations: Changing the calling number from an internal director}
number to FT 64 fonnat. Changing the called number to K.164 format if any other format
•
For calls to internal destinations: No normalization is required.
was used (according to local dial rules).
As shown in the example, you can normalize the calling-party number for calls to external
destinations by configuring an extemal phone number mask (in E.164 format) at the phone.
You can normalize the called-party number by using translation patterns where you would also
apply the extemal phone number mask to the calling-party number. In the figure, examples for
phones that are located in Hamburg, Germany, and San Jose, California, are given.
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Localized Call Egress at Gateways
The figure illustrates how you can implement localized call egress at gateways.
Localized Call Egress at Gateways
Calling Localized E 164
Called. Localized E 164*
Localized Call Egress
Called Number
Cisco Unified
Communications
Called Parly
Transformation CSS
(GW. DP)
Manager
Globalized
Call Routing
Calling Number
Calling Party
Trans form a lion CSS
(GW. OP)
GW gateway. DP device pool
The only requirement is to change the calling and called number from global F.164 format to
localized F., 164 fonnat.
You can change the format by configuring called- and calling-party transformation patterns,
puttingthem into partitions, and assigning the appropriate called-and calling-party
transformation CSS to gateways. You can configure called- and calling-party transformation
CSS at the de\ice (gateway or trunk) and al the device pool.
The tables that are presented in this section refer to the example that is provided by the figure.
Hie first table shows the configuration ofthe called-party transfoniiation patterns that are
applicable to the SanJosegateway (based on partition andcalled-party transformation CSS).
Transformation Pattern
Performed Transformation
\+.!
DDI PreDot, Prefix 011
\+.1 XXXXXXXXXX
DDI PreDot
\+1408.XXXXXXX
DDI PreDot
Note
In this example, the San Jose gateway does not use number types. Therefore, 011 has to
be prefixed on international calls, and the 1 of national calls is conserved. Forlocal calls,
only the last seven digits are used.
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The next table shows how you would configure the called-party transformation patterns that are
applicable to a gateway in Hamburg, Germany (based on partition and called-party
transformation CSS).
Transformation Pattern
Performed Transformation
\+ !
DDI PreDot, number type: international
\+49.1
DDI PreDot, number type: national
\+4940 !
DDI PreDot. number type: subscriber
Note
In this example, the Hamburg gateway is using number types instead of international (00) or
national (0) access codes (in contrast to the San Jose gateway, which does not use number
types).
The next table shows how you would configure calling-party transformation patterns that are
applicable to the San Jose gateway (basedon partitionand calling-party transformation CSS).
Transformation Pattern
Perfonned Transformation
\+ i
DDI PreDot; number type: international
\+1. XXXXXXXXXX
DDI PreDot; number type: national
VH408.XXXXXXX
DDI PreDot; number type: subscriber
Note
In the example, subscriber, national, and international number types are used at the San
Jose gateway for the calling-party number. If no number types were used, due to the fixedlength numbering plan, the number type could also be determined by its length (seven-digit
numbers when the source of the call is local, 10-digit numbers when the source of the call is
national, or more than 10 digits when the source of the call is international). In reality,
however, countries that use the NANP typically use 10-digit caller IDs for both national and
local callers
Having nonlocal calling-party numbers implies the use of TEHO or PSTN backup over the IP
WAN. This scenario is not permitted in some countries or by some PSTN providers. Some
providers verify that the calling-party number on PSTN calls that they receive matches the
locally configured PSTN number. If a different PSTN number is set for the caller ID, eit.ier
the call is rejected or the calling-party number is removed or replaced by the locally
assigned PSTN number
The final table shows how you would configure calling-party transformation patterns that are
applicable to a gateway in Hamburg, Germany (based on partition and calling-party
transformation CSS).
Transformation Pattern
Performed Transformation
\+.!
DDI PreDot, number type: international
\+49.l
DDI PreDot, number type; national
\+4940.!
DDI PreDot, number type: subscriber
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Localized Call Egress at Phones
fhe figure illustrates how \ou can implement localized call egress at phones.
Localized Call Egress at Phones
Localzed Call Egress
The only requirement is that you change the calling number from global li. 164 fonnat to
localized F.164 fonnat.
You can change the fonnat by configuring calling-party transformation patterns, putting them
into partitions, and assigning the appropriate calling-parly transformation CSS to IP phones. As
mentioned earlier in this lesson, you can configure calling-party translomialion CSS at the
phone and at the de\ ice pool.
The two tables that are presented in this subtopic arc in reference to the example that is
provided by the figure. The first table shows how you would contigurc the calling-party
transformation patterns that are applicable to a phone that is located in San Jose (based on
partition and calling-part} transformation CSS).
Transformation Pattern
Performed Transformation
VM.XXXXXXXXXX
DDI PreDot
VH408XXXXXXX
DDI PreDot
Note
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In this example, international calls are shown in standard normalized format (E 164 format
with + prefix) because there is no W calling-party transformation pattern. National calls are
shown with 10-digit caller IDs, and local calls are shown with 7-digit caller IDs.
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The next table shows how you would configure the calling-party transformation patterns that
arc applicable to a phone that is located in Hamburg, Germany (basedon partitionand callingparty transformation CSS).
Transformation Pattern
Performed Transformation
\+49 !
DDI PreDot, Prefix 0
\+4940.l
DDI PreDot
Note
Because there is no \+! calling- party transformation pattern, international calls are
preserved in normalized format (E.164 with + prefix). As opposed to the San Jose example,
phones that are located in Hamburg do prefix the national access code (using 0, which is
equivalent to the long-distance 1 in the NANP). The reason is that, in Germany, variablelength PSTN numbering plans are used and therefore national and local numbers cannot be
distinguished based on their length (like in the United States, with 7- and 10-digit numbers).
When the national access code 0 is prefixed to numbers that are used by national callers, a
user can identify national calls by their leading 0.
Note
When users call back PSTN callers, the globalized number is used for the outgoing call.
Therefore, there is no need to edit the localized number from a call list and add PSTN
access codes and national or international access codes.
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Globalized Call-Routing Example: Emergency Dialing
In a multisite deployment with centralized call processing, it might be desirable to simplify
emergency dialing b\ introducing a globalized emergency number (or one globalized
emergency number for each emergency service).
Globalized Call-Routing Example:
Emergency Dialing
Note local emergency numbers (for example, 112 in the European Union,
999 in the United Kingdom. 000 in Australia).
Introduce a corporate emergency number (for example, 888) that can be
used at all sites (globalized emergency number).
Allow local emergency numbers in addition to corporate
emergency number:
Per site. Only local emergency number is permitted (for example, 112
can be used m the European Union, 999 can be used only in the United
Kngdom. and soon|.
Globally All local emergency numbers are permitted at all sites (roaming
users can use home emergency number).
Local emergency numbers are globalized to corporate emergency number
(for example, 888) at call ingress by translation patterns.
Corporate emergency number is localized at egress gateway by called-party
transformation CSS (for example, 868 is changed to 112 at EU gateway, to
999 at UK gateway, and to 000 at Australian gateway].
Having a globalized emergency numberallowsroamingusers who might not be awareof the
local emergencv dial rulesto use a corporate emergency number that is accessible from all
sites.
In addition, however, localized emergency dialing should still be supported, so that a user can
dial eitherthe locally rele\ ant emergency numberor the corporate emergency number.
Here is how to implement such a solution;
•
•
You introduce one or more corporate emergency numbers.
In addition. \ou allow localized emergency dialing. It can he limited to local emergency
dialing rules persite (for example, an Austrian emergency number can be dialed only from
phones that are located in Austria), or you can globally enable all possible local emergency
numbers. Having all possible local emergency numbers thatare globally enabled would
allow a roaming user to use the emergency number that is local to thesite where the user is
located, or theemergency number thatthe userknows from the home location of the user
(forexample, a UK userdials 999 while roaming in Austria), or the corporate emergency
number.
•
Ifa user dials a localized emergency number, that numberis first normalized (that is.
translated) to the corporate emergency number. A route pattern exists only forthis
corporate emergency number, and you configure the corresponditig route listto use the
local route group.
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•
At the gateway that is used to process the call, you localize the corporate emergency
number (the globalized emergency number) by using called-party number transfonnations
at the gateway. This localization ensures that, regardless of which emergency number was
dialed, the gateway that sends out the emergency call uses the correct number as expected
at this site.
Note
In deployments with more complex emergency calls, like in the United States with E911,
such a solution is not applicable because there are other requirements for emergency calls.
In such a scenario, the emergency call is routed via a dedicated appliance (Cisco
Emergency Responder) that is reached via a computer telephony integration (CTI) route
point.
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Globalized Call-Routing Example:
Emergency Dialing (Cont.)
Called-Party
Route
000
Transformation
Pattern
Australian User in E U
3 Translation
112
Patlems 000
112 999
Cailed-Party
Transformation
Route List
/
/
f
Syslem
*
nn
-' :'
Mask BBS
Route Lisl
Syslem
Roule
Group
Default
• 33
Called-Party
Transformation
UK Gateway
8B8 ^ 999
LRG
Any User in UK
In the example, a corporate emergency number of 888 has been established. In addition.
Australian. E.U.. and UK emergene; numbers are supported at all sites oflhe enterprise. The
appropriate numbers (000. 112. and 999) are translated (normalized) to the corporate (global)
emergency number 888. A route pattern888 exists, which refersto a roule lisl that has been
configured to use the local routegroup. You will considertwo sites in this example: one in the
European Union and one in the United Kingdom. Hach site has its own PSTN gateway (or
"GW" in the figure): phonesat each site are configured with a site-specific device pool. The
de\ ice pool of each site has its local routegroup that is set to a site-specilic route group.
You will examine fouremergenc} calls:
•
A IK user dials 999 (IK emergency number):
The dialed UK emergency number999 is translated lo the corporate emergency number
888. After translation, the 888 route pattern is matched. The route list of the route pattern
refers to the local routegroup. Because the emergency call was placed from a UK phone,
the local routegroup in the devicepool of the phone refers to the UK gateway. At thai
gateway, a global transformation of the called number (from 888 to 999) is configured.
Therefore, the call exits the UK gateway with a destination numberof 999, which is the
appropriate emergency number to be used in the United Kingdom.
•
Any user who is located in the United Kingdom dials888 (corporate emergency
number):
Because no local emergency number was dialed exceptthe corporate emergency number
888. no translation is required. The call immediately matches routepattern 888. The route
listof the route pattern refers to the local route group. Because theemergency call was
placed from a UK phone, the local route group in the device pool of the phone refers to the
UKgate\sa\. Atthat galewa\. a global transformation of the called number (from 888 to
999) isconfigured, "fherefore. the call exits the UK gateway with a destination number of
999. uhich is the appropriate emergency numberto be used in the United Kingdom.
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•
An E.l. user dials 112 (E.U. emergency number):
The dialed E.U. emergency number 112 is translated to the corporate emergency number
888. After translation, the 888 route pattern is matched. The route list of the route pattern
refers to the local route group. Because the emergency call was placed from an E.U. phone,
the local route group in the device pool of the phone refers to the E.U. gateway. At that
gateway, a global transformation of the called number (from 888 to 112) is configured.
Therefore, the call exits the E.U. gateway with a destination number of 112, which is the
appropriate emergency number to be used in the E.U.
•
An Australian user, currently located at an E.U. site, dials 000 (Australian emergency
number):The dialed Australian emergency number 000 is translated to the corporate
emergency number 888. After translation, the 888 route pattern is matched. The route list
of the route pattern refers to the local route group. Because the emergency call was placed
from an E.U. phone, the local route group in the device pool of the phone refers to the E.U.
gateway. At that gateway, a global transformation of the called number (from 888 to 112)
is configured. Therefore, the call exits the E.U. gateway with a destination number of 112,
which is the emergency number in the European Union.
Note
The Australian user can use an E.U. phone (with an E.U. extension), or use their own device
with device mobility enabled, or use an E.U. phone with their own extension (by using Cisco
Extension Mobility). In all three scenarios, the emergency call would work fine as described
earlier. The reason is that the device pool of the phone will be the E.U. device pool in all
three scenarios (with device mobility enabled, the home device pool would be replaced by
the roaming device pool), and hence the local route group is always the EU-GW.
The only problem would be if the Australian user were using their own device with device
mobility disabled. In this case, the local route group would refer to the Australian gateway,
and therefore the call would be sent through the Australian gateway instead of through the
local E.U. gateway. The localized egress number would be appropriate for an Australian
gateway (transformed to 000), so that the user would get connected to an Australian
emergency service.
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Considering Globalized Call-Routing
Interdependencies
This topic describes how globalized call routing interacts with other dial plan features.
Globalized Call-Routing Interaction
with
Other Dial Plan Features
Globalized call routing simplifies the implementation
of several dial plan features in an international
environment:
•TEHO
•AAR
• SRST or CFUR
• Cisco Device Mobility
• Cisco Extension Mobility
Globalized call routingsimplifies the implementation of severaldial plan features in an
international deployment, fhe affected dial planfeatures include TEHO, automated alternate
routing (AAR). Cisco Unified SRST andCEUR. Cisco Device Mobility, andCisco Extension
Mobility.
If TEHO is configured, the appropriate TEHO gateway is used for the PSTN call. The TEHO
route list can include the Default Local Route Groupsettingas a backup path. In this case, if
the priman. (TEHO) path is nota\ailable.the gateway that is referenced by the local route
group of the applicable device pool will be used for the backup path. I(The device pool
selection is not static, but Cisco Unified Device Mobility is used, the gateway olThe roaming
site will be used as a backup for the TEHO path.
Thesame situation applies lo Cisco Extension Mobility. When a user roams to another site and
logs in to a local phone. PSTN calls will use the local gateway (ifTEHO isnot configured) or
the local gateway uill be used as a backup (if TEHO is configured). The local gateway
selection is not based on the Cisco Extension Mobility user profile, but on the device pool ot
the phone where the user logs in. fhe line CSS. however, isassociated with the user profile,
and therefore the user can dial PSTN numbers the same way that the user does at home, fhe
localized input is then globalized. After call routing and path selection occur, the globalized
number is localized again based ontherequirements of theselected egress device. The
localized input fonnat that theuser used can be completely different from the localized format
that is used at call egress.
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Globalized Call Routing—TEHO Advantages
fhis subtopic reviews the advantages of using globalized call routing in an international dial
plan that uses TEHO.
Globalized Cal! Routing—TEHO
Advantages
• With local route groups, there is no need to have duplicate
TEHO patterns for each originating site.
• Local PSTN backup is selected by local route group.
• Called number has to be in a format that can be manipulated
at primary path (TEHO gateway) and backup gateway.
• Ifgateways are in separate countries, having the called
number in globalized format is easier.
As discussed earlier,when you are using local route groups, there is no need to have duplicated
TEHO route patterns for each originating site. Instead, the local PS'fN gateway is selectedby
the local route group feature when the TEHO path cannot be used.
Whencombining globalized call routing with local route groups,you do not have to care about
the variouspossible input formats for the TEHO call-routing decision. No matter how the user
dialed the number, it is changed to globalized format before it is routed. Because the called
number is then localized after call routing and path selection, you can localize the called- and
calling-party number differently at the primary gateway (TEHO gateway) and the backup
gateway (localgateway). However, the global transformations that you configure for each
egress gateway all refer to a single format—a globalized format regardless of how the user
dialed the destination. This globalized format that is combined with local route groups for local
backup gateway selection, makes implementing TEHO much simpler. Without globalized call
routing, youwould haveto perform localization at the egress gateway differently foreach
originating site.
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Globalized Call Routing—TEHO Example
The figure shows an exampleof TEHO when globalized call routing is used.
Globalized Call Routing—TEHO
Example
EU. Gateway Called_Par!
000 1406-
US-
5551234
TEHO-
|
\* i -> PreDot,
Pattem
ISDN Number
Type1 International
E U User
900 14085551234
Globalization
• (Localized
Roule Ust
TEHO-U S
Ingress
(Three
UK Gateway
Called-Party
Roule Ust
-j^|&! Transformation
TEHO-U S
Translalion
9 5551234
Patterns)
Route
Groups
1)U S RG
2) Default
LRG
User n U S .Area
U.S. Gateway
Called-Party
.s3*2T*
Transformation
Code 408
Primary Path
Backup Path
At the call ingress side, there are three PSTN dial rules: E.U.. UK,and United StatesThe same
rules applv to the egress gateways: the E.U.. UK, and U.S. gateways all require dilTerent digit
manipulation when you are sending calls to the PSTN.
As long as users areallowed to roam between sites and TEHO with local backup is inplace,
users can dial each PSTN destination differently at each site. In addition, if the TEHO path is
notavailable, the local gateway (which again can be anyof the three) is used for backup. With
globalized call routing, vou do not have to consider all possible combinations ofingress and
egress, butyou consider call ingress and call egress independent of each other.
All that vou need to configure is translation patterns for each of the PSTN dial rules (E.U.. UK.
and United States), "fhen vou create TEI10 route patterns that refer to the TEHO gateway as the
first choice, and to the local gateway as the backup, using the local route group feature. At the
egress gatewavs. vou configure the called- and calling-party transfonnations, where you do not
match on all possible input formats again, buton a globalized format only.
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Summary
fhis topic summarizes the keypoints that werediscussed in this lesson.
Summary
• Multisite dial plans should support selective PSTN breakout
with backup gateways, PSTN backup for on-net calls, TEHO,
and intersite calls using access codes and site codes.
* With the addition of an access code and site code to directory
numbers at each site, directory numbers no longer have to be
globally unique.
• When calls are routed to the PSTN, calling directory numbers
have to be transformed to PSTN numbers, and access codes
used on dialed patterns have to be removed to ensure that
calling number and called number are in accordance with
PSTN numbering schemes.
• Selective PSTN breakout means that different gateways are
used for PSTN access, depending on the physical location of
the caller.
Summary (Cont.)
When the PSTN is used as a backup for intersite calls,
internal directory numbers and internally dialed patterns have
to be transformed to ensure that calling number and called
number are in accordance with PSTN numbering schemes.
When you implement TEHO, calls to the PSTN are routed
differently, based on the physical location of the caller and
the PSTN number that was dialed. This difference ensures
that the call uses the IP WAN as much as possible and
breaks out to the PSTN at the gateway that is closest to the
dialed PSTN destination.
Globalized call routing is a dial plan concept in which the call
routing is based on E. 164 numbers with a + prefix.
Globalized call routing reduces the complexity of dial plans
substantially and makes it easier to implement features such
as device mobility, extension mobility, AAR and CFUR, or
TEHO in international deployments.
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References
For additional information, refer to these resources:
•
Cisco Svstems. Inc. Cisco UnifiedCommunications System 8.x SRND, April 2010.
http:/.•''www.cisco.coni'en'US'VJoes'voice ip comm/cuem/srnd/Xx.'uc8x.html
•
Cisco Systems, Inc. Cisco Unified ( ommunications Manager Administration Guide
Release'8.0(1/. Eebruary 2010.
http:"'www cisco.com.civI'S'docs'voice ip comm/cuem/admin/8 0 |/ccmdg/hccin-80lcm.htinl
•
Cisco Svstems. Inc. Cisco IOS Voice Configuration Library (with Cisco IOS Release 15.0
updates). July 2007.
http:;\\v\vv.cisco.coni'cnT'S'doc.s/ios'!2 .Vvvf c'cisco ios voice configuration librarv £
lossarv/vcl.htm
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Module Summary
lliis topic summarizes the key points that were discussed in this module.
Module Summary
Specific issues apply to Cisco Unified Communications
Manager multisite deployments. These issues include
bandwidth consumption in the IP WAN, IP WAN reliance,
suboptimal call routing, and NAT or security issues.
Special solutions can solve issues that apply to multisite
deployments. These solutions include the use of various
codecs, SRST and MGCP fallback, selective PSTN breakout,
and the use of Cisco Unified Border Element.
Connection options for multisitedeployments include various
types of trunks and gateways.
A multisite dial plan should support site-code dialing, PSTN
backup, and TEHO. Globalized call routing simplifies
international multisite dial plans.
This module discussed the issues thatapply to Cisco Unified Communications Manager
multisite deployments and their possible solutions. Itdescribed thevarious connection options
for muitisite deployments and how they are implemented. Itthen described how to implement a
multisite dial planthat covers site-code dialing, public switched telephone network (PSTN)
backup, and tail-end hop-off (TEHO).
References
For additional information, refer to these resources:
•
Cisco Systems. Inc. Cisco Unified Communications System 8.xSRND, April 2010.
•
CiscoSystems, Inc. Cisco Unified Communications Manager Administration Guide
http:.'7'vvvvw.cisco.com/en/US/docs/voice_ip_c()mm/cucm/snid/8x/uc8x.htnil
Release 8.0(1). February 2010.
hitp:/.'vv-w-w.cisco.com/cn/US/docs/voice_ip_comm/cucm/adinin/8 0_l/ccmcfg/bccm-80lcm.html
•
Cisco Systems. Inc. Cisco IOS Voice Configuration Library (with Cisco IOS Release 15.0
updates). July 2007.
hup; 7ww w.cisco.com/'en/US/does/ios/l2..3/vvf_c/cisco. ios_voice...configuration librarv_g
lossaryvcl.htm
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Module Self-Check
Use the questions here to review whatyou learnedin this module. The correct answersand
solutions are found in the Module Self-Check Answer Key.
01)
Which of the following is not an issue in Cisco Unified Communications Manager
multisite deployments? (Source: Identifying Issues in a Multisite Deployment)
A)
D)
C)
D)
E)
Q2)
Which of these statements does not apply to IP networks? (Source: Identifying Issues
in a Multisite Deployment)
A)
B)
C)
D)
Q3)
A)
B)
C)
VoIP packets are large and sent at a high rate.
The Layer 3 overhead of a voice packet is not significant.
Voice packets have small payload size and are sent at high packet rates.
D)
Packetized voice has the same overhead as circuit-based voice.
In a multisite deployment, IP phone
and
packets are affected by WAN
failures. (Source: Identifying Issues in a Multisite Deployment)
A)
data, video
B)
signaling, data
C)
data, media
D)
signaling, media
Which two of the following are dial plan issues in multisite deployments? (Choose
two.) (Source: Identifying Issues in a Multisite Deployment)
A)
B)
C)
D)
F)
Q6)
tmm
IP packets can be delivered in incorrect order.
Buffering results in variable delays.
Tail drops result in constant delays.
Bandwidth is shared by multiple streams.
Which statement most accurately describes overhead for packetized voice? (Source:
Identifying Issues in a Multisite Deployment)
Q4)
Q5)
availability
quality
bandwidth
security
Call Admission Control
overlapping directory numbers
overlapping E.164 numbers
variable-length addressing
centralized call processing
centralized phone configuration
Which of these is a requirement for performing address translation for Cisco IP
phones? (Source: Identifying Issues in a Multisite Deployment)
A)
use DHCP instead of fixed IP addresses
B)
exchange media streams with the outside world
C)
use DNS insteadof hostnames in Cisco UnifiedCommunications Manager
D)
exchange signaling information with the outside world
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Q7)
Which of the following is not a solution for multisite environments? (Source:
Identifying Multisite Deployment Solutions)
A)
B)
C)
D)
Q8)
QoS
site codes
SRST
MGCP
When implementing QoS. how is the quality of voice streams provided? (Source:
Identifying Multisite Deployment Solutions)
Q9)
Which two statements are tme about bandwidth solutions in a multisite deployment?
(Choose two.) (Source: Identifying Multisite Deployment Solutions)
A)
B)
WAN bandwidth can be conserved by using low-bandwidth codecs within a
RTP-header compression compresses the RTP header to 2 bytes.
C)
WAN bandwidth can be conserved by deploying local media resources.
remote site.
D)
Voice compression is part of RTP-header compression.
E)
Multicast MOH from branch router flash totally eliminates the need to send
MOH over the WAN.
QIO)
Which tuo statements are tme about availability? (Choose two.) (Source: Identifying
Multisite Deployment Solutions)
A)
CFNB is required to enable main-site phones to call remote-site phones during
B)
C)
SRS'f provides a fallback for Cisco IP phones.
MGCP fallback allows the gateway to use local dial peers when the call agent
SRST fallback.
is not reachable.
D)
AAR is required to enable phones to reroute calls over the PS'fN when the IP
WAN is down.
F)
MGCP fallback and SRS'f cannot be implemented at the same device.
Ql 1) Whichof the following are notdial-plan solutions for multisite Cisco Unilied
Communications Manager deployments? (Choose two.) (Source: Identifying Multisite
Deplovment Solutions)
QI2)
1-154
A)
B)
C)
D)
access and site codes
TEHO
globalized call routing
shared lines
E)
overlap signaling
Which Cisco IOS feature provides signalingand media proxy functionality in order to
eliminate the need for NAT? (Source: Identifying Mullisite Deployment Solutions)
A)
B)
C)
Cisco Unilied Border Element in flow-through mode
Cisco PIX Firewall
Cisco Unitied IP-to-Proxy Gateway
D)
Cisco Unilied Border Element in flow-around mode
implementing Cisco Unified Communications Manager. Pari 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Q13)
Q14)
Whichofthe following is nota connection option for a multisiteCisco Unified
Communications Managerdeployment? (Source: Implementing Multisite Connections)
A)
SIP trunk
B)
C)
SIP gateway
H.323 gateway
D)
H.225 trunk
Which two commands are required to enable MGCP at a gateway when using the
configuration server feature? (Choose two.) (Source: Implementing Multisite
Connections)
Q15)
A)
B)
mgcp
seep
C)
D)
E)
ccm-manager config server ip-address
ccm-manager config
ccm-manager seep
Which parameter is set in the H.323 gateway configuration window in order to strip the
called party number to a certain number of digits? (Source: Implementing Multisite
Connections)
QI6)
Q17)
A)
B)
C)
Called Party Digits Mask
Significant Digits
Calling Party Transformation Mask
D)
F)
Number-Length
Discard Digit Instruction
Which two tmnks are not configured with the IP address ofthe next signaling device in
the path? (Choose two.) (Source: Implementing Multisite Connections)
A)
H.225 trunk
B)
nongatekeeper-controlled intercluster trunk
C)
SIP trunk
D)
gatekeeper-controlled intercluster tmnk
E)
MGCP trunk
Where do you configure SIP timers and features for an SIP trunk? (Source:
Implementing Multisite Connections)
A)
B)
C)
D)
Q18)
SIP profile
SIP security profile
SIP tmnk security profile
common trunk profile
Which ofthe following needs to be specified in the gatekeeper configuration window
when adding a gatekeeper to Cisco Unified Communications Manager? (Source:
Implementing Multisite Connections)
A)
B)
IP address ofthe gatekeeper
C)
zone name
D)
technology prefix
>2010 Cisco Systems, Inc
H.323 ID of the gatekeeper
Multisite Deployment Implementation
1-155
QI9)
Which statement about implementing PSTN backup for the IP WAN is true? (Source:
Implementing a Dial Plan for International Multisite Deployments)
Q20)
A)
B)
C)
In distributed deployments. PSTN backup for intersite calls requires CFUR.
PSTN backup requires a signaling proxy at each site.
The on- and oft-net paths are required in route groups for PSTN backup in a
centralized deplovment.
D)
CFUR allows remote-site phones to use the PSTN for calls to the main site.
What is required for implementing access and site codes in a centralized deployment?
(Source: Implementing a Dial Plan for International Mullisite Deployments)
A)
B)
C)
D)
Q2I)
Which ofthe following is not a valid type of number code for incoming ISDN PSTN
calls? (Source: Implementing a Dial Plan for International Multisite Deployments)
A)
B)
C)
D)
Q22)
Q23)
a separate translation pattern per destination site, which is configured with a
CSS that prov ides access to the destination site
a separate translation pattern per source site, which is configured with a CSS
that provides access to the source site
a separate translation pattern per source site, which is configured with the
partition ofthe destination site
a separate translation pattern per destination site, which is configured wfth the
partition ofthe destination site
international
national
subscriber
directorv- number
fhe PSTN egress gateway can be selected in which two of these ways? (Choose two.)
(Source: Implementing a Dial Plan for International Multisite Deployments)
A)
B)
C)
D)
by the partition ofthe calling device
based on the CSS ofthe gateway
bv the local route group feature
based on the matched roule pattern when route patterns exist once per site
F)
by the standard local routegroup that is configured at the gateway device pool
Where can digit manipulation be performed when digit manipulation requirements vary
for the on- and off-net paths? (Source: Implemenlinga Dial Plan for International
Multisite Deployments)
Q24)
A)
per route group ofthe route list
B)
C)
D)
route pattern
directorv number
translation pattern
When implementing TEHO for national calls and using the local PSTN gateway as a
backup, how many route patterns arerequired fora cluster with three siteslocated in
different area codes?(Source: Implementing a Dial Plan for International Multisite
Deplov ments)
A)
B)
C)
D)
1-156
3. when not using the local route group feature
6. when using the local route group feature
9. when not using the local route group feature
4. when using the local route group feature
Implementing Cisco Unitied Communications Manager. Part2 (CIPT2) v8 0
© 2010 Cisco Systems, Inc.
Q25)
Q26)
Which of these is used to globalize the callingparty number of inbound PSTN calls?
(Source: Implementing a Dial Plan for International Multisite Deployments)
A)
globalization type
B)
called number
C)
D)
inbound gateway identifier
number type
The implementation of globalized call routing does not simplify the deployment of
which two of these features? (Choose two.) {Source: Implementing a Dial Plan for
International Multisite Deployments)
A)
B)
C)
D)
E)
F)
G)
©2010 Cisco Systems. Inc.
TEHO
Device Mobility
AAR
MOH
Cisco Extension Mobility
SRST
local conference bridges
Multisite Deployment Implementation
1-157
Module Self-Check Answer Key
1-158
on
E
02)
C
Q3i
C
O-ti
D
05)
A.C
061
B
Q7)
D
OS)
Voice packets are given absolute pnority over other traffic
Q9)
C. V:
QIO)
B.C
qui
I).)-.
012)
A
013)
B
014)
C. D
Q15>
B
016)
A. D
017)
A
Q18)
B
019)
C
Q20)
A
0^1)
D
022)
c. n
02?)
A
024)
D
025)
D
Q26)
D.G
Implementing Cisco Unified Communications Manager. Part 2 (CIPT2) v8 0
©2010 Cisco Systems, Inc
Module 2
Centralized Call-Processing
Redundancy Implementation
Overview
The capability to use centralized call-processing devices that are located at remote sites
depends on the availability of Cisco Unified Communications Manager at the main site. To
provide these devices with a backup, you can use Media Gateway Control Protocol (MGCP)
fallback and Cisco Unified Survivable Remote Site Telephony (SRST).
This module describes the mechanisms for providing call survivability and device failover in
remote sites. It describes how to configure Cisco IOS routers as Cisco Unified SRST gateways
and how to use Cisco Unified Communications Manager Express in Cisco Unified SRST mode.
Module Objectives
Upon completing this module, you will be able to implement call-processing resiliency in
remote sites by using Cisco Unified SRST, MGCP fallback, and Cisco Unified
Communications Manager Express in Cisco Unified SRST mode. This ability includes being
able to meet these objectives:
•
Describe the mechanisms for providing call survivability and device failover in remote
sites, including the functions, operation, and limitations of each mechanism
•
Configure Cisco Unified SRST to provide call survivability for IP phones, and MGCP
fallback for gateway survivability
•
Configure Cisco Unified Communications Manager Express to provide telephony services
to IP phones if the connection to the centralized call agent is lost
2-2
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v80
©2010Cisco Systems, Inc.
Lesson 1
Examining Remote Site
Redundancy Options
Overview
"fhis lesson provides an overview ofthevarieties ofremote site redundancy inCisco Unified
Communications Manager multisite installations, 'fhese various mechanisms areillustrated to
give you an understanding of how the technologies interact to deliver reliable communication
sen.ices. Theremote siteredundancy mechanisms include the Media Gateway Control Protocol
(MGCP) and Cisco Unilied Survivable Remote Site Telephony (SRST).
Objectives
Upon completing this lesson, youwill be able to describe the mechanisms forproviding call
survivability and device failover inremote sites, including thefunctions, operation, and
limitations of each mechanism. This ability includes being able to meet these objectives:
•
Describe remotesite redundancy optionsand comparetheir characteristics
•
Describe how Cisco Unified SRST works
•
Describe how MGCP fallback works
•
Describe Cisco Unified SRST versions, their protocol support, their features, and the
required Cisco IOS Software releases
•
Describe dial plan requirements for MGCP fallback and Cisco UnifiedSRST
Remote Site Redundancy Overview
This topic describes the various technologies that are used toprovide remote site redundancy
for small and medium remote sites in a Cisco Unified Communications Manager environment.
Cisco Unified SRS'f and MGCP gatewav fallback are the key components to the delivery of
fail-safe communication services.
SRST and MGCP Fallback
In the case of a WAN failure, the Cisco Unified Communications
Manager servers are unreachable for devices located in remote
sites. Service survivability is provided by two technologies:
* Cisco Unified SRST provides call control services to remote IP phones.
• MGCP fallback provides voice gateway functions when the MGCP call agent
is not in control of the media gateway
Main Site
Remote Site
RSIN
Cisco Unifed
Communications
Manager
Cisco Unified Communications Manager supports Cisco Unitied IP phones at remote sites that
are attached to Cisco multiservice routers across the WAN. Before Cisco Unified SRST was
available, when the WAN connection between a router and the Cisco Unified Communications
Manager failed or when connectivity with Cisco Unified Communications Manager was lost.
Cisco Unified IP phones on the network became unusable for the duration ofthe failure.
Cisco Unified SRST overcomes this problem and ensures that Cisco Unified IP phones offer
continuous (although minimal) service by providing call-processing support for Cisco Unified
IP phones directly from the Cisco Unified SRS'f router. The system automatically detects a
failure and uses Simple Network-Enabled Auto Provision (SNAP) technology to autoconfigure
the branch office router to provide call processing for Cisco Unified IP phones that are
registered with the router. When the WAN link or connection to the primary Cisco Unitied
Communications Manager subscriber is restored, call processing reverts lo the primary Cisco
Unified Communications Manager.
MGCP gatewav fallback is a mechanism that allows a Cisco IOS router to continue to provide
voice gateway functions even when the MGCP call agent is not in control ofthe media
gateway, fhese voice gatewav functions are implemented through a fallback mechanism that
activates the so-called default technology application. The gateway then works in the same way
as a standalone H.323 or Session Initiation Protocol (SIP) gateway.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
©2010 Cisco Systems. Inc
Remote Site Redundancy Technologies
'fhe table lists the capabilities ofdifferent remote site redundancy technologies.
Remote
Site Redundancy Technologies
Cisco Umfied SRST
Cisco Unified
MGCP
Fallback
Cisco
Unified
Communications
Manager
Express in
Unified
SRST
SIP SRST
dGCP-eontrolled
SCCP
SIP
SCCP
gateways
phones
phones
phones
Cisco Unified
Communications
SRST Mode
Provides
redundancy for
Faitoack to
Deivered
service
Cisco IOS
defaut
technology
ISDN cal
preservation
Basic
Basic
telephony
SIP proxy
service
service
Manager
Express
Yes
Yes
Yes
(no MGCP)
(no MGCP)
(no MGCP)
Yes
Yes
Analog/CAS
call preservation
Maximum
number of
(NA)
1500
450
phones
To use Cisco UnifiedSRSTas your fallback modeon an MGCP gateway, you must configure
Cisco Unified SRST and MGCP fallback on the same gateway. MGCP and Cisco Unified
SRST have had the capability to be configured on the same gateway since Cisco IOS Software
Release 12.2(11)T.
Cisco Unified SRSTalso provides a basic set of featuresto SIP-based IP phones. This set of
Cisco Unified SRST basic features is also known as Cisco Unified SIP SRST. Cisco Unified
SIP SRST has to be enabledand configured separately on Cisco IOSrouters. Cisco Unified
SRST versions 3.3 and earlier provide a SIP Redirect Server function; in subsequent versions,
this function acts as a back-to-back user agent (B2BUA).
Cisco Unilied Communications Manager Express in Cisco Unified SRST mode provides more
features to a smaller maximum number of IP phones by falling back to Cisco Unified
Communications Manager Express mode. The main feature enhancements include presence,
Cisco Extension Mobility, and support of local voice-mail integrations.
VoIP call preservation sustains connectivity for topologies in which signaling is managed by an
entity (such as Cisco Unified Communications Manager) that is different from the other
endpoint and that brokers signaling between the two connected parties.
Call preservation is useful when a gateway and the other endpoint (typically a Cisco Unifi ;d IP
phone) are eolocated at the same site and the call agent is remote. In such a scenario, the call
agent, the gateway with the remote endpoint, will more likely experience connectivity failures.
Cisco Unified Communications Manager Express version 8.0 supports a maximum of 450 IP
phones (Cisco IOS 3945E router) while Cisco Unified SRST version 8.0 supports up to 1500 IP
phones on the same platform. Refer to "Cisco Unified Communications Manager Express 8.0"
(http://v\^^v.eisco.coni/en.TJS/prod/collatcral/voiccsvv/ps6788/vcallcon/ps4625/data_shcct_c78567246.html) for more details about the supported number of phones.
i 2010 Cisco Systems. Inc.
Centralized Call-Processing Redundancy Implementation
2-5
When to Use MGCP Fallback
MGCP gateway fallback that is configured as an individual feature can be used by a public
switched telephone network (PSTN) gateway if H.323 (orsome other voice application Iis
configured as a backup service. Cisco Unified SRST and MGCP fallback must be configured
on the same gateway if this singlegateway should provide Cisco Unified SRSI' fallback service
to phones and MGCP gateway fallback.
When to Use MGCP Fallback
MGCP fallback allows MGCP-controlled gateways to
fall back to a default telephony application like H.323
orSIP:
• Might be used as a standalone feature to provide analog
phones locally attached to a Cisco IOS router with
survivability
• Is most often used together with Cisco Unified SRST, which
has been supported on the same Cisco IOS router since
Cisco IOS Software Release 12.2(11)T
• Provides gateway functionality to Cisco Unified IP phones
in SRST mode
• No call preservation for ISDN PRI or BRl interfaces
Although MGCP gatewav fallback is most often used together with Cisco Unified SRST to
provide gateway functions to IP phones in Cisco Unified SRS'f mode, it can also be used as a
standalone feature. One example is that for a fax application server that uses a PRI ISDN
interface that is controlled by MGCP, connectivity to the PSTN can be preserved by MGCP
gateway fallback. Another example of an MGCP-fallback standalone configuration is a
mechanism that allows analog interfaces that arc controlled by Skinny Client Control Protocol
(SCCP) to stay in service even when the WAN connection to the Cisco linified
Communications Manager is down.
MGCP gatewav fallback preserves active calls from remote site IP phones lo the PSTN when
analog or channel associated signaling (CAS) protocols are used, for ISDN protocols, call
preservation is impossible, because Layer 3 ofthe ISDN stack is disconnected from the MGCP
call agent and is restarted on the local Cisco IOS gateway. Consequently, for active ISDN calls,
all call-state infonnation is lost in cases of switchover to fallback operation.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
©2010 Cisco Systems. Inc
When to Use Cisco Unified SRST
Cisco Unified SRST provides Cisco Unified Communicalions Manager with fallback support
for Cisco Unified IP phones that are attached to a Cisco router ona local network.
When to Use Cisco Unified SRST
Cisco Unified SRST provides limited call control
service to SCCP and SIP phones in remote sites
during WAN outage:
» Supports up to 1500 IP phones during fallback service
• Supports secure voice fallback (ifsecurity is enabled)
• Allows simple, one-time configuration for SRST fallback
service
• Only basic telephony feature support
- No message-waiting indication
- No advanced features such as Cisco Extension Mobility or
presence
Cisco Unified SRST enables routers to provide basic call-processing support for Cisco Unified
IP phones when they loseconnection to remote primary, secondary, and tertiary Cisco Unified
Communications Manager installations or when the WAN connection is down.
Cisco UnifiedSRSf also supportssecurity features. If IP phones are configured with security
mode authenticated or encrypted in Cisco Unified Communications Manager and secure Cisco
Unified SRST is deployed, securityfeatures of Cisco IP phones are preservedduring fallback.
Cisco Unified SRSTcan support SIP phoneswith the standard RFC 3261 feature locallyand
across SIP WAN networks. With Cisco Unified SIP SRST, SIP phones can place calls across
SIP networks in the same way that SCCP phones do.
Cisco Unified SIP SRST supports the following call combinations: SIP phone to SIP phone,
SIP phone to PSTN or router voice port, SIP phone to SCCP phone, and SIP phone to WAN
VoIP using SIP.
SIP proxy, registrar, and B2BUA servers are key components of a SIP VoIP network. These
servers are usually located in the core of a VoIP network. If SIP phones that are located at
remote sites at the edge ofthe VoIP network lose connectivity to the network core (because of a
WAN outage), they may be unable lo make or receive calls. Cisco Unified SIP SRST
functionality on a SIP PSTN gateway provides service reliability for SIP-based IP phones in the
event of a WAN outage. Cisco Unified SIP SRST enables the SIP IP phones to continue to
make and receive calls to and from the PSTN and also to make and receive calls to and from
other SIP IP phones.
When the IP WAN is up. the SIP phone registers with the SIP proxy server and establishes a
connection to the B2BUA SIP registrar (B2BUA router). But any calls from the SIP phone go
to the SIP proxy server through the WAN and out to the PSTN.
) 2010 Cisco Systems. Inc.
Centralized Call-Processing Redundancy Implementation
2-7
When the IP WAN has failed or the SIPproxy server hasgonedown, the call from the SIP
phone cannot getto the SIP proxy server. The SIP phone instead goes through the B2BUA
router and out to the PS'fN.
Note
The B2BUA acts as a user agent to both ends ofa SIP call. The B2BUA is responsible for
managing all SIP signaling between both ends of the call, from call establishment to
termination Each call is tracked from beginning to end, allowing the operators of the B2BUA
to offervalue-added features to the call. To SIP clients, the B2BUA acts as a user agent
server on one side and as a user agent client on the other (back-to-back) side. The baste
implementation of a B2BUA is defined in RFC 3261.
Cisco Unified SRSTdoes not supportenhanced features such as presence or Cisco Extension
Mobility. Message Waiting Indicator (MWI) is also not supported in fallback mode.
2-8
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
© 2010 Cisco Systems, Inc
When to Use Cisco Unified Communications Manager Express
in SRST Mode
Cisco Unified Communications Manager Express in SRST mode enables routers to provide
call-processing support to Cisco Unified IP phones by Cisco Unified Communications Manager
Express ifthe phones lose connection toremote primary, secondary, and tertiary Cisco Unified
Communications Manager installations or if the WAN connection is down.
When to Use Cisco Unified Communications
Manager Express in SRST Mode
Cisco Unified Communications Manager Express in
SRST mode provides backup telephony service
during WAN outage:
- Supports automatic Cisco Unified SRST configuration (basic
features only).
• Additional features can be applied globally by templates or on
a per-phone basis by manual configuration.
• All features available in Cisco Unified Communications
Manager Express are supported.
- Features not supported by standard Cisco Unified SRST
have to be statically configured (by templates or per
phone).
- Mix of preconfigured phones and automatically configured
phones is supported.
When Cisco Unified SRST functionality is provided by Cisco Unified Communications
Manager Express, you can use automatic provisioning of phones like with standard Cisco
Unified SRST. However, due to the wide feature support of Cisco Unified Communications
Manager Express, more features can be utilized, compared with standard Cisco Unified SRS'f.
Examples of features that are provided only by Cisco Unified Communications Manager
Express in SRST modeare Call Park,presence,Cisco Extension Mobility, and accessto Cisco
Unity Voice Messaging services using SCCP.
These features, however, cannot be configured automatically when a phone falls back to SRST
mode. If a certain feature is applicable to all phones or directory numbers, the configuration can
be applied by a corresponding template. If features have to be enabled on a per-phone (or perdirectory number) basis, they have to be statically configured.
Phones that do not require unique feature coniiguration can be configured automatically so that
only those phones that require individual configuration have to be statically configured in Cisco
Unified Communications Manager Express.
>2010 Cisco Systems, Inc.
Centralized Call-Processing Redundancy Implementation
Cisco Unified SRST Operation
This topic describes die function of Cisco Unified SRST.
Cisco Unified SRST Function;
Normal Situation
• Branch IP phones register with Cisco Unified Communications
Manager
• IP phones exchange keepalive messages with the central Cisco
Unified Communications Manager across the WAN.
• Cisco Unified Communications Manager manages all call
processing
Main Site
Cisco Unified
Communications
Manager
-W.N
Cisco Unified Communications Manager supports Cisco Unified IP phones al remote sites that
are attached to Cisco multiserv ice routers across the WAN. fhe remote site IP phones register
with Cisco Unitied Communications Manager. Keepalive messages are exchanged between IP
phones and the central Cisco Unified Communications Manager across the WAN. Cisco
Unified Communications Manager at the main site manages the call processing for the branch
IP phones.
2-10
Implementing Cisco Unified Communications Manager, Part 2 (C1PT2) v8.0
©2010 Cisco Systems, Inc
Cisco Unified SRST Function: Switchover Signaling
When Cisco Unified IP phones lose contact with Cisco Unified Communications Manager, they
register with the local Cisco Unified SRST router tosustain the call-processing capability that
is necessary to place and receive calls.
Cisco Unified SRST Function:
Switchover Signaling
• Ifthe WAN linkfails, IP phones lose contact with Cisco Unified
Communications Manager
• IP phones register with local gateway.
• Gateway queries IP phones for configuration and it autoconfigures
itself.
Remote Site
Main Site
Register
~T
T1
Cisco Unified
Communications
Manager
Cisco Unified SRST configuration provides the Cisco Unified IP phones with the alternative
call control destination ofthe Cisco Unified SRST gateway.
When the WAN link fails, the Cisco Unified IP phones lose contact with the central Cisco
Unified Communications Manager but then register with the local Cisco Unified SRST
gateway.
The Cisco Unified SRST gateway detects newly registered IP phones, queries these IP phones
for their configuration, and then autoconfigures itself. The Cisco Unified SRST gateway uses
SNAP technology to autoconfigure the branch office router to provide call processing for Cisco
Unified IP phones that are registered with the router.
) 2010 Cisco Systems, Inc
Centralized Cal I-Processing Redundancy Implementation
Cisco Unified SRST Function: Call Flow After Switchover
Cisco Unified SRS'fensures that Cisco Unified IP phones offercontinuous service by
providing call-processing support directly from the CiscoUnified SRST router, using a basic
set of call-processing features.
Cisco Unified SRST Function:
Cal! Flow After Switchover
• Calls between remote site phones survive switchover.
• Remote site calls using the local non-MGCP gateway survive switchover.
- Gateway provides call processing for duration of WAN failure.
Cisco Unified SRST features such as autoprovisioning and failover are
supported
Main Site
Remote Site
Public E.164
Cisco Unifed
Communications
"£L*
j£-
Calls ~*q|S
Manager
The Cisco Unified SRST gateway uses the local PSI'N breakout. Cisco Unified SRST features,
such as call preservation, autoprov isioning. and failover are supported.
During a WAN connection failure, when Cisco Unified SRST is enabled, Cisco Unified IP
phones display a message informing users that ihe phone is operating in Cisco Unified
Communications Manager fallback mode. This message can be adjusted.
While in Cisco Unified Communications Manager fallback mode, Cisco Unified IP phones
continue to send out keepalive messages to attempt to re-establish a connection with Cisco
Unified Communications Manager at the main site.
2-12
implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
>2010 Cisco Systems, Inc
Cisco Unified SRST Function: Switchback
Cisco Unified IP phones attempt to re-establish aconnection with Cisco Unified
Communications Manager at the main site periodically when they are registered with aCisco
Unified SRST gateway.
Cisco Unified SRST Function:
Switchback
• WAN link isrestored, and IPphones areable tore-establish contact with
Cisco Unified Communications Manager.
• Cisco IP Phones re-register with their primary Cisco Unified Communications
Manager
• Call processing reverts tothe primary Cisco Unified Communications
Manager
Main Site
RemoteSite
Reregister
Public E 164
Cisco Unified \**r
^ _
j&.^.
Calls
Communications
Manager
The default time that Cisco Unified IP phones wait before attempting to re-establish a
connection to aremote Cisco Unified Communications Manager is generally up to 120
seconds.
When the WAN link or connection to the primary Cisco Unified Communications Manager is
restored, after aconfigured waiting behavior, the Cisco Unified IP phones reregister with their
primary Cisco Unified Communications Manager. Three switchback methods are available on
the Cisco IOS router: immediate switchback, graceful switchback (after all outgoing calls on
the gatewav are completed), or switchback after aconfigured delay. Once switchback is
completed.'call processing reverts to the primary Cisco Unified Communications Manager, and
SRST returns to standby mode.
>2010 Cisco Systems. Inc.
Centralized CalI-Processing Redundancy Implementation
2-13
Cisco Unified SRST Timing
It typically takes three times the keepalive period for aphone lo discover that its connection to
Cisco Unified Communications Manager has failed. The default keepalive period is 3(1 seconds.
Cisco Unified SRST Timi
Cisco Unifed SRST Router
' ii i
—•"••'
TCPKeepalrvepelault30Sec
;
h
WAN connection fails
Afterthree missed keepalive messages,
i
Apprai 60 Sec
i
Aopror, 60 Sec
E Time for SRST
Phone regrsters with SRST ruuler
j Registration Process
I 10-20 Sec
L
TCP Keepalive
*i SRST router pulls IPphone coniiguration
•*j Phone fully associatedwith SRST rouler
: WANconnection restored phone
Detault 30 Sec
—ti re establishesTCPconnector keepalive
Switchback Timer
j from Unified CM* received agan
Default 120 Sec
• Unified CM - Cisco Unrfied CommunicationsManager
Ifthe IP phone has an active standby connection that is established with a Cisco Unilied SRST
router, the fallback process takes 10 to 20 seconds after the connection with Cisco Unified
Communications Manager is lost. An active standby connection to a Cisco Unified SRS'f
router exists onlv- ifthe phone has a single Cisco Unified Communications Manager in its Cisco
Unified Communications Manager group. Otherwise, the phone activates a standby connection
to its secondary Cisco Unified Communications Manager.
Note
The time that it takes for an IP phone to fall back to the Cisco Unified SRST router can vary
depending on the phone type Phones suchas the Cisco Unified IPPhone 7902G, 7905G.
and 7912G models can take approximately 2 5 minutes to fatl back to SRST mode.
Ifa Cisco Unified IP phone has multiple Cisco Unified Communications Manager systems in
its Cisco Unified Communications Manager group. Ihe phone progresses through its list before
attempting to connect with its local Cisco Unified SRST router. Therefore, the time that passes
before the Cisco Unified IPphone eventually establishes a connection with the Cisco Unified
SRST router increases witheach attempt to contact to a Cisco Unified Communications
Manager. Assuming that each attempt to connect to aCisco Unified Communications Manager
takes about 1minute, the Cisco Unified IPphone inquestion could remain offline for 3 minutes
or more following a WAN link failure. You can reduce this time by setting the keepalive timer
toa smaller value. You can configure the keepalive timer by using the Cisco CallManager
service parameter Station Keepalive Interval.
While in SRST mode. Cisco Unified IP phones periodically attempt to re-establish a connection
with Cisco Unified Communications Manager at the main site. The defaulttime that Cisco
Unified IP phones wait before attempting to re-establish a connection to Cisco Unified
Communications Manager is generally 120 seconds.
2-14
Implementing CiscoUnified Communications Manager, Part 2 (CIPT2) v8.0
)2010 Cisco Systems. Inc
MGCP Fallback Operation
This topic describes howthe MGCP gateway fallback mechanism works.
MGCP Gateway Fallback:
Normal Situation
MGCP gateways register with Cisco Unified Communications Manager.
MGCP gateways exchange keepalive messages with the central Cisco
Unified Communications Manager across the WAN.
Cisco Unified Communications Manager is the MGCP call agent.
Main Site
Remote Site
Cisco Unified
Communications
Manager
MGCP gateway fallback is a feature that improves the reliability of MGCP branch networks. A
WAN link connects the MGCP gateway at a remote site to the Cisco Communications Manager
at a central site, which is the MGCP call agent. If the WAN link fails, the fallback feature keeps
the gateway working as an H.323 or SIP gateway and re-homes back to the MGCP call agent
when the WAN link is active again. MGCP gateway fallback works along with the Cisco
Unified SRST feature.
Cisco IOS gateways can maintain links to up to two backup Cisco Unified Communications
Manager servers in addition to a primary Cisco Unified Communications Manager. This
redundancy enables a voice gateway to switch over to a backup server if the gateway loses
communication with the primary server. The secondary backup server takes control ofthe
devices that are registered with the primary Cisco Unified Communications Manager. The
tertiary backup takes control ofthe registered devices if both the primary and secondary backup
Cisco Unified Communications Manager systems fail. The gateway preserves existing
connections during a switchover to a backup Cisco Unified Communications Manager.
When the primary Cisco Unified Communications Manager server becomes available again,
control reverts to that server. Reverting to the primary server can occur in several ways:
immediately, after a configurable amount of time, or only when all connected sessions are
released.
) 2010 Cisco Systems. Inc
Centralized Call-Processing Redundancy Implementation
2-15
MJ3CP Gateway Fallback: Switchover
The MGCP gateway performs a switchover to its default technology when the keepalives
between Cisco Unified Communications Manager and the Cisco MGCP gateway are missing.
MGCP Gateway Fallback:
Switchover
If the WAN link (ails. MGCP gateways lose contact with Cisco Unified
Communications Manager.
MGCP gateway tnes to connect to the fallback Cisco Unified
Communications Manager.
MGCP gateway falls back to its default application (H.323 or SIP).
Mam Site
Remote Siti
Cisco Unified
Communicati
Manager
The MGCP gatewav fallback feature provides the following functionality:
•
MGCP gateway fallback support: All active MGCP analog, F.l CAS, and Tl CAS calls
are maintained during the fallback transition. Callers are unaware ofthe fallback transition,
and the active MGCP calls are cleared only when the callers complete their calls (hang up).
Active MGCP PRI backhaul calls are released during fallback. Any transient MGCP calls
(that is. calls that are not in the connected state) are cleared at the onset ofthe fallback
transition and must be attempted again later.
•
Basic connection services in fallback mode: Provides basic connection services for IP
telephony traffic that passes through the gateway. When the local MGCP gateway
transitions into fallback mode, the default H.323 session application assumes responsibility
for managing new calls. Onlv basictwo-party voice calls are supported during the fallback
period. When a usercompletes an active MGCP call,the MGCP application processes the
on-hook event and clears all call resources.
2-16
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
©2010 Cisco Systems. Inc
MGCP Gateway Fallback: Switchback
The switchback or re-home mechanism is triggered by the re-establishment ofthe TCP
connection between Cisco Unified Communications Manager and the Cisco MGCP gateway.
MGCP Gateway Fallback:
Switchback
When the WAN link is restored, MGCP gateway starts the re-home function.
MGCP gateway reregisters with Cisco Unified Communications Manager.
Gateway switches back lo nomial MGCP application mode.
Main Site
Remote Site
Cisco Unified
Communications
Manager
ITic MGCP-gateway-fallback feature provides the re-home functionality to switch back to
MGCP mode.
Re-home function in gateway-fallback mode detects the restoration of a WAN TCP connection
to the primary Cisco Unified Communications Manager server. When the fallback mode is in
effect, the affected MGCP gateway repeatedly tries to open a TCP connection to a Cisco
Unified Communications Manager server that is included in the prioritized list of call agents.
This process continues until a Cisco Unified Communications Manager server in the prioritized
list responds. The TCP open request from the MGCP gateway is recognized, and the gateway
reverts to MGCP mode. The gateway sends a RestartlnProgress (RSIP) message to begin
registration with the responding Cisco Unified Communications Manager.
All currently active calls that are initiated and set up during the fallback period are maintained
by the default H.323 session application, except ISDN Tl and EI PRI calls. Transient calls are
released. After re-home occurs, the new Cisco Unified Communications Manager assumes
responsibility for controlling new IP telephony activity.
)2010 Cisco Systems. Inc
Centralized CalI-Processing Redundancy Implementation
2-17
MGCP Gateway Fallback Process
The MGCP gateway maintains a remote connection to a centralized Cisco Unified
Communicalions Manager cluster by sending MGCP keepalive messages to the Cisco Unilied
Communications Manager server at 15-second intervals.
MGCP Gateway Fallback Process
MGCP Gateway
! He- .si;
TCP Keepalive Detault 15 Sec
M—
-j -.. •rv-.,-.
alien:, yar-j :••;
J WAN connection fails
: After two missed keepalive messages.
; .->'niu.>','jli(,rri MdrSs!'. •
Gateway falls back to default application
H 323 or SIP
WAN connection restored, gateway
re-establishes TCP connection; keepalive
Perform switchback immediately after active can
have finished, after a fixed amount of time,
from Csco Unified Communications Manager
rece ved again
or al a fixed tune of day
If the active Cisco Unified Communications Manager server fails to acknowledge receipt ofthe
keepalive message within 30 seconds, the gateway attempts to switch over lo thenextavailable
Cisco Unified Communications Manager server.
If none ofthe Cisco Linified Communications Manager servers responds, the gateway switches
into fallback mode and reverts to the default H.323 session application tor basic call control.
H.323 is a standardized communication protocol that enables dissimilar devices to
communicate with each other by using a common set of codecs, call setup and negotiating
procedures, and basic data-transport methods. Thegateway processes calls on itsown using
H.323 until one ofthe Cisco Unified Communications Manager connections is restored.
2-18
ImplementingCisco Unitied Communications Manager, Part 2 (CIPT2)vB.O
) 2010 Cisco Systems, Inc.
Cisco Unified SRST Versions and Feature
Support
This topic describes Cisco Unified SRST versions, their protocol support and features and the
required Cisco IOS Software release.
Cisco Unified SRST Versions
Cisco Unified
Feature
Communications
Cisco Unified
Cisco Unified
Cisco Unified
^o/.T,658
SRS?e.O
SRST 71
SRST 43
il SRST Mode
Minimum Cisco
12.4(11)XJ
IOS Software 15.0{1) forversion
Release
&0
124(22)YB
15.0(1)XA
15.0<1)Mon ISR
12,4(11JXZ
G2
Presence
/
Extension
Mobiity
Eight active
cats per line
^ (new in 8.0)
Support for
E.164 numbers
"' (new in ad)
with + prefix
Five additional
MOH streams
(new in 8.0)
(SCCP only)
The version ofthe Cisco Unified SRST application depends on the release ofthe Cisco IOS
Software that is running on the router. Each Cisco IOS Software release implements one
particular Cisco Unified SRST version. You can upgrade to anewer version ofCisco Unified
SRST via aCisco IOS update. Some ofthe recent Cisco IOS Software releases have higher
memory requirements than older releases, so make sure that you consider these requirements
before upgrading.
n
For detailed information about Cisco Unified SRST versions and their hardware and feature
support, refer to the Cisco Unified Survivable Remote Site Telephony Version 8.0 data sheet"
http://w^vv.cisco.com/cn/LlS/prod/collatcraI/voicesw/ps6788/vcallcon/ps2l69/
data_shect c78-57048i.html.
>2010 Cisco Systems, Inc.
Centralized CalI-Processing Redundancy Implementation
2-19
Cisco Unified SRST 8.0 Platform Density
The maximum number of IP phones that are supported by the Cisco Unified SRST feature
depends on the Cisco IOS router platform that is used.
Cisco Unified SRST 8.0 Platform
Density
j
Platform
Maximum Number of"
IP Phones
800 Series
4
1861
15
2801-2851
25-100
2901-2951
35-250
3825, 3845
350, 730
3925-394 5E
730-1500
The figure shows asummarv ofthe maximum number ofphones that Cisco Umtied SRST
routers can accommodate. For more details, such as minimum memory requirements, refer to
-Cisco Unified SRST 8.0 Supported Firmware. Platforms, Memory, and Voice Products
(imp'/.'vvwvvci^co.coni.ctTU'S./docs/voice ipeomm/cusrst/rcquircmcnts/guidc/srsXOspc.himl)
Note
These maximum numbers of IP phones are for common Cisco Unified SRST configurations
only Systems with large numbers of IP phones and complex configurations may not work on
all platforms and can require additional memory or ahigher performance platform.
2-20
implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8 0
© 2010 Cisco Systems. Inc
1-106
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) vS.O
© 2010 Cisco Systems, Inc.
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
1Connection options for multisite deployments include
gateways and trunks.
When implementing MGCP gateways, you do most of the
configuration in Cisco Unified Communications Manager.
When implementing H.323 gateways, you must configure
both Cisco Unified Communicalions Manager and the
gateway with a dial plan.
Trunk support in Cisco Unified Communications Manager
includes SIP and three types of H.323 trunks.
SIP trunk implementation includes trunk and dial plan
configuration in Cisco Unrfied Communications Manager.
When configuring nongatekeeper-controlled ICTs, you must
specify the IP address ofthe peer; gatekeeper-controlled
ICTs and H.225 trunks require that you configure an H.323
gatekeeper instead
References
For additional infonnation. refer to these resources:
•
Cisco Sv stems. Inc. Cisco Unified Communications System 8.x SRND, April 2010.
http:/ www.cisco.com'cn'l 'S/does/voice ip comni/'cucm/smd/oVucSvlitmi
•
Cisco Svstems. Inc. Cisco UnifiedCommunications ManagerAdministration Guide
Release 8.0(1>. Februarv 2010.
http:/';vuv wxisco.com/cn/1 :S.;docs.'\oicc_iji_coinm/cucm/admin/S I) l/ccmefg/hccm-80!cin.litinl
•
Cisco Systems. Inc. Cisco IOS Voice Configuration Library (with Cisco IOS Release 15.0
updates}. July 2007.
hltp:/v\v\v\.ciM.'o.coni,'cn.'US/'docs/iosT2 3/vvf_c/cisco_ios voice_conf!giii'ation_librarv_t
iossarv \cl.htm
© 2010 Cisco Systems, Inc.
Implementing Multisite Deployments
1-105
Unified CM* Gatekeeper-Controlled ICT
and H.225 Trunk Configuration (Cont.)
Choose previously
configured gatekeeper.
Trunks can register as
terminal or gateway with the
gatekeeper. Choose
Gateway as the terminal
a^
Enter the gatekeeper zone il
Enter the prefix that should
which the trunk should be
be registered with the
gatekeeper.
registered.
* Unified CM - Cisco Unified Communications Manager
Finally, you need to provide the gatekeeper information. From the drop-down list, choose the
gatekeeper that this trunk should register to, and then choose the terminal type. Cisco Unified
Communications Manager can register trunks as terminals or gateways with an H.323
gatekeeper. Usually die terminal type is set to Gateway.
In the Technology Prefix field, enter the prefix, which should be registered with the gatekeeper.
Note
The prefix that you enter is the prefix that the trunk will register with the gatekeeper. It can,
but does not have to, include a technology prefix. In the example, a prefix of 408 is used.
More information about prefixes and technology prefixes is provided in the Implementing
Cisco Voice Communications and QoS (CVOICE) course.
Then enter the zone in which the trunk should be registered.
Tip
1-104
The H.323 zone name is case-sensitive. Make sure that it matches the zone name that has
been configured at the gatekeeper.
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) vS.O
>2010 Cisco Systems, Inc.
Unified CM* Gatekeeper-Controlled ICT
and H.225 Trunk Configuration (Cont.)
Unified CM* Administration: Device > Trunk > Add New
i.flaiKi.f^j.jijiWiMifiSffi'Mx^'
Enter trunk name,
Choose trunk
descnpiion. and
type and click
device pool
Next
| ' Unrfied CM - Cisco Unified Communications Manager |
After vou have configured the gatekeeper, you can add the gatekeeper-controlled trunk.
Navigate to Dc\ice > Trunk and click Add New. fhen choose the trunk type. As discussed
earlier in this lesson, there are two tvpes of gatekeeper-controlled 11.323 trunks: gatekccpeicontrolled ICTs (which vou have to use when connecting to a version of Cisco CallManager
earlier than version 3.2) or 11.225 trunks (which are used to connect to Cisco Unified
Communications Manager Version 3.2 or later, as well as other H.323 devices such as
gateways or conferencing svstems).
After selecting the trunk tvpe. enter a name and description for the trunk and choose the device
pool that should be used.
)2010 Cisco Systems. Inc
Implemenling Multisite Deployments
1-103
Cisco Unified Communications Manager GatekeeperControlled ICT and H.225 Trunk Configuration
This subtopic describes how to implement gatekeeper-controlled H.323 trunks in Cisco Unified
Communications Manager.
Unified CM* Gatekeeper-Controlled ICT
and H.225 Trunk Configuration
Cisco Unified Communications ManagerAdministration:
Device > Gatekeeper > Add New
i
?
Add the gatekeeper to Cisco Unified Communications Manager.
Add gatekeeper-controfled intercluster trunk or H.225 trunk.
Enter IP address
Catek«*pT IiJ»i
gatekeeper.
Enter description.
Make sure
gatekeeper is
enabled.
| • Unified CM =Cisco Unifed Communicalions Manager |
To adda gatekeeper-controlled H.323 trunk (an H.225 trunkor a gatekeeper-controlled ICT),
you first need to add a gatekeeper to Cisco Unified Communications Manager. Navigate to
Device >Gatekeeper andclickAdd New. In the Gatekeeper Configuration window, enterthe
IP address ofthe H.323 gatekeeper and a description, 'fhen makesure that the Enable Device
check box is checked.
1-102
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
) 2010 Cisco Systems, Inc.
Cisco Unified Communications
Manager Nongatekeeper-Controtled
ICT Configuration (Cont.)
^'•^•msammmtm
• tritlsl^-.l-.,!™^
^
Enter IP address
of device on other side.
Then enter the II* address or addresses ofthe Cisco Unified Communications Managerservers
ofthe other cluster.
Note
) 2010 Cisco Systems,'
Because the nongatekeeper-control led ICT does not use a gatekeeper for address
resolution, you must manually enter the IP addresses of the devices on the other side.
Implementing Multisite Deployments
1-101
Intercluster and H.225 Trunk Implementation
Thistopic describes how to implement H.323 trunks in CiscoUnitied Communications
Manager.
Cisco Unified Communications Manager
Nongatekeeper-controlled iCT Configuration
Cisco Unified Communications Manager Administration:
Device > Trunk > Add New
BMkTaFlnd/Ua
E3
Enter trunk name,
First, crioose trunk
description, and
device pod
type and dick
Next
The figure shows how to add a new nongatekeeper-controlled ICT. First, you navigate to
Device > Trunk-and then click Add New.
Next, you must choose the appropriate trunk type. After you click Next, the Trunk
Configuration window appears, where you can configure the nongatekeeper-controlled ICT.
Enter a device name and description, and choose the device pool that should be used.
1-100
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) vS.O
>2010 Cisco Systems, Inc.
Cisco Unified Communications
SIP Trunk Configuration (Cont.)
SIP trunk security profiles are used to enable and disable security features on
SIP trunks; you configure them by navigatingto System > Security Profile >
SIP Trunk Security Profile Adefauft profile (with security disabled) exists.
SIP profiles are used to set timers and some feature settings. You configure
them by navigating to Device > Device Settings > SIP Profile. A default profile
exists
Enter IP address of
other device at end
of SIPtrimk
You must choose
SIP trunk,
security profile,
and SIP profile
These parameters
are mandatory.
no default values
exist
In the SIP Information area of the Trunk Configuration window, enter the destination addr:ss.
This IP address is for the dev ice that is located on the other end ofthe SIP trunk. This device
can be a Cisco Unified Border Element, Cisco Unified Communications Manager Express, or
any other SlP-capable device, such as a third-party SIP proxy server.
In addition, you must choose a SIP trunk security profile and a SIP profile. Both parameters are
mandator, and do not have a default value.
The SIP trunk security profile is used to enable and configure security features on SIP trunks,
such as Transport Layer Security (TLS) with two-way certificate exchange, or SIP digest
authentication. One default SIP trunk security profile exists: the nonsecure SIP trunk profile,
which has security disabled. You can configure additional SIP trunk security profiles by
nav igating to System > Security Profile > SIP Trunk Security Profile.
The SIP profile is used to set timers, Real-Time Transport Protocol (RIP) port numbers, and
some feature settings (such as Call Pickup Uniform Resource Identifiers [URIs], call hold
ringback. or caller ID blocking). One default SIP profile exists: It is called a standard SIP
profile. You can configure additional SIP profiles by navigating to Device > Device Settings >
SIP Profile.
) 2010 Cisco Systems, Inc
Implementing Multisite Deployments
SIP Trunk Implementation
Thistopic describes how 10 implement SIPtrunks in CiscoUnified Communications Manager.
Cisco Unified Communications
Manager SIP Trunk Configuration
Cisco Unified Communications Manager Administration: Device
> Trunk > Add New
».?d.n
SIB In,nl
Dev>u P'Mo col
SIP
DHJfiDnjr,
T^kU
t.v.™ »«J"
„3
V
Drtult
.
common :=.,.=. C»n*,u. '»- . N3ne
-
cadi cl«u*i^o;n"
. » Sr.- ™ D"Uk,ll
V
HUHBSJ-.M...1
.N=r«
-
LOC.OT '
-cc.Ncr
-
AAA Ofou;
<
V
PicHtCiCiur,.,;,,'
•,00,
B.3.«C.Cn,r.;U-mor
3*4,T>n.
Non«
*s
V
,
C
t4.*J.n.d
yln^^T '.idio ~ii is aj
:t™„l-..i.,u
1, ' i t , Nim
!*«-«« .ot
To add a SIP trunk in Cisco Unified Communications Manager, navigate to Device > Trunk
and click Add New. Then, in the Trunk Type drop-down list, choose SIP Trunk and click
Next.
In the Trunk Configuration window, enter a name and description for the SIP tmnk and choose
the device pool that should be used.
1-98
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Trunk Types Used by Special Applications
This subtopic describes additional trunk types that can be configured in Cisco Unified
Communications Manager.
Trunk Types Used by Special
Applications
You can configure additional trunk types in Cisco
UnifiedlCommunications Manager that are used by
specialjapplications.
• Cisco: Extension Mobility Cross Clusters:
Special trunks configured between Cisco Unified
Communications Manager clusters that should allow
Ci^co EMCC logins
- Orly SIP supported
• Call Qontrol Discovery:
- Spbcial trunks configured in Cisco Unified
Ccjnmunications Manager clusters that refer to anSAFenabled network providing Call Control Discovery features
H.323 and SIP supported
Some applications require special trunk types to be configured.
For example, when implemenling Cisco Fxtension Mobility Cross Clusters (FMCC) a
dedicated trunk has to be configured between the Cisco Unified Communications Manager
clusters that allow users ofthe remote cluster to log in locally using Cisco FMCC. These
trunks, which are exclusivelv configured for Cisco EMCC. have to use the SIP protocol; 11.323
is not supported bv Cisco FMCC. Another application that requires special trunks to be
configured is Call Control Discovery (CCD). Whenyou use CCD. internal directorv' numbers
as well as the associated external PS'fN numbers are advertised and learned from a Service
Advertisement Framework (SAF)-enabled network, fhese trunks can be cither SIP or 11.323
and must be explicitly enabled for SAF.
Note
More information about Cisco EMCC and SAF trunks will be provided in the corresponding
lessons of this course.
© 2010 Cisco Systems. Inc
Implementing Multisite Deployments
Gatekeeper-Controlled ICT and H.225 Trunk Configuration
Overview
The figure illustrates the most important configuration elements for implementing agatekeepercontrolled ICT or H.225 trunk inCisco Unilied Communications Manager.
Gatekeeper-Controlled ICT and H.225
Trunk Configuration Overview
Cisco Unified
Communications
10.1.1 1
Manager
GK Prefix. 408
GK Prefix:
10.2.1.1
Cluster
409
Cisco Unified
Com municat ions
Manager
Cluster
Gatekeeper-controlled ICT and
H.225 trunk configuration:
Cisco Unified
Communications
Manager
Cluster
* Gatekeeper (GK)
• Trunkpointingto gatekeeper
• Route pattern, route list, route group
The trunk (which points to the gatekeeper), the route group, the route list, and the route pattern
configuration are the elements ofthe gatekeeper in which you have to specify the IP address of
the gatekeeper. This implementation is like the implementation ofagateway.
1-96
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
© 2010 Cisco Systems. Inc.
Trunk Implementation Overview
This topic describes how to configure trunks in Cisco Unified Communications Manager.
Nongatekeeper-Controlled ICT and SIP
Trunk Configuration Overview
Cisco Unifed
Communications
Nongalekeeper-Conlrolled :CT
102 1 1
Manager Cluster
Cisco Unified
Communica
Manager
Access and Site
Cluster
Code. 9 222
4- Digit
Directory
Numbers
Nongatekeeper-controlled ICT
and SIP trunk configuration:
Cisco Unified
Communicalions
Manage
- Trunk with IP address of peer
• Route pattern, route list, route group
Cluster
The figure illustrates the most important configuration elements for implementing a SIP or
nongatekeeper-controlled ICT in Cisco Unified Communications Manager, "fhese elements arc
the configuration ofdie trunk itself, in which you have to specify the IP address ofthe peer, as
well as the route group, route list, and route pattern configuration, fhis implementation is like
the implementation of a gatewav.
'2010 Cisco Systems. Inc
ImplementingMultisite Deployments
Cisco IOS H.323 Gateway Configuration
This subtopic describes how to configure a Cisco IOS router as an H.323 gateway.
Cisco IOS H.323 Gateway Configuration
10-digit numbers are sent to Cisco Unrfied Communications Manager on
incoming PSTN calls in this configuration.
Digit manipulation on gateway or in Cisco Unified Communications Manager
is required to get to 4-digit directory numbers.
interface FastEthernet0/0
ip address 10.1.1.101 255.255.255.0
h323-gateway voip interface
h323-gateway voip bind ercaddr 10.1.1.101
1
dial-peer voice 11 voip
destination-pattern 511555....
session target ipv4:10.1.1.1
incoming called-number 9T
codec gTllulaw
1
dial-peer voice 21 pots
destination-pattern 9T
direct-inward-dial
port 0/0/0:23
When configuring an H.323 gateway, the first task is to enable H.323 at one IP interface. If
multiple IP interfaces are present it is recommended that you use a loopback interface.
Otherwise, if the interface that has been selected for H.323 is down, the H.323 application will
not work, even if-other interfaces could be used to route the IP packets. In this example, there is
only one Ethernet interface, and H.323 has been enabled on that interface, using the h323gateway voip interface and h323 gateway voip bind srcaddr IP address commands.
In contrast to MGCP gateways in which the call agent takes care of call routing, H.323
gateways require local dial plan configuration. In the example, the H.323 gateway is configured
with a VoIP dial peer that routes calls that are placed to the PSTN number 511555... ofthe
gateway toward Cisco Unified Communications Manager. The gateway receives these calls
from the PSTN because 511555 1001-1003 is the direct inward dialing (DID) range ofthe
PSTN interface (port 0/0/0:23). In addition, the PSTN gateway is configured with a POTS dial
peer that routes all calls starting with 9 out to the PSTN, using the ISDN PRI (port 0/0/0:23).
Note that the configured digits of a destination pattern in a POTS dial peer are automatically
stripped off. Therefore, the 9 is not sent out to the PSTN. In the other direction, the gateway
does not perform any digit manipulation because VoIP dial peers do not strip off any digits
automatically. Cisco Unified Communications Manager receives H.323 call setup messages for
calls that were received from the PSTN in their entire length (usually 10 digits). Because the
internal directorv numbers arc four digits, either Cisco Unified Communications Manager or
the H.323gatewav need to be configured to strip the leadingdigits so that the remaining four
digits can be used to route the call to internal directory numbers.
Note
More informationon how to implement digit manipulation is provided in the next lesson of
this module.
1-94
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) v8.0
J2010 Cisco Systems, Inc.
Cisco Unified Communications Manager H.323 Gateway
Configuration
This subtopic describes how to configure an 11.323 gateway in Cisco Unified Communications
Manager.
Cisco Unified Communications
Manager H.323 Gateway Configuration
Cisco Unified Communications ManagerAdministration: Device
> Gateway > Add New > H.323 Gateway
Configure gateway IP address,
desenption, and device pool.
:c.;:::
-
-""•-
•tlon
..^--,r, ;,,-,•
- s ">o
*:j
-•:••
<
-"•" -'"
Is
•<. •-.••-
l1.!
10-digit numbers recei\ed from the
- inbiund (•<•<-
-
NO"!
>
•
, «,,. ,
PSTN are stripped down to 4-digit
internal directory numbers.
v
zm
Hi.i-.-,•,] N,™ . ,
'JE-.ll.;rt..t,Lr
I,
i„ . , -
:-,-,, ,d
>•:•>•,
To add an H.323 gateway to Cisco Unified Communications Manager, navigate to Device >
Gateway and click Add New. fhen, from the Gateway Type drop-down list, choose H.323 and
click Next.
In the tiateway Configuration window, enter the IP address ofthe 11.323 gateway in the Device
Name field, enter a description, and select the device pool that should be used. If Cisco Unified
Communications Manager should consider only some ofthe called digits, you can set the
significant digits parameter to the numberof least significant digits that shouldbe used for
routing inbound calls. In the example that is provided in the previous topic, in which the
gateway sends complete 10-digit PSTN numbers to Cisco Unified Communications Manager,
setting the significant digits to 4 wouldallow the incoming calls lo be routed to internal
director; numbers without any additional configuration (such as translation patterns).
© 2010 Cisco Systems. Inc
Implementing Multisite Deployments
1-93
Step2
Configure oneor more VoIP dial peerspointing to Cisco Unified Communications
Step 3
Configure one or moredial peers pointingto the PSTN.
Manager.
1-92
Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) vS.O
© 2010 Cisco Systems, Inc.
H.323 Gateway Implementation Review
This topic describes how to implement an H.323 gateway.
H.323 Gateway Implementation
Overview
Configure H.323 gateway in
Cisco Unified Communications
PSTN
Manager.
Configure Cisco IOS gateway:
10.1.1.1
IP address used for H.323
Dial peer pointing to Cisco
Unified Communications
Manager
- Dial peer pointing to PSTN
1001-1003
To implement an H.323 gatewav. you first must add the gatewav to Cisco Unified
Communications Manager. When adding the gateway, you need lo specify the IP address ofthe
gateway.
Note
More information about the H.323 protocol and H.323 gateway characteristics have been
provided in the ImplementingCisco Voice Communications and QoS (CVOICE) course.
MGCP gateway implementation with Cisco Unified Communications Manager has been
covered in detail in the Implementing Cisco Unified Communications Manager Part 1
(CIPT1) course This topic is only a high-level review of H 323 gateway implementation.
Then vou need to configure the Cisco IOS gateway by following these steps:
Step 1
Configure the H.323 gateway, specifying its H.323 ID and the IP address to use.
You do this configuration on any interface, typically on a loopback interface, Fnsure
that you use the same IP address that you configured in Cisco Unified
Communications Manager for the H.323 gateway.
Note
Ifthe IP address that is configured in Cisco Unified Communications Manager does not
match the IP address that is used by the gateway, Cisco Unified Communications Manager
considers the H 323 signaling messages to be sent from an invalid (unknown) source and
ignores them. However, itdoes not ignore the messages ifpromiscuousoperation has been
permitted (thisservice parameter can be configured in Cisco Unified Communications
Manager)
© 201D Cisco Syslems. Inc.
Implementing Multisite Deployments
1-91
mgcp
mgcp call-agent 10.1.1.1 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp rtp payload-type g726rl6 static
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Configuring Cisco IOS Gateway for MGCP—Example
The figure shows an example of a Cisco IOS MGCP gateway that pulls its configuration from a
configuration server.
Configuring Cisco IOS Gateway for
MGCP—Example
controller El
0/1/0
framing bdb3
Only the commands in red
have been manually
configured. Commands in
linecode
y. i
black have been added via
crc4
EiirealoTS
S
U
-,hvi,~! a:-.. -
isdn incoming-voice voice
TFTP Commands in blue are
mandatory for manual MGCP
configuration.
-r.-yp
interface SerialQ/1/0:15
1,tJ" i!'•''••-''
l5 ••'"r-r^^-r
ccm-manager music-on-hold
ccm-manager config server 10.1,1,1
ccm-manager config
:-<;. j. '.H\i-*<u>,il
::.
..: : 2427
service-type
mgcp
version 0.1
mgcp rtp unreachable timeout 1000 action n otify
mgcp modem passthrough voip mode nee
mgcp package-capability rtp-package
mgcp package-capability sat-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp rtp payload-type g726rl6 static
mgcp profile default
In the example, there is one Cisco Unified Communications Manager server (providing call
processing and TFTP services) with the IP address 10.1.1.1. 'fhere is a Cisco IOS MGCP
gateway with a connection to the PSTN using an Fl interface (port 0/1/0). The gateway and its
HI PRI endpoint have been added to Cisco Unified Communications Manager. At the gatewav.
the commands ccm-manager config server 10.1.1.1 and ccm-manager config server have
been entered. No MGCP configuration commands have been manually entered, because the
MGCP configuration is automatically downloaded and applied by the configuration server
feature.
After the gatewav downloaded its cnf.xml configuration file from the Cisco Unified
Communications Manager TFTP server, these MGCP commands were added and saved to
NVRAM:
controller SI 0/1/0
framing crc4
linecode hdb3
pri-group timeslots 1-31 service mgcp
!
interface Serial0/l/0:15
isdn switch-type primary-4ess
isdn incoming-voice voice
isdn bind-13 ccm-manager
i
ccm-manager mgcp
ccm-manager music-on-hold
© 2010 Cisco Systems, Inc
Implementing Multisite Deployments
—
If Foreign Exchange Station (FXS) or Foreign Exchange Office (FXO) interfaces are
to be MGCP-controlled, enable MGCP on the corresponding plain old telephone
service (POTS) dial peersby usingthe service mpepapp command.
—
Enable MGCP.
Note
More information about manual configuration of MGCP gatewaysis provided inthe CVOICE
course.
•
Mixed use of configuration server and manual configuration:
—
Follow the same procedure asfor the MGCP gateway configuration, using the
configuration server.
—
—
Disable the configuration server, using the ccm-managerconfig command,
Manually remove configuration that is received from the configuration server, or
add more configuration to it.
Note
Be aware that, as long as the configuration server isactive onthe Cisco IOS gateway, every
time the MGCP endpoint is resetfrom Cisco Unified Communications Manager, the Cisco
IOS configuration also will be rewritten. In addition, when you reload theMGCP gateway,
the MGCP configuration will be rewritten as long as the configuration server is enabled.
Therefore, itis common practice to use the configuration only for initial configuration when
manual changesare required. After you modify the downloaded configuration, you
deactivate the configuration serverso thatthe manually addedchangesare preserved
L
M
'••»
Also, whenyou reset an MGCP gatewayor MGCP endpoint inCisco Unified
Communications Manager, thegateway orendpoint is not automatically resetat therouter.
Therefore, when the configuration server feature is not enabled on the Cisco IOS gateway,
X
^>*
you need to follow the next procedure inorder to reset an MGCP gatewayor MGCP
endpoint.
-+
m,
First, reset the MGCP gatewayor MGCP endpoint inCisco Unified Communications
Manager. Then enterthe no mgcp command, followed by the mgcp command in
configuration at the Cisco IOS gateway.
L
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems, Inc
I
Cisco IOS Gateway MGCP Configuration Methods Review
This subtopic reviews how to configure a Cisco IOS MGCP gateway to integrate with Cisco
Unified Communications Manager.
Cisco IOS Gateway MGCP
Configuration Methods Review
Various ways exist to configure MGCP on the Cisco
IOS gateway:
• Configuration server
Cisco IOS configuration will be downloaded automatically.
Recommended and easiest way.
• Manual configuration
Cisco IOSconfiguration for controller and MGCP must be
entered manually.
- Fast way to configure fractional T1/E1.
• Mixed
Initial configuration received by configuration server.
Configuration server will be disabled, and configuration
can be altered manually.
Ailer adding the MGCP gatewav in the Cisco Unified Communications Manager web
administration, vou need to configure the Cisco IOS MGCP gateway to register ittothe Cisco
Unified Communications Manager, 'fhere are three methods for configuring a Cisco IOS
Software-based gateway to register itto Cisco Unified Communications Manager via MGCP:
• Cisco IOS MGCP gateway configuration with theuse of a configuration server:
—
Specifv the IP address ofthe configuration server (Cisco Unified Communications
—
If more than one Cisco Unified Communications Manager TFTP server is deployed
Manager TFTP server).
inthe Cisco Unified Communications Manager cluster, configure the gateway with
all Cisco Unified Communications Manager TFTP server IP addresses.
—
•
Fnable the configuration server feature.
Manual Cisco IOS MGCP gateway configuration:
— Specify the IP address ofthe MGCP call agent (Cisco Unified Communications
Manager server).
—
Ifmore than one Cisco Unified Communications Manager server is used for call
processing (that is. running the Cisco CallManager service), configure the gateway
with aprimary and redundant call agent by specifying the IP addresses oftwo Cisco
Unified Communications Manager call-processing servers.
—
Configure global MGCP parameters.
Examples ofglobal MGCP configuration commands arc mgcp packet and mgcp
rtp commands.
)2010 Cisco Systems. Inc
Implementing Multisite Deployments
L
MGCP Gateway Implementation Review
L
This topic describes how to implement an MGCP gateway.
L
Cisco Unified Communications Manager
Gateway Configuration Procedure
1
Add an MGCP gateway.
2
Add voice modules.
3
Add VICs to the modules.
4
Add and configure MGCP endpoints.
5
Configure the MGCP gateway.
L
L
L
To implement an MGCP gateway, you first need to add the gateway to Cisco Unified
Communications Manager. Next, you add voice modules and voice interface cards (VICs) to
the gatewav. and finally, you configure the endpoints.
Note
More informationabout the MGCP and MGCP gateway characteristics are provided in the
ImplementingCisco Voice Communications and QoS (CVOICE) course. MGCP gateway
implementation with Cisco Unified Communications Manager has been covered in detail in
Implementing Cisco Unitied Communications Manager, Part 1 (CIPT1) course. This topic is
only a high-level review of MGCP gateway implementation.
After adding the MGCP gateway and its endpointsand configuring the endpoints in Cisco
Unified Communications Manager, you need to configure the MGCP gateway itself. Cisco
Unified Communications Manager stores at its TFTPserver an XMLconfiguration file that can
be downloaded by the MGCP gateway. Alternatively, you can configure the gateway manually.
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With a gatekeeper-controlled ICT. you configure only one trunk. That trunk then communicates
via the gatekeeper with all other clusters that are registered to the gatekeeper. If a cluster or
subscriber becomes unreachable, the gatekeeper automatically directs the call to another
subscriber in the cluster or rejects the call if no other possibilities exist, fhis action allows the
call to be rerouted over the PSTN (if required) with little incurred delay. With a single Cisco
gatekeeper, it is possible to have 100 clusters that arc registering a single trunk each, with all
clusters able to call each other. With nongatekeeper-controlled trunks, this same topology
would require 99 trunks to be configured in each cluster. The gatekeeper-controlled ICT should
be used for communicating only with other Cisco Unified Communications Managers,because
the use of this trunk with other 11.323 devices might cause problems with supplementary
services. In addition, a gatekeeper-controlled ICT must be used for backward compatibility
with Cisco Unified Communications Manager versions earlier than Version 3.2 (referred lo as
Cisco CallManager).
Ihe H.225 trunk is essentially the same as the gatekeeper-controlled IC'1. except that't has the
capability of working with Cisco Unified Communications Manager clusters (Version 3.2 and
later), as well as other 11.323 devices, such as Cisco IOS gateways (including Cisco Unified
Communications Manager Express), conferencing systems, and clients. This capability is
achieved through a discovery mechanism on a call-by-call basis. This type of trunk is the
recommended H.323 tmnk if all Cisco Unified Communications Manager clusters are at least
Version 3.2.
© 2010 Cisco Systems. Inc.
Implementing Multisite Deployments
1-85
*
H.323 Trunk Comparison
The table compares characteristics ofthe three available H.323 trunk types.
H.323 Trunk Comparison
IP address
resolution
Gatekeeper call
admission
Scalabiity
Nongaiefteeper-
Gatekeeper-
ControlledlCT
ControBedlCT
IP address specified
in trunk configuration
IP address resolved by H.323 RAS
{gatekeeper)
NO
Yes,byH3Z3RflS (gatekeeper)
Limited
Scalable
Cisco Unified
Peer
Communications
Manager
eJS^jESiSSS
The nongatekeeper-controlled ICT isthe simplest, since itdoes not use a gatekeeper. Itrequires
the IPaddress ofthe remote Cisco Unified Communications Manager server or servers to be
specified, because the dialed number isnot resolved toan IP address by a gatekeeper. Call
Admission Control (CAC) can be implemented by locations but not by gatekeeper CAC.
Scalability is limited because no address resolution is used andall IP addresses have to be
configured manually. The nongatekeeper-controlled ICT points to theCisco Unified
Communications Manager server ofthe other cluster.
You may define up to three remote Cisco Unified Communications Manager servers in the
same destination cluster. The trunk will automatically load-balance across all defined remote
Cisco Unified Communications Manager servers. In the remote cluster, it is important to
configure a corresponding ICT (nongatekeeper-controlled) thathas a Cisco Unified
Communications Manager group containing the same servers that were defined as remote Cisco
Unified Communications Manager servers in the first cluster. Asimilar configuration is
required in each Cisco Unified Communications Manager cluster that isconnected by the ICTs.
Fora larger number of clusters, the gatekeeper-controlled ICT should be used instead ofthe
nongatekeeper-controlled trunk. The advantages of using the gatekeeper-controlled tmnk are
mainly the overall administration ofthe cluster and failover times. Nongatekeeper-controlled
trunks generally require that a full mesh oftrunks be configured, which can become an
administrative burden asthe number ofclusters increases. In addition, ifa subscriber server ina
cluster becomes unreachable, there will be a 5-second (default) timeout while the call is
attempted. Ifan entire cluster isunreachable, the number ofattempts before either a call failure
ora rerouting ofthe call over the PSTN will depend onthe number of remote servers that are
defined for the tmnk and on the number oftrunks in the route list or route group. Ifthere are
many remote servers and many nongatekeeper-controlled trunks, the call delay can become
excessive.
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H.323 Trunk Overview
The figure illustrates varioustypes olT1.323 trunks.
H.323 Trunk Overview
Cisco Unified
Communications Manager
Nongatekeeper-Con trolled ICT
Cluster B
Cisco Unified
Communications
Manager
Cluster A
Cisco Unified
Com mu nica tions
Manager
Cluster D
In the example, the Cisco linified Communications Manager cluster Auses anongatekeepercontrolled ICT to Cisco Unified Communications Manager cluster B. In addition. Cisco
Unified Communications Manager cluster A isconfigured with a gatekeeper-controlled ICT .
The gatekeeper-controlled ICf points to agatekeeper, which is used for address resolution. In
this example, the gatekeeper can route calls between Cisco Unified Communications Manager
clusters A. C. and D.
© 2010 Cisco Systems, Inc.
implementing Multisite Deployments
1-83
SIP Trunk Characteristics
This subtopic describes the characteristics of a SIP trunk.
SIP Trunk Characteristics
Distributed dial plan
Can be connected to any device supporting SIP, including Cisco IOS
gateways, Cisco Unified Border Element, remote Cisco Unified
Communications Manager dusters. SIP network servers (proxy), and others
Simple, customizable protocol; rapidly evolving feature set
Main
Site
SIP uses the distributed call-processing model, so a SIP gateway or proxy has its own local dial
plan and performs call processing on its own. A Cisco Unified Communications Manager SIP
tmnk can connect to Cisco IOS gateways, a Cisco Unified Border Element, other Cisco Unified
Communications Manager clusters, or a SIP implementation with network servers (such as a
SIP proxy).
SIP is a simple, customizable protocol with a rapidly evolving feature set.
Note
When you use SIP trunks, Media Termination Points (MTPs) might be required if the
endpoints cannot agree on a common method of dual tone multifrequency (DTMF)
exchange.
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>2010 Cisco Systems, Inc.
Cisco IOS Gateway Protocol Comparison Review
fhe figure reviews the advantages and disadvantages of 11.323 gateways. MGCP-controIlcd
gateways, and SIP gateways.
Cisco IOS Gateway Protocol
Comparison Review
Dial plan drectty on
the gateway
Translations defined
per gateway
Pros
Regional
requrements that can
be met
More specific cal
routing
Advanced tax support
Cons
Complex configuration
Centrafzed dial plan
configuration
Centralzed gateway
configuration
Dal plan directly on the
gateway
Translations defined
per gateway
Third-party telephony
Simple gateway
configuration
system support
Easy implementation
Third-party gateway
inleroperablity
Support of QSIG
supplementary services
Extra cal I-routing
configuration for survivability
Third-party
end-device support
Lessfeaturesupport
fach ofthe three gateway protocols has advantages and disadvantages when compared with
each other. There is no generally "best" gateway protocol. You should select the most
appropriate protocol, depending on the individual needs and demands in a Cisco Unified
Communications Manager environment.
Note
The Implementing Cisco Voice Communications and QoS (CVOICE) course provides
detailed information on functions and features ofthe H.323, MGCP, and SIP.
© 2010 Cisco Systems, Inc.
Implementing Multisite Deployments
Cisco IOS Gateway Protocol Functions Review
The table reviews Cisco IOS gateway protocol functions by protocol.
Cisco IOS Gateway Protocol Functions
Review
^^^^H Function
sip
MGCP
H.323
Clients
Dumb
Intelligent
NFAS
Not supported
Supported
Supported
QSIG
Supported
Not supported
Not supported
More effort to
Easy to
implement
Easy to
implement
TCP and UDP
TCP
TCP or UDP
ASCII
Binary (ASN.1)
ASCII
No
Yes
Yes
Yes'
Yes
Yes
No
Yes
Yes
Fractional T1/E1
Signaling protocol
Code basis
Call survivability
FXO caller ID
Call applications
usable
implement
^^H
Intelligent'
I' Support irtroQuced with Cisco Unrfied Communications Manager Version 8.0 |
As shown in the table, the three main gateway signaling protocols—MGCP, H.323, and SIP—
provide various features and functions when implemented with Cisco Unified Communications
Manager and Cisco IOS gateways.
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>2010 Cisco Systems, Inc.
Cisco Unified Communications Manager Connection Options
Overview
j
In Cisco Unified Communications Manager, you can configure gateways and trunks for
connections to the public switched telephone network (PSTN) or to other VoIP domains.
Cisco Unified Communications
Manager Connection Options Overview
• Gateways
- H.323 (any H 323 device when not using a gatekeeper)
- MGCP
Skinny Client Control Protocol
• Trunks
H.323
• Direct connection to another Cisco Unified
Communications Manager cluster
• Connection to any H.323 device via a gatekeeper
SIP
* Connection to any SIP device
Gateways areconfigured bv the VoIP protocol (hat they use. Cisco Unified Communications
Manager supports H.323 gateways. Media Gateway Control Protocol (MGCP) gatewavs. and
Skinny Client Control Protocol (SCCP) gatewavs. Trunks can be configured as H.323 trunks
(three types are available) or SIP trunks.
Trunks and gatewavs arcconfigured when connecting to devices that allow access to multiple
endpoints. If the destination is a single endpoint. phones areconfigured. Phones can be
configured as SCCP. SIP. or 11.323.
When Cisco Unified Communications Manager routes calls lo a device that is using MGCP.
SCCP. or SIP. it is obvious which type of deviceto add. becausethese protocols can be
configured onlv with either a gateway or a trunk. Inthe case of 11.323. however, an H.323
gateway as well as an H.323 trunk can be configured, and it is important to know whether to
use the gateway or the trunk. You use H.323 trunks only when connecting toanother Cisco
Unified Communications Managerserver (eithera clusteror a standalone Cisco Unified
Communications Manager server, in the caseof Cisco Unified Communications Manager
Business fdition) or when using an H.323 gatekeeper.
H.323 gatewavs are configured when connecting to any other H.323 device that isnot an
endpoint. Such dev ices can be Cisco IOS H.323 gateways or11.323 gateways ofother vendors.
© 2010 Cisco Systems. Inc
Implementing Multisite Deployments
1-79
Multisite Connection Options
This topic provides an overview of connection options in multisite environments.
Connection Options for Multisite
Deployments
Main
Site
The figure shows a Cisco Unified Communications Manager cluster at the main site with these
connections to other sites:
•
Intercluster trunk (ICT) to another cluster located at a different site
•
An H.323 gateway that is located at a remote site
•
A Session Initiation Protocol (SIP) trunk that is connected to an Internettelephony service
provider (ITSP) via a Cisco Unified Border Element
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Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0
12010 Cisco Systems, tnc.
Lesson 3
Implementing Multisite
Connections
Overview
Cisco Unified Communications Manager multisite deployments can use various connection
options between sites. Ihis lesson describes connection options and explains how toconfigure
them.
Objectives
Upon completing this lesson, you will be able to configure gateways and trunks in multisite
nvironmenls. This ability includes being able to meet these objectives:
Identify the characteristics ofthe trunk and gateway types that aresupported by Cisco
Unified Communications Manager
Describe how lo implement MGCP gateways
Describe bow to implement 11.323 gateways
Describe various types of trunks that aresupported by Cisco Unified Communications
Manager
Describe how to implement SIPtrunks in Cisco Unitied Communications Manager
Describe how to implemeni intercluster and H.225 trunks in Cisco Unitied
Communications Manager
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© 2010 Cisco Systems; Inc.
References
for additional information, refer to these resources:
• Cisco Svstems. Inc. Cisco Unified Communicalions System 8.x SRND, April 2010.
lutp: Asww.cisco.com/en/US/docs.voice ip comm/cucm/^rnd/8\/uc8\.h(inl
• Cisco Svstems. Inc. ('isco (Unified Communications Manager Administration Guide
Release 8.0(1). February 2010.
Imp: 'wwu,ciscn.com/eiiT,S/doev\oice_ip_comm/cucm/admin/8 OJ/cemciU/bccm-801
cm.him!
©20)0 Cisco Systems. Inc
Multisite Deployment Implementation
1-75
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
• Multisite deployment solutions include QoS, efficient use of
IP WAN bandwidth, backup scenarios in caseof WAN failure
access and site codes, and the use ofthe Cisco Unified
Border Element.
•QoS allows certain communication flows to be processed
with higher priority than others.
You can conserve bandwidth by using low-bandwidth codecs
deploying mixed conference bridges or transcoders using '
RTP-header compression, and deploying local media
resources.
Summary (Cont.)
• Cisco Unified Communications Manager availability features
include fallback for IP phones, CFUR, AAR, and CFNB and
mobility features such as Cisco Device Mobility, Cisco '
Extension Mobility, and Cisco Unified Mobility.
•You can build multisite dial plan solutions by using Cisco IOS
gateway and Cisco Unified Communications Manaqer dial
plan tools.
ACisco Unified Border Element in flow-through mode
completely hides internal devices, such as Cisco Unified
Communications Manager and IP phones, from the outside
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'2010 Cisco Systems. Inc.
Cisco Unified Border Element in Flow-Through Mode
The figure illustrates how the use of aCisco Unified Border Element protects inside devices
such afasco Unified Communications Manager and IP phones by acting as asignaling and
media proxy.
Cisco Unified Border Element in
Flow-Through Mode
Company A
Signaling and
Media Packets
Public IP Address B
Repackaged
10.1 1 1
ITSP
Cisco Unified
Communications
Manager
SCCP
Private IP
10.2.1.5 Address:
Private IP Network' 1000.0/8
..1.1.110 10 31.1
Signaling: 10.
RTP-1Q2.1.5.O10311
Public IP
AddressA
10.3.1.1
|~| Signaling: A(Public IP) to B(Public IP) |
—
RTP: A(Public IP) to B(Public IP)
In the example. Cisco Unilied Communications Manager has aprivate IP addrcs of 10.1.1.1.
and the IP phone has aprivate IP address of 10.2.1.5. ACisco Unified Border Element
o, t the Cisco Unified Communications Manager cluster to the outside world, in this case.
^ ninternet telephony service provider (ITSP,. The Cisco Uinfie Border hcment is
configured in flou-through mode and uses an internal private IP address ot 10... Iand an
external public IP address of A.
When Cisco Unified Communications Manager wants to signal calls to the ITSP. it does not
Tend he packets to the IP address ofthe ITSP (IP address D). Instead it sends them to the
ternalkddressoHhe^
configuration Cisco Unified Border Element then establishes asecond cal leg to the 1ISP.
unCi
blic IP address Aas the source and IP address B(ITSP) as the desUr.Uo^Occ the
call is set up the Cisco Unified Border Element terminates Rl Ptoward the ITSP. using its
pX II' address, and sends the received RTP packets to the internal IP phone, using its
internal IP address.
This solution allows Cisco Unified Communications Manager and IP phones to communicate
on v htne ntemal. priv ate IP address ofthe Cisco Unified Border Element, he onlv IP
dre h sible to the ITSP is the public IP address ofCisco Unified Border f.lcment.
Multisite Deployment implementation
2010 Cisco Systems, inc
1-73
NAT and Security Solutions
This topic describes solutions to NAT and security issues in multisite environments.
CiscoUnified Border Element as a Solution to
NAT and Security Issues in Multisite
Environments
Cisco Unified Border Element acts as an application proxy:
• Splits calls between inside and outside into two call legs.
• Supports signaling interworking{SIPand H.323).
• Two modes:
- Signaingonly(flow-around)
- Signaling andmedia (flow-through)
*h?!6^,31
£!?ms (IP phores- Cisco Unified Communications Manager)
do not need IPconnectivity to outside.
" &nl*,Fisco
^"'^ Bordw dement needs tohave ^public
IP address
- Solves NAT and security issuesforinternal devices.
- Only Cisco Unified Border Element itself needs to behardened.
" ?SJ^-a9en?
and flp P^ndition perform similar functions and can
sometimes be used as an alternative.
" °!lnert;alte™fives avaiable (trusted relay points, proxies in Cisco
Adaptive SecuntyApptance [ASA])..
When Ctsco Unified Communications Manager servers and IP phones need to connect lo the
Internet. Cisco Unified Border Element can be used as an application proxy. When used in this
way, Cisco Unified Border Element splits off-net calls inside and outside into two separate call
,,'.-!f°
°rder Element also features signaling interworking from SIP to SIP SIP
to 11.323, H.323 to SIP. and H.323 to H.323.
The Cisco Unified Border Element can function in two modes:
• Flow-around: In this mode, only signaling is intercepted by Cisco Unified Border
Element Media exchange occurs directly between endpoints (and>*M around Cisco
Unified Border Element). Only signaling devices (Cisco Unified Communications
Manager) are hidden from the outside.
' ,Fi°r"?M0U?h:
In thiS m0dc" both siSnalinS **media slr<*ms are intercepted by Cisco
Unified Border Element (byflowing through Cisco Unified Border Element). Both Cisco
Unified Communications Manager and IP phones are hidden from the outside.
In flow-through mode, only Cisco Unified Border Element needs to have apublic IP address
so NAT and security issues for internal devices (Cisco Unified Communicalions Manager '
ouZe
it should
sh tTu
7 I"'-against
B£CaUSe
CiSC° Unified B0nler Hlement is "P""" ^the
outside, it
be hardened
attacks.
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©2010 Cisco Systems. Inc.
Localization of callingparty numbers a. phones: Caller IDs sPl yed at phon n
also be localized so that the end users are not lim.ted to seeing all callers ,n W»A™*
fonnat Again, global transfonnations (ofthe calling-party number only in this case) can
Tsed sothat caller 1Ds. which might be different at each site. -™^^^'^
format Similarh. the globalized calling-party number is also maintained in call lists so that
users can place callbacks to globalized numbers without needing lo edit the number.
Substantial simplification of dial plans: With local route groups and global
transfonnations. globalized call routing drastically reduces the s™^™^™%£*ld]
plans Features such as TEHO. AAK. SKST. CEUR. Cisco Device Mobil, y. and Cisco
Extension Mobilitv can be implemented much more easily in international deployments.
Multisite Deployment Implementation
© 2010 Cisco Systems. Inc
1-71
Globalized Call Routing Advantages
There are several advantages ofglobalized call routing that are especially applicable to
international multisite deployments.
Globalized Call-Routing Advantages
• Globalized call routing allows one format to be used to store
orconfigure PSTN numbers (E.164 with + prefix).
• Speed dials, fastdials, call forward destinations, AAR
destinations, etc., look the same at all sites.
• Manually typed numbers following local rules areglobalized
at ingress.
• Local requirements are managed atthe egress device, not by
separate patterns and associated route lists.
• Common format can be used in address books and can be
exchanged worldwide.
• Calling party numbers shown at phones can be shown in
localized format, but globalized format can be used for
callbacks.
• International dial plans are substantially simplified.
Hereare the advantages:
• Universal format to store or configure PSTN numbers: With globalized call routing
you can configure or save all PSTN numbers in auniversal format that you can use
worldwide, regardless of local PSTN dial rules.
• PSTN destinations that are configured in Cisco Unified Communications Manager
and that are independent ofthe site that is used for dial-out: Speed dials fast dials call
forward destinations. AAR destinations, and Cisco Unified Mobility remote destinations
share the same format. Because all call routing is based on this format, calls to these
numbers will work from anywhere within the cluster, regardless ofthe requirements ofthe
local PSTN gateway.
• Transformation of input at call ingress: When implementing globalized call routing vou
can allow end users to manually dial numbers as they normally do. Ifyou globalize their
localized input during call ingress, any input format can be supported, while call routine
itself is based on a standardized format.
• Localization at call egress: The various requirements that are applicable at various egress
devices can be easily managed during call egress—that is, atler call routing and path
selection-^ using features such as global transformations that allow digit manipulation at
the egress device, regardless ofthe matched route pattern and route list.
• Utilization across different devices: Address book entries can be shared by all devices
that support E.164 format and the +prefix. This situation allows the utilization of
centralized directories regardless ofthe used endpoint. For example, cell phones and Cisco
Unified Communications Manager can synchronize their directories from one and the same
source (tor example, aLightweight Directory Access Protocol [LDAP] directory)
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•
Internal to Internal
•
—
Calling-party number: directorv' number
—
Called-party number: directorv' number
Internal to External
Calling-part} number: f.164
—
Called-party number: H.164
At the right side ofthe figure, call egress is illustrated by two types of call targets:
•
Gateways
...
When sending calls to the PSTN, the localized E.164 format is used tor both the callmg-
and the called-partv number. The fonnat of these numbers (especially ofthe called-party
number) can significant differ based on the location ofthe gateway (for example, various
international access codes in the United States [0111 versus the LTJ [00]).
•
Phones
, ,
,
„ ,
,,,
When acall from an internal phone is sent to another internal phone, the call should be
received at the phone with both the calling and called number using internal director}
numbers Because this fonnat is also used by globalized call routing, there ,s no need ior
localized call egress in this case. When acall from an external caller is sent to an internal
phone! most users (especially users in the United States) prefer lo see the calhng number ,n
localized fonnat. For example, acall from the local area code should be displayed with
sev en digits. The called number is the director}' number and usually is not displayed al the
phone.
It is evident from the figure that, in several situations, the numbers that are provided at call
ingress do not conform to the normalized format to be used for call routing. 1he same situation
occurs with cal! egress, where the normalized fonnat is not always used when the call is being
delivered. Therefore, localized call ingress has to be normalized (that is. globalized), and
globalized fonnat has to be localized at call egress.
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Globalized Call Routing—Three Phases
The figure provides an overview about the three phases that are involved in globalized call
routing.
On the left side ofthe figure, call ingress is illustrated by two types ofcall sources:
• External callers: Their calls are received by Cisco Unified Communications Manager
through agateway or tmnk. In aPSTN gateway, calling- and called-party number are
usuallv provided in localized E.164 format.
• Internal callers: Their calls are received from internal phones. Ifcalls to internal
destinations (for example, phone to phone), calling- and called-party numbers are typically
provided as internal directory numbers. Ifcalls to extemal destinations (for example, phone
to PSTN), the calling number is the directory number (at call ingress time) and the called
number depends onthe local dial rules for PSTN access. These dial rules can differ
significantly per location.
The center ofthe figure illustrates the standards that are defined for normalized call routing As
mentioned earlier, because most calls use global E.164 format, this process is also referred to as
globalized call routing. Here arethe defined standards:
•
•
External to internal
—
Calling-party number: E.164
—
Called-party number: directory number
External to external (if applicable)
—
Calling-party number: E.164
Called-party number: E.164
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Globalized Call-Routing Overview
(Cont.)
Afterthe call is routed and path selection has been
performed, theegress device can change from
globalized to local format;
• Applies to calling- and called-party number for all calls routed
to gateways and trunks
• Applies to calling-party number for calls routed from
gateways and trunksto phones
* Does not apply to called-party number for all calls routed to
phones (as internal directory numbers are the standard
formatfor internal destinations)
• Does not apply to calling-party numbers for calls routed from
internal to internal (as internal directory numbers are the
standard format in this case)
Alter the call has been routed and path selection (ifapplicable) has been performed, the
destination dev ice might need to change the nomialized numbers lo local format. 1his situation
is referred to as localized cal! egress.
Localized call egress applies tothese kinds ofnumbers:
• Calling- and called-partv numbers for calls that are routed to gateways and trunks: If
the PSTN or the telephonv svstem on the other side of atrunk docs not support globalized
call routine the called- and calling-party numbers need lo be localized from global format.
An example would be to change the called-party number +494012345 to 011494012345
before sending the call out tothe PSTN in the United Slates.
• Calling-party numbers for calls that are routed from gateways or trunks to phones:
fhe nhone user mav want to see caller IDs in alocal format rather than aglobal format, for
example, auser at aU.S. phone may want to see PSTN callers who are located in the same
area code as 7- or 10-digil numbers and not with »1 followed by 10 digits.
1ocalized call egress is not needed for the called-party number ofcalls that are routed to
phones because internal director}' numbers are the standard (normalized) formal for internal
destinations (regardless ofthe source ofthe call). These numbers might have been dialed
ditTerentlv initially. In that case, however, this localized call ingress was normalized before call
routing.
Localized call egress is also not required for the calling-party number of internal calls (internal
to internal), because typically the standard for ihe calling-parly number ot such calls is lo use
internal directory numbers.
Note
When internal directory numbers are not unique (for example, when there are overlapping
directory numbers at various sites), the called- and calling-party numbers of internal calls
can be globalized at call mgress and localized at call egress just like external calls.
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Assuming that all internal directory numbers are unique, this format is the most common:
• Normalized called-party numbers: K. 164 global format with a+prefix is used for
extemal destinations. Therefore, called-number normalization is achieved by globalization
Internal directory numbers are used for internal destinations. Normalization is achieved bv'
stripping or translating the called number to internally used directory numbers.
• Normalized calling-party numbers: E.164 global format with a+prefix is used for all
calling-party numbers, except for those formats ofcalls from internal to internal Such
purely internal calls use the internal directory number for the calling-party number.
Ifsources of calls (users at phones, incoming PSTN calls at gateways, calls received through
tmnks. and so on) do not use normalized format, the localized call ingress needs to be
normalized before being routed. This principle applies to all received calls (coming from
gateways and tmnks as well as from phones), and it applies to both the calling- and called-partv
Note
Except for the mentioned internal calls (where the destination is adirectory number and in
the case of an internal source, the source is a directory number), all numbers are normalized
to E.164 global format. Therefore, this call-routing implementation model is referred to as
globalized callrouting.
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Globalized Call-Routing Overview
Globalized call routing simplifies the implementation of international Cisco Communications
Manager depknments.
Globalized Call-Routing Overview
Globalized call routing simplifies implementation of
International multisite deployments:
• Gail routing ss based on globalised nnmDers
• Called number: E.164 format with + prefix (except for calls
to internal destinations)
- Calling number E.164 format with +prefix (except for
calls from internal to internal)
• If sources ofcalls (users at phones, incoming PSTN calls at
gateways, etc.) use different format, their localized call
ingress needs to be globalized before being routed.
Applies to phones aswell asto gateways and trunks
Applies to called- and calling-party numbers
Globalized call routing consists ofihe following main components:
. Normalization: Localized call ingress (that is, local dial rules) is normalized to acommon
fonnat (E 164 with +prefix). This action would not be necessary ifall endpoints and users
dialed destinations onlv in normalized format (like with 4dialing from directories).
However it is serv unlikelv that only +dialing would be pennilted. When manually typing
anumber, users still want to follow their local dial rules. Therefore, normalization of tins
input is required.
. Routing that is based on global numbers: When all dialed numbers have been globalized
to L164 format, local dial rules do not apply during call routing. They were relevant only
during call ingress. All call routing occurs based on numbers in globalized format.
. Ioealization of numbers before handing off the call: After call routing and path
selection, the local dial rules ofthe selected device have to be used, for example when a
user calls an international PSTN destination through aU.S. gateway 011 has to be prefixed
to the number, while in Europe 00 is commonly used. Localized call egress ,s implemented
at the gatewav or tmnk that routes the call out oi the cluster.
In general, call routing is based on normalized numbers. As mentioned earlier, the most
Lion format that if used is the globalized format, where both the called- and caling.part}
numbers are globalized for calls that are not exclusively internal. For such internal calls
^8^00 numbers can be used as long as they are unique. If, for example, over apping
die'on numbers are used, then it is common to use the globalized format also or rouing
such int'ersite calls. End users do not have to dial phones at other sites by using the ET 64
formal, but their localized ingress (typically including site codes) will be globalized before the
call is routed.
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Dial Plan Components in Multisite Deployments
The table shows dial plan components and their configuration elements inCisco Unified
Communications Manager and in Cisco IOS gateways.
Dial Plan Components in Multisite
Deployments
Dial Plan Component
Cisco IOS Gateway
Cisco Unitied
Communications Manager
Endpoint addressing*
SilSS6^"'
<**"Mnic
POTS, dialpeers
Dfrectorynumtter
Cal routing and path
„ ,
Dialpeers
Routepatterns,routegroups,
route Isfs, translationpatterns,
Voicetranslation profiles
using prefix, dlgjt-«trlp.
forward-digit*, and
Translationpatterns, route
patterns, route lists, significant
digits, catted-amicaMhg-parfy
transformations, fccoming
caibd-artdcailng-pafty
selection
partitions. CSSs
Digit manipulation
num-exp commands
Caling privieges
COR and COR Hats
Cal coverage
appicaa'ons, ephone hunt
raalpeers/catr""" " ~~
Partitions, CSSs, lime
schedules, time periods, FACs
Line groups, hunt lists, hunt
groups
Note
Alt these elements have been discussed in other courses, such asImplementing Cisco
Voice Communications and QoS (CVOICE) and Implementing Cisco Unified
Communications Manager, Part 1(CIPT1). However, information onhow to use these
elements to implement a dial plan in multisite deployments is provided in a separate lesson
of this module.
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Least Cost Routing, tail-end hop-off (TEHO), and PSTN backup: Can be implemented
b> appropriate call routing and path selection that is based on priorities.
Globalized call routing: In this dial plan implementation, all received calls arc normalized
toward a standardized fonnal. fhe formal that isused in call routing isglobalized format,
because all numbers are represented in E. 164 format with a4 prefix, 'fhe process of
nomializine the numbers as dialed by end users (localized ingress) istherefore also referred
to as globaUzation. Once the localized input has been globalized during ingress, the call is
routed based on globalized numbers. After call routing and path selection, the called
number is localized during call egress, depending on the selected egress device.
Multisite Deploymenl Implementation
©2010 Cisco Systems, Inc
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Dial Plan Solutions
This topic provides an overview about dial plan solutions in multisite deployments
Dial Plan Solutions for Multisite
Deployments
• Overlapping and nonconsecutivenumbers
- Solved by access code and (unique) site code.
- Allows routing independent directory numbers.
- Appropriate digit manipulation required.
• Variable -length numbering
- Dial string length determined by timeout or use of# key.
- Overlapsendingand receiving.
* DID ranges and PSTN addressing
- Use ofinteractive voice response applications (AA B-ACD etc
or attendant required rf no DID numbers.
- Directory numbers appended to PSTN number (with variablelength dial plans—if supported by PSTN).
• Numberpresentation (ISDN TON)
• LeastCost Routing, TEHO, PSTN backup
- Call routing and path selection based on prioritized paths.
• Globalized call routing
Dial plan issues in multisite deployments can be solved in the following ways:
• Overlapping and nonconsecutive numbers: Solved by implementing access codes and
site codes tor intersite dialing. This approach allows call routing that is independent of
directorv numbers. Appropriate digit manipulation (removal ofsite codes in called number
of outgoing calls) and prefixing ofsite codes in calling number ofincoming calls are
required.
• Variable-length numbering plans: Dial string length is determined by timeout Overlap
sending and receiving is enabled, allowing dialed digits to be signaled one by one instead
of being sent as one whole number.
• Direct inward dialing (DID) ranges and E.164 addressing: Solutions for mapping of
internal directory numbers to PSTN numbers include DID, use ofattendants or interactive
voice response (IVR) applications to transfer calls, and extensions that are added to PSTN
numbers invariable-length numbering plans.
' 5ITT1
'SDN (,ype °fofnumbers
number'orthatT0N>:
Di8jt manipulation
that ,s basedTmbT^reSeu!aIi0n
on TON enables the '"standardization
are signaled
using
different TONs.
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Mobility Solutions
This subtopic provides an overv iew about mobility solutions that solve issues that are the result
ofroaming users and devices, and multiple telephones (office phone, cell phone, home phone,
and so on).
Mobility Solutions
When users or devices roam, the resulting limitations in
features can be solved by mobilitysolutions:
• Cisco Device Mobility
- Solves issuesthat result from roaming devices (region, location,
SRST reference, AAR group, CSS. etc.)
- Makes Cisco Unified Communications Manager aware of
physical location of IP phone (usually software phone such as
Cisco IP Communicator)
• Cisco Extension Mobility
Solves issueofmissing personal IPphone setting that results
from using a different IPphone in another office (directory
number. CSS, etc.)
- Allows users to log in to IP phone and get personal configuration
applied to currently used IP phone
» Cisco Unified Mobility
Solves issues ofhaving different phones (office IP phone, cell
phone, home officephone, etc )
Allows usersto be reached by a single number, independent of
thephonethat is actually used
When users or de\ ices roam between sites, issues arise that can be solved by these mobility
solutions:
• Cisco De* ice Mobilitv: Solves issues that arc caused by roaming devices, including
invalid device configuration settings such as regions, locations, SRST reference. AAR
groups, calling search spaces (CSSs). and so on. The Cisco Device Mobility feature of _
Cisco Unitied Communications Manager allows device settings that depend on the physical
location ofthe device to be automatically overwritten ifthe device appears in adifferent
phvsical location.
. Cisco Extension Mobilitv: Solves issues that are the result of roaming users using shared
guest IP phones that are located in other offices. Issues include wrong directory number,
missing IP Phone Services subscriptions. CSS. and so on. Cisco Extension Mobility allows
users to log in to guest phones and to replace the configuration ot the IP phone with the II
phone configuration ofthe logged-in user.
• Cisco I'nified Mobilitv: Solves issues of having multiple phones and consequently
multiple phone numbers, such as an office phone, cell phone, home (office) phone, and so
on. Cisco Unitied Mobility allows users to be reached by asingle number, regardless ot the
phone that isactually used.
Note
Cisco Device Mobility and Cisco Extension Mobility will be discussed in detail in later
lessons of this course. Cisco Unified Mobility has been discussed in detail in the
Implementing Cisco Unified Communications Manager, Part 1(CIPT1) course
Multisite Deploymenl Implementation
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Automated Alternate Routing
The figure illustrates how AAR improves availability in multisite environments.
Automated Alternate Routing
AAR allows rerouting of calls over PSTN if not enough
bandwidth for VoIP calls:
• Alternate destination isderived from theextemal phone number mask and ,
prefix configured per AAR group.
• Individual destinations canbe configured perphone.
m
Cisco Unified
Communications
Manager
CAC Failure lo IP Phone
of User X (1009)
Ifa call over the IP WAN is not admitted by CAC, the call can be rerouted over the PSTN
using AAR. The AAR feature includes an option that allows the alternate number to be set per
IP phone. This option is also known as Call Forward No Bandwidth (CFNB) In the example
because the remote site does not have PSTN access, the call is not rerouted to the IP phone over
the PSTN (instead of over the IP WAN). It is alternately rerouted to the cell phone ofthe
affected user. AAR and CFNB improve availability in multisite environments by making it
possible to reroute on-net calls thatfailed CAC overthe PSTN.
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Using CFUR to Reach Users of Unregistered Software IP
Phones on Their Cell Phones
This subtopic describes how CFUR can be used to route calls lo the cell phones ofusers who
have shut down PCs that have asoftphone installed.
Using CFUR to Reach Users of Unregistered
Software IP Phones on Theif Cell Phones
When a user at the main site shuts down a laptop with
Cisco IP Communicator:
• The Cisco Unified Communications Manager ofthemain site does
not route callstothe affected IPphone directory number.
- CFUR allows routing toalternate numbers ofuser (for example, a
cell phone number)
Main Site
Communicator
PC Shutdown
Cell
1007 Unregistered
Phone
CFUR
9 1512 555-1999
E12 555-1999
_y
If amobile user has alaptop with asoftphone (for instance, Cisco IP Communicator) and shuts
down the laptop. CFUR can be used to forward calls placed to the softphone to the cell phone
of auser The user docs not have to set up Call Forward All (Cl'A) manually before closing the
softphone application. However, ifthe softphone is not registered, calls are forwarded to the
cell phone ofthe user. This action is another application ofthe CFUR feature that improves
availabilitv in Cisco Unified Communications Manager deployments.
Note
This application isfor CFUR, which isnot related to SRST.
Multisite Deployment Implementation
© 2010 Cisco Systems. Inc.
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Using CFUR to Reach Remote-Site IP Phones Over the PSTN
During WAN Failure
This subtopic describes how you can use CFUR to route calls to IP phones at remote locations
during IP WAN failure.
Using CFUR to Reach Remote-Site IP Phones
Over the PSTN during WAN Failure
The remote site lost connectivity to main site. Phones
are registered to remotegateway:
• Cisco Unified Communications Manager for main site does
not route calls tothe affected IP phone directory numbers.
- CFUR allows routing to alternate numbers for affected
(unregistered) IP phones.
1
As discussed before, IP phones that are located at remote locations can use an SRST gateway as
abackup for Cisco Unified Communications Manager in case of IP WAN failure The gateway-
can use its local dial plan to route calls that are destined for the IP phones in the main site over
the PSTN. But how should intersite calls be routed from the main site to the remote site while
the IP WAN is down?
The problem in this case is that Cisco Unified Communicalions Manager does not consider anv
other entries in its dial plan ifadialed number matches aconfigured but unregistered directory'
number. Therefore, if users al the main site dial internal extensions during the IP WAN outage
their calls will fail (or go to voice mail). To allow remote IP phones to be reached from the IP '
phones at the main site, you can configure CFUR for the remote-site phones CFUR should be
configured with the PSTN numbers that are used at the remote site so that internal calls for
remote IPphones are forwarded to the appropriate PSTN number.
J
J
J
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J
Fallback for IP Phones: Fallback Mode
fhe figure illustrates the operation ofCisco Unified SRST when IP phones lose connectiv.h
tvith their primary- Cisco Unified Communications Manager.
Fallback for IP Phones: Fallback Mode
• Communication between Cisco Unified Communications
Manager and IP phonesis broken.
• IP phones register with local gateway (either SRST or Cisco
Unified Communications Manager Express in SRST mode).
Main Sile
Remote Site
Register
Cisco Unified
Communications
Manager
Remote
Gateway
When Cisco IP phones lose contact with Cisco Unified Communications Manager, they register
with the local Cisco Unified SRST router lo sustain the calI-processing capability that is
necessary to place and receive calls.
The Cisco Unified SRST gatewav automatically detects afailure, queries IP phones for
configuration, and automaticallv configures itself. The Cisco Unified SRST gateway uses
Simple Netuork-lnablcd Auto Provision (SNAP) technology to autoconfigure the branch
office router to prm ide call processing for Cisco IP phones that are registered with the router.
Cisco Unified Communications Manager Express in SRST mode can be used instead of
standard Cisco Unified SRST functionality. In this case. IP phones register with Cisco Unified
Communications Manager F.xpress when they lose the connection to their primary Cisco
Unified Communications Manager server. Cisco Unified Communications Manager F.xpress m
SRST mode provides more features than standard Cisco Unified SRSI.
Multisite Deployment Implementation
© 2010 Cisco Systems. Inc
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L
Fallback for IP Phones: Normal Operation
This subtopic describes how fallback for IP phones improves availability in amultisite
deployment with centralized call processing.
L
Fallback for IP Phones: Normal
L
Operation
• Remote IP phones are registeredwith Cisco Unified
Communications Managerover IPWAN.
L
• Cisco Unified Communications Manager controls IP phones.
I
Main Site
1
Cisco Unified
Communications
Manager
£
Remote
Gateway
1
Fallback for IP phones is provided by the Cisco Unified SRST feature and improves the
availability of remote IP phones.
AWAN link connects IP phones at aremote site to the Cisco Communications Manager at a
1
central site, which is the call-processing device. Ifthe WAN link fails, Cisco Unified SRST
enables the gateway to provide call-processing services for IP phones. IP phones register with
the gateway (which is listed as abackup Cisco Unified Communications Manager server in the
server group configuration ofthe IP phones). The Cisco Unified SRST obtains the
1
configuration ofthe IP phones and can route calls between the IP phones or out to ihe PSTN.
The figure illustrates normal operation ofCisco Unified SRST while the connectivity between
IP phones and their primary server (Cisco Unified Communications Manager) is okay:
' «f!T?e
'Pph°nes ^ reSis,ered with Cisco lJn'ted Communications Manager over the IP
WAN.
• Cisco Unified Communications Manager manages call processing for IP phones.
J
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MGCP Fallback: Fallback Mode
The figure illustrates operation of MGCP fallback in fallback mode-when the connectivity
to
the call agent (Cisco Unified Communications Manager) is lost.
MGCP Fallback: Fallback tVlode
• Communication between Cisco Unified Communications
Manager and MGCP gateway is broken.
• MGCP gateway falls back toits default call-control
application (H.323 or SIP).
Main Site
When the MGCP gatewav loses the connection to its call agent, it falls back to its detault callcontrol application (POTS. H.323. or SIP). The gateway now uses alocal dial plan
configuration, such as dial peers, voice translation profiles, and so on. Ilence. .1 can operate
independent ofits MGCP call agent. Without MGCP fallback, the MGCP gateway would not
be able to process calls when the connection to its call agent is lost.
Multisite Deployment Implementation
© ?010 Cisco Systems.
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MGCP Fallback: Normal Operation
This subtopic describes how MGCP fallback improves availability in amultisite environment.
MGCP Fallback: Normal Operation
• MGCP gateway isregistered with Cisco Unified Communications
Manager over IP WAN.
• Cisco Unified Communications Manager isthe MGCP call aqent
controlling the MGCP gateway.
MGCP gateway fallback is afeature that improves the availability ofremote MGCP gateways.
AWAN link connects the MGCP gateway ataremote site to the Cisco Unified
Communications Manager at acentral site, which is the MGCP call agent. Ifthe WAN link
fails, the fallback feature keeps the gateway working as an H.323 or SIP gateway and re-homes
back to the MGCP call agent when the WAN link becomes active again.
The figure illustrates normal operation of MGCP fallback while the connectivity lo the call
agent (Cisco Unified Communications Manager) is okay:
• The MGCP gateway is registered with Cisco Unified Communications Manager over the IP
WAN.
• Cisco Unified Communications Manager is the call agent ofthe MGCP gateway that is
controlling its interfaces. The gateway does not have (or does not use) alocal dial plan
because all call-routing intelligence is at the call agent.
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PSTN Backup
The figure illustrates how calls can use the PSTN as abackup in case oflP WAN failure.
PSTN Backup
Intersite calls are rerouted over the PSTN in the case of an IP
WAN failure.
Main Site
Cisco Unified
Communications
Manager
1001-1099
1001-1099
In the example, calls to the remote site arc configured to use the IP WAN first and then use the
PS 1N as a backup option.
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1-53
Availability
This topic describes solutions to availability issues in multisite deployments.
Availability Options
1 PSTN backup
1 MGCP fallback
1Fallback for IP phones:
- Cisco Unified SRST
- Cisco Unified Communications Manager Express
in Cisco Unified SRST mode
CFUR
AAR and CFNB
Mobilitysolutions:
- Cisco Extension Mobility
- Cisco Device Mobility
- Cisco Unified Mobility
These options can improve availability:
• Public switched telephone network (PSTN) backup: Use the public switched telephone
network (PSTN) as a backup for on-net intersite calls.
B ^GCP fallback: Configure an MGCP gateway to fall back and use the locally configured
plain old telephone service (POTS), H.323, or SIP dial peers when the connection to its call
agent .s lost. This approach enables you to effectively make the gateway by using alocallv
configured dial plan, which is ignored when the gateway is in MGCP mode.
' !?i,bcaCk f°r IP phonCS: IP Phones thal reSister °ver the IP WAN can have alocal Cisco
IOS SRST gateway that is configured as abackup Cisco Unified Communications Manager
server mtheir Cisco Unified Communications Manager group configuration When the
connection to the primary Cisco Unified Communications Manager server is lost thev can
reregister wuh the local Cisco Unified SRST gateway. Alternatively, Cisco Unified '
Communications Manager Express can be used in SRST mode, which provides more
features than standard Cisco Unified SRST.
• Call Forward Unregistered (CFUR): CFUR is acall-forwarding configuration of IP
phones that becomes effective when the IP phone is not registered.
• Automated alternate routing (AAR) and Call Forward No Bandwidth (CFNB)- AAR
cacSrFvp
T?Cd°Ver thCconfiguration
PS™ When ofcalls
over thewhich
IP WAN
are »oteffective
admi«ed by
£ ,J1 1S• a;al|-forwardin8
IP phones,
becomes
when AAR is used.
• Mobility solutions: When users or devices roam between sites, they can lose features or
Fx^lnPrh
l^ and^ati°nr,beCaUSe
°f' Change
" th£irsuch
actUal
^sicaI
loc*ion.these
Cisco
Extension Mobility
Cisco Dev.ce Mobility
can solve
issues.
In addition
features allow integration ofcell phones and home office phones by enabling reachability
on any device via a single (office) number.
1-52
Implementing Cisco Unified Communications Manager, Part 2(CIPT2) vS.O
«-^»auimy
©2010 Cisco Systems, Inc.
Preventing Too Many Calls by CAC
This subtopic describes different methods of limiting the number ofconcurrent calls by CAC.
Preventing Too Many Calls by Cai
Admission Control
CAC is used to limit the number of calls:
• Locations
Bandwidth limit applicable to components within a cluster.
Includes calls to and from gateways and trunks.
• RSVP-enabled locations
- Special implementation oflocations.
Allows calls to flow through two routers (RSVP agents).
- Call leg between routers uses RSVP to limit bandwidth.
• SIP precondition
Like RSVP-enabled locations but between SIPdomains
(e.g., two clusters interconnected via a SIP trunk).
« Gatekeepers
- Bandwidth control for calls between H.323 gatekeeper
zones.
Cisco Unified Communications Manager allows the number of calls to be limited by these
CAC mechanisms:
• 1ocations' Cisco Unified Communications Manager location-based CAC is applicable to
calls between two entities thai arc configured in Cisco Unified Communications Manager.
These entities can be endpoints such as phones or devices that connect to other call-routing
domains such as trunks or gateways. However. CAC applies to the devices that are part o
the Cisco Unified Communications Manager cluster, even ifthey represent an external call
routing domain (in case of trunks). If ingress and egress device are in different locations,
the maximum bandwidth that is configured per location is checked at both ends, t alls
within a location arc not subject tothe bandwidth limit.
. Resource Reservation Protocol (RSVP)-cnabled locations: RSVP is aspecial way ,i
configure locations. When RSVP is configured to be used bclween apair ol locations, the
audio streams fious through two routers, so-called RSVP agents. The cal eg between the
t«o RSVP agents is subject to Cisco IOS RSVP CAC. Like with standard locations, ingress
and egress devices arc both part ofthe Cisco Unified Communications Manager cluster.
. Session Initiation Protocol (SIP) Preconditions: SIP Preconditions ««»lut«jn Hke
RSVP-enabled locations except that it is designed for SIP trunks on y With SH
Preconditions, calls through aSIP trunk flow through alocal Cisco IOSrouter at each end
ofthe SIP trunk splitting the call into three call legs-just like with RSVP-enabled
locations. However, in this case the call is not within acluster but bclween clusters.
. Gatekeepers- Gatekeepers arc used in the 11.323 world and provide address resolution and
CAC funSs HI 323 gatekeepers can be configured lo limit the number ofcalls between
H.323 zones.
Mullisite Deployment Implementation
© 2010 Cisco Systems.
1-51
Alternatives to Multicast MOH from Branch Router Flash
This subtopic shows alternatives that you should consider when multicast MOH from branch
router flash cannot be used.
Alternatives to Multicast MOH from
Branch Router Flash
If multicast MOH from branch router flash cannot be
used, consider these alternatives;
- Using multicast MOH:
- Reduces the number of MOH streamssent overthe IP
WAN,
- Requires less bandwidth than unicast MOH.
• Using G.729 for MOH to remote sites:
- Putremote IP phones and MOH server into different
regions.
- Allow only G.729 between these two regions.
- These actions reduce the required bandwidth per MOH
stream.
Ifmulticast MOH from branch router fiash cannot be used (for instance, because the branch
router does not support the feature or does not have aCisco Unified SRST feature license) vou
can consider these alternatives:
• Using multicast MOH: When using multicast MOH over the IP WAN you can
significantly reduce the number of required MOH streams. Thus, less bandwidth is required
compared with multiple unicast MOH streams. The IP network, however, has to support
multicast routing for the path from the MOH server to the remote IP phones.
• Using G.729 for MOH to remote sites: Ifmulticast MOH is also not an option (for
instance, because multicast routing cannot be enabled in the network), you may still be able
to reduce the bandwidth that is consumed by MOH. When you change the codec that is
used for the MOH streams to G.729 and you potentially enable cRTP on the IP WAN each
individual MOH stream requires less bandwidth and hence reduces the load on the WAN
link. The bandwidth savings are identical to the bandwidth savings that you achieve when
"f'y/29 and cRTP for standard audio streams, which was discussed earlier To use
U729 for MOH streams, you have to put the MOH server and the remote IP phones into
different reg.ons. and you need to limit the audio codec between these two regions to 8
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Implementing Cisco Unified Communications Manager, Pari 2(CIPT2) v8.0
©2010 Cisco Systems, Inc.
Multicast MOH from Branch Router Flash: Cisco IOS
Configuration Example
This subtopic shous the required Cisco IOS commands to enable multicast MOH from branch
router flash.
Multicast MOH from Branch Router Flash:
Cisco IOS Configuration Example
Cisco Unified
Com muni cation s
Manager
MOH
Configuration
DA 239 1 1 1
DP 16334
Ma" Hops
TTL 1
Main Site
_._
WAM.
call manage r-£a llbJ ck
max
ephone a
dn
Remote Site
1
1
ip source- add
ess
10.2 .2.2
moh moh-f. le.
multicast moh 239
1.1. 1 port
16384
In the example, the name ofthe audio file on the branch router Hash is moh-lile.au. and the
configured multicast address and port number are 239.1.1.1 and 16384. respectively. Ihe
optional route command can be used to specify asource interface address lor the multicast
stream If no route option is specified, the multicast stream will be sourced from the configured
Cisco Unified SRST default address as specified by the ip source-address command under the
Cisco I'nificd SRST configuration (10.2.2.2 in this example). Note that you can stream only a
single audio file from flash and that you can use only asingle multicast address and port
number per router.
ACisco Unified SRST license is required regardless of whether the SRST functionality will
actually be used. The license is required because the configuration for streaming multicast
MOH from branch router flash is done in the SRST configuration mode and, even ,tSRSI
functionality will not be used, at least one IP phone (using the max-ephones command) and
one extension (using the max-dn command) must be configured.
Multisite Deployment Implementation
© 2010 Cisco Systems. Inc
1-49
5. The router at the remote site is configured as an SRST gateway. In its Cisco Unified SRST
configuration, multicast MOH is enabled with destination address 239 11Iand port
16384. The Cisco Unified SRST gateway streams MOH all the time (even if not in fallback
mode).
6. The IP phone listens to the multicast MOH stream that was sent from the Cisco Unified
SRST gateway to IP address 239.1.1.1, port 16384, and plays the received MOH stream.
At no time do MOH packets cross the IP WAN.
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implementing Cisco Unrfied Communications Manager, Part 2(CIPT2) .8.0
©20,0 Cisco Systems, Inc.
Multicast MOH from Branch Router Flash Example
The figure illustrates how multicast MOH from branch router Hash works.
Multicast MOH from Branch Router
Flash Example
When remote phone is put on hold Clko Unified Communicate Manager signals phone
to listen to DA 2391 1 1 DP 16384 andplay thereceived stream
Unified
Unified
Communicalions
Communications
Manager MOH Packets
Manager MOH
Packets Created
Dropped Here (TTL
Configuralion
Here by SRST MOH
Exceeded)
Identical MOH
DA 239 1 1 1
DP 163B4
Max Hops
TTL 1
In the example the Cisco Unified Communications Manager MOH server is configured for
multicast MOH with adestination (multicast group) address of239.1.1.1. the destination port
16384. anda max-hops TTI.value of 1.
The Cisco Unilied SRST gateway that is located at the remote silc is configured with the same
destination IP address and port number as the Cisco Unified Communications Manager MOH
server.
When a remote phone is put on hold, here iswhat happens:
1, According to the MRGL ofthe remote phone, the Cisco Unified Communications Manager
MOH server is used as the media resource for MOH.
2. Cisco Unified Communications Manager signals the IP phone to receive MOH on IP
address 239.1.1.1-port 16384.
} The Cisco Unified Communications Manager MOH server sends multicast MOH packets
toIP address 239.1.1.1. port 16384. with a TTL value of 1.
4 The router that is located at the central site drops the multicast MOH packet that is sent out
by the Cisco Unified Communications Manager MOH sewer because TTL has been
exceeded.
Multisite Deployment Implementation
© 2010 Cisco Systems, Inc
1-47
Instead of setting the max-hops parameter for MOH packets to 1,you can useone of these
methods:
•
Configure an access control list (ACL) on the WAN interface: Configure an ACL onthe
WAN interface at the centralsite to disallow packetsthat are destined to the multicast
group address or addresses from being sent out the interface.
Note
Depending on the configuration of MOH in CiscoUnified Communications Manager, a
separate MOH stream for each enabled codec is sent per multicast MOH audio source The
streams are incremented either based on IP addresses or based on port numbers
(recommendation is per IP address). Assuming that one multicast MOH audio source and
G.711 a-law, G.711 mu-law, G.729, and the wideband codec are enabled, there will be four
multicast streams Makesure that allof them are included in the ACL to prevent MOH
packets from being sent to the IP WAN.
•
Disable multicast routing on the WAN interface: Donotconfigure multicast routing on
the WAN interface, to ensure that multicast streams are not forwarded into the WAN.
When you use multicast MOH from branch router flash. G.711 has to be enabled between the
Cisco Unified Communications Manager MOH serverandthe remote IP phones. This action is
necessary because the branchSRST MOH feature supportsonly G.711. Therefore, the stream
that is set upby Cisco Unified Communications Manager in thesignaling messages also has to
be G.711. Because the packets arenot sentacross the WAN, configuring the high-bandwidth
G.711 codec is not a problem as long as it is enabledonly for MOH. All other audio streams
(such as calls between phones) that are sent over the WAN should use the low-bandwidth
G.729 codec.
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
©2010 Cisco Systems, Inc.
Multicast MOH from Branch Router Flash
This subtopic explains how you can use multicast MOH from branch router fiash toreduce the
bandwidth that is required on the IP WAN.
Multicast MOH from Branch Router
Flash
Multicast MOH from branch router flash is a feature for multisite
deployments with centralized call processing:
• Mworks only with multicast MOH
• It is based on MOH capabilities of SRST
• Cisco IOS SRST gateway is configured lor multicast MOH
MOH stream is sent independent of SRST stalus (even if no failover scenario)
• Cisco United Communications Manager and IP phone are not aware that SRST
gateway is invoked
IP phone is configured lo use Cisco United CommunicationsManager multicast
MOH server
Cisco Unified Communications Manager MOH server is configured for max-hops 1
Cisco Unilied Communications Manager MOH server and branch router use same
multicast adoress arx) port number
Cisco UnifiedCommunications Manager signals its MOH server address and port
nurnberto IPphone
Cisco UnifiedComniumcat ons Manager MOHserver packets are dropped at WAN
router Because max-hops value TTL in IP packet header has bean exceeded
SRSTrouter generates multicast MOH stream with same multicast address and
portthai was used by Cisco Unified Communications Manager MOH
IP prwne listens lo signaled address and portand plays received stream
Multicast MOH from branch router fiash is a feature for multisite deployments that use
centralized call processing.
Ihe feature works only with multicast MOI 1and isbased on MOH capabilities ofCisco
Unified SRST. The Cisco IOS SRSTgateway is configured for multicast MOH and
continuously sends a MOH stream, regardless ofitsSRST mode (standby orfallback mode).
In fact, neitherCisco Unified Communications Manager nor the remoteIP phones are aware
that the Cisco Unified SRST gateway isinvolved. To them, itappears asthough a multicast
MOH stream has been generated by the Cisco Unified Communications Manager MOH server
and has been receded by the remote IP phones.
Therefore, the remote IPphones are configured to use the centralized Cisco Unified
Communications Manager MOII server astheir MOH source, fhe Cisco Unified
Communications Manager MOH server isconfigured for multicast MOH (mandatory), and the
max-hops \alue in the MOI Iserver configuration is set to 1for the affected audio sources. The
max-hops parameter specifies the Time to Live (TTL) value that isused in the IP header ofthe
RTP packets, "fhe Cisco Uni lied Communications Manager MOI 1server and the Cisco IOS
SRST gatewav that is located at the remote site have to use the same multicast address and port
number for their streams. This way, MOH packets that are generated bythe Cisco Unified
Communications Manager MOH server atthe central site are dropped by the central-site router
because ITI. has been exceeded. As a consequence, the MOII packets donotcross the IP
WAN. The SRST gateway permanently generates amulticast MOH stream with an identical
multicast IP address and port number. The IP phone simply listens tothis stream as it appears
to becoming from the Cisco Unified Communications Manager MOH server.
© 2010 Cisco Systems, Inc
Multisite Deployment Implementation
1-45
Mixed Conference Bridge
The figure shows how ahardware conference bridge in mixed mode can be used to reduce the
bandwidth that is required on the IP WAN.
Mixed Conference Bridge
Mixed (hardware) conference bridoes enable low-bandwidth
codecs to beused by remote conference participants.
Hardware
Conference Bridge
(Mixed)
Cisco Unified
Communications
Manager
-WAN-
Remote Site
In this example, ahardware conference bridge is deployed at the main site. The hardware
conference bndge is configured to support mixed conferences, in which members use various
codecs. Headquarters IP phones that join the conference can use G.711, while remote IP phones
canjoin the conference using a low-bandwidth codec.
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Implementing Cisco Unified Communications Manager, Part 2(CIPT2) v8.0
>2010 Cisco Systems, Inc.
Note
Calls between IP phones at headquarters and remote IP phones do not require a
transcoder. They simply use the best allowed codec that is supported on both ends. G.729.
Atranscoder is invoked only when the two endpoints of acall cannot find acommon codec
that is permitted by region configuration. This principle is illustrated in this example. The
remote IP phones (which support G.711 and G.729) are not allowed to use GV11 over the
IP WAN and the headquarters voice-mail system and software conference bridge do not
support G729 Cisco Unified Communications Manager detects this problem that is based
on its region configurations, and the capability negotiation that is performed during call setup
signaling identifies the need for a transcoder.
.
Bandwidth 8kb/s between BR and XCODER: This bandwidth ensures that the RTF
streams between remote IP phones and the transcoder. which are sent over the IP WAN. do
not use (J.711.
Bandwidth 64 kb/s between headquarters and XCODER: This bandwidth is required in
order for the G.711 -onh devices at headquarters lo be allowed to send G.711 to the
transcoder.
Multisite Deployment Implementation
© 2010 Cisco Systems, Inc
1-43
Guidelines for Transcoder Configuration
When implementing transcoders in order to allow G.711-only devices to communicate with
remote IP phones using G.729. you need to consider the following guidelines.
Guidelines for Transcoder Configuration
Cisco Unified CommunicationsManager transcoder support
requires the following configuration:
• Cisco Unified Communicalions Manager supports only hardware transcoder
resources.
- Hardware transcoding media resource is requred.
- Transcoding media resource has lo be configured h Cisco IOS router.
- Transcoding media resource has to be added to Cisco Unified
Communications Manager.
• IP phones and G.711-only devices (such as third-party voice mail or software
conference bridges) in headquarters are pui into a dedicated region (e.g.,
headquarters).
- Remote IP phones are put into a dedicated region (e.g., BR).
• Transcoders are put into a dedicated region (e.g., XCODER).
• Limitaudio bandwidth between regions headquarters and BR to 8 kb/s.
• Limitaudio bandwidth between regions headquarters and XCODER to 64
kb/s.
• Limitaudio bandwidth between regions XCODER and BR to 8 kb/s.
As a first step, jou need to implement the transcoding media resource. CiscoUnified
Communications Manager does not support software transcoding resources. Therefore, the only
option is to use a-hardware transcoding resource by first configuring the transcoder at the Cisco
IOS router and then adding the transcoder toCisco Unified Communications Manager.
The second step isto implement regions in a way that only G.729 ispermitted onthe IP WAN,
and the transcoder can be used ifrequired. To do so, you place all IPphones and G.711-only
devices, such as third-party voice-mail systems orsoftware conference bridges that are located
in the headquarters, in one region. You place remote IPphones in another region (called, for
example, branch, or BR). Thetranscoding resource is putinto a third region (called for
example. XCODER).
Now the maximum codec for calls within and between regions have tobe specified as follows:
•
Bandwidth 64 kb/s within BR: This bandwidth allows local calls between remote IP
phones to use G.711.
•
Bandwidth 64 kb/s within headquarters: This bandwidth allows local callswithin the
headquarters to use 0.711. These calls are not limited to calls between IP phones. They
also include calls to the G.711-only third-party voice-mail syslem orcalls that use the
G.711-only softwareconference bridge.
•
•
Bandwidth 64 kb/s within XCODER: Because this region includes only the transcoder
media resource, this setting isnot relevant since there are no calls within this region.
Bandwidth 8 kb/s between BR and headquarters: Thisbandwidth ensures that calls
between remote IP phones and headquarters devices (such as IP phones, software
conference bridge, and voice-mail system) do not use G.711 as the bandwidth for calls that
traverse the IP WAN is limited.
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Implementing Cisco Unified Communications Manager, Part 2 (CIPT2) u8.0
52010 Cisco Systems, Inc.
Transcoders
The figure illustrates how you can use transcoders to reduce the bandwidth that isrequired on
the IP WAN.
Transcoders
Ifa device supports G 711 only, transcoders enable low-bandwidth
codecs to be used over the IP WAN
In the example, a third-part; voice-mail system that supports only G.711 is deployed atthe
main site. One Cisco UnifiedCommunications Manager server is providing a software
conference bridge (which also supports G.711 only). Ifremote phones are configured lo use
G.729 over the IPWAN. they cannot join conferences oraccess the voice-mail system. To
allow these IP phones touse G.729 and toaccess the G.711-only services, you deploy a
hardware transcoder at the main site.
Remote IP phones now send G.729 voice streams to the transcoder over the IP WAN. The
transcoder changes the stream to G.711 and then passes iton to the conference bridge orvoicemail sNStem.
)2010 Cisco Systems, Inc
Multisite Deployment Implementation
1-41
Codec Configuration in Cisco Unified Communications
Manager
The codec that is used for a call isdetermined by the region configuration in Cisco Unified
Communications Manager.
Codec Configuration in Cisco Unrfied
Communications Manager
Thecodec that will be used depends on the region
configuration in Cisco Unified Communications Manager;
• Each region is configured with a maximum audiobandwidth per call:
- Within the configured region
- Toward specific other regions (manually added)
Toward all otherregions (which havenotbeen manually added)
• Region is assigned to a device pool.
• Device pool is assigned to a device.
• Whichcodec actuallygets used depends on the capabilitiesof the
two devices:
- Best codec is the one that is supported by both devices and
does not exceed bandwidth requirements of codec that is
permitted in the region configuration.
- If devicescannotagree on a codec, a transcoding deviceis
invoked.
Each region in Cisco Unified Communications Manager isconfigured with the maximum audio
bandwidth requirements to be used per call:
•
Within the configured region
•
Toward a specific other region (manually configured)
•
Toward all other regions (not manually configured)
Regions are assigned to device pools (one region per device pool), and adevice pool is
assigned to each device. Which codec isactually used depends on the capabilities ofthe two
devices that are involved in the call. The assigned codec is the one that is supported by both
devices and does not exceed the bandwidth requirements ofthe codec that is permitted in
region configuration. If devices cannot agree on a codec, a transcoder is invoked.
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Implementing Cisco Unrfied Communications Manager, Part 2(CIPT2) v8.0
>2010 Cisco Systems, Inc.
Low-Bandwidth Codecs and RTP-Header Compression
The figure illustrates the effect of using RTP-header compression to conserve bandwidth on the
IP WAN.
Low-Bandwidth Codecs and
RTP-Header Compression
The use of a low-bandwidth codec and cRTP reduces
the bandwidth requirements of a call on a WAN link.
82.4
kb/s
EuHaiEIS2IuilE5Z3
6
1 20 I
8
I 12 1 ieo
Cisco Unified
Communications
Manager
Remote Site
In the example, a \oice packet for a call that has default settings (G.711 codec and a 20-ms
packetization period) is being passed along aFrame Relay link. The frame has atotal size ot
206 Bcomprising 6Bof frame Relay header. 20 Bof IP header. 811 of UDP header. 12 Bof
RTP header, and 160 Bofdigitized voice, fhe packet rate is50 packets per second (p/s).
resulting ina bandwidth need of 82.4 kb/s.
When %ou use cRTP and change the codec toG.729. the required bandwidth changes as
follows: The frame now has atotal size of28 or30 Bper frame comprising 6bytes ofFrame
Relay header. 2or 4Bof cRTP header (depending on whether the UDP checksum is
preserved), and 20 Bof digitized, compressed voice. The packet rate is still 50 p/s (because the
packetization period was not changed), resulting in bandwidth needs of 11.2 or 12 kb/s.
Seven G.729 calls with cRTP enabled require less bandwidth than one G.711 call without cRl'P
(assuming that cRTP is used without preserving the UDP checksum).
Note
While the audio codec configuration affects the end-to-end path, cRTP only affects WAN
links where cRTP isenabled RTP header compression isconfigured on a per link basis.
© 2010 Cisco Systems. Inc
Multisite Deployment Implementation
1-37
Deploying transcoders or mixed conference bridges: If low-bandwidth codecs are not
supported by all endpoints, you can use transcoders sothat low-bandwidth codecs can be
used across the IP WAN. Then have the voice stream transcoded to G.711. For conferences
with local members using G.711 and remote members using low-bandwidth codecs, you
can deploy mixed conference bridges (hardware only) that support members with various
codecs.
Deploying local music on hold (MOH) servers or using multicast MOH from branch
router flash: Deploying local MOH servers means thatCisco Unified Communications
Manager servers have to be present at each site. In centralized call-processing models in
which this requirement does not apply, itisrecommended that you use multicast MOH
from branch router flash. This approach eliminates the need ofstreaming MOH over the IP
WAN. Ifthis approach isnot an option, you should use multicast MOH instead ofunicast
MOH to reduce the number of MOH streams that have to traverse theIPWAN. Multicast
routing should be enabled in the network in order for multicast MOH function properly.
Limiting the number ofvoice calls using CAC: Use CAC to avoid oversubscription of
WAN bandwidth by too manyvoice calls.
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Implementing Cisco Unified Communications Manager, Pari 2(CIPT2) v8.0
©2010 Cisco Systems. Inc.
Solutions to Bandwidth Limitations
"fhis topic describes solutions to bandwidth limitations.
Options to Reduce WAN Bandwidth
Bandwidth in the IP WAN can be conserved by:
• Using low-bandwidth codecs over the IP WAN
Fordevices requiring G.711, deploytranscoders.
For remote conference participants, deploy mixed
conference bridges or transcoders.
• Using RTP-header compression
• Deploying local annunciators or disabling remote
annunciators
• Deploying local conference bridges
• Deploying local MTPs (ifrequired)
• Deploying transcoders or mixed conference bridges (if
needed)
• Deploying local MOH server or using multicast MOH from
branch router flash
• Limiting numberofvoice calls by using CAC
You can conserve bandwidth on the IP WAN by using these methods:
• Using low-bandwidth codecs: When you use low-bandwidth (compression) codecs, such
as G.729. the required bandwidth for digitized voice is 8kb/s. compared to the 64 kb/s that
is required b\ G.71! (Layer 2overhead not considered).
• I. sing RTP-header compression: When using RIP-header compression (compressed
RTP. orcRTP). vou can compress the IP. User Datagram Protocol (UDP). and RTP header
to 2or 4B(depending on whether the UDP checksum is preserved), compared lo the 40 B
that Urequired bv these headers ifcRTP is not used. It is enabled per link. It can be
selecmeh used on a slim WAN link (in general, below 768 kb/s) and does nol need lobe
enabled end-to-end across all WAN links.
• Deploying local annunciators or disabling remote annunciators: Ifspoken
announcements are not required, you can disable the use ofannunciators for IP phones that
do not have a local annunciator. Otherwise, you can deploy local annunciators. Cisco
Unilied Communications Manager supports annunciators that are running only on Cisco
Unilied Communications Manager servers (provided by the Cisco IP Voice Media
Streaming Application service). Therefore, you can implement local annunciators only if
\ou deploy alocal Cisco Unified Communications Manager cluster or ifyou are using
clustering o\er the IP WAN.
• Deploving local conference bridges: If you deploy local conference bridges, the IP WAN
is not used ifall conference members are at the same site as the conference bridge.
m Deploving local Media Termination Points (MTPs): If MTPs are required, you can
deploy them locally at each site to avoid the need lo cross the IP WAN when using Ml P
services.
© 2010 Cisco Systems. Inc.
Multisite Deployment Implementation
Local vs. Remote Conference Bridges
The figure illustrates how local conference bridges can keep voice traflic oft*the IP WAN.
Local vs. Remote Conference Bridges
A local conference bridge at the remote site keeps RTP streams
away from the IP WAN for conferences hosting only remotely
located participants. Use MRGLs to control which phones use
which conference bridge.
You can use the same solution for MTPs.
Cisco Unified
Communication a
Manager
As shown in the figure, if a local conference bridge is deployedat the remotesite, it keeps
voice streams off the IP WANfor conferences in which all members are physically located at
the remote site. You can implement the same solution for MTPs. MRGLs specify which
conference bridge (or MTP)should be used and by which IP phone.
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Implementing Cisco Unified Communications Manager, Part2 (CIPT2) v8.0
©2010 Cisco Systems, Inc.
Disabled Annunciator
The figure shous howyou can conserve bandwidth on the IP WAN by sendingdisabling
annunciator streams to remote phones.
Disabled Annunciator
In multisite deployments with centralized call processing, MRGLscan
be used to disable annunciator streams to the remote site.
In multisitedeployments with distributed call processing, local
annunciators can be used at each site
C'Sco Unified
Communicalions
Manager
If announcements should not be sent over the IP WAN, Media Resource Group Lists (MRGLs)
canbe used so thatremote phones do not have access lo the annunciator media resource.
Note
Because not every call requires annunciator messages, and because the messages are
usually rathershort, the bandwidth that should be preserved by disabling the annunciator is
marginal.
©2010 Cisco Systems. Inc
Multisite Deployment Implementation
QoS Advantages
QoS can improve the Quality of Voice (QoV) calls by giving priority to RTP packets.
QoS Advantages
With QoS enabled, voice traffic has absolute priority
over other traffic.
• Prevents jitter caused by variable queuing delays
• Ensures enough bandwidth for signaling
• Prevents packetloss caused bytail dropsin queues
Voice
(Highest)
Voice is always served first
Data [High, Such as
IP Precedence: 4)
Data (Medium, Such
as IP Precedence: 2)
Data (Low, Such
as Precedence: 0}
With QoS enabled, voice traffic is given absolute priority queuing ("PQ" in the figure) over all
other traffic. This approach prevents jitter, which is caused by variable queuing delays. It also
prevents lost voice packets, which are caused by tail drops that occur when buffers are
complete. To avoid the complete blocking ofother traffic, you should limit voice bandwidth.
The number ofvoice calls should also be limited by CAC so that there isnot more voice traffic
than there is bandwidth that has been reserved for it.
Finally, to ensure proper service for voice calls, you should configure QoS to guarantee a
certain bandwidth for signaling traffic. Otherwise, despite the fact that the quality ofactive
calls may be okay, calls cannot betorn down, and new calls cannot beestablished.
Note
QoS is not discussed further in this course. For more information, refer to the Implementing
Cisco Voice Communications and QoS (CVOICE) course.
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) 2010 Cisco Systems, Inc.
QoS
This topic describes how quality ofservice (QoS) can solve voice quality issues.
QoS Review
QoS allows certain
communication flows to be
processed with higher
priority than others:
i
Traffic is identified.
'.-.
Traffic is divided into
Voice Mission-Critical Best-Effort
classes.
,;>
QoS policy is applied per
class.
Voice M_•
• il Best-Effort
Away* Laaaa^
irst
QoS refers to the capabilit\ ofanetwork lo provide belter service to selected network traffic.
The primarv goal of QoS is 10 provide better service, including dedicated bandwidth, controlled
jitter and latency {required bv some real-lime and interactive traffic), and improved loss
characteristics, by giving priority lo certain communication flows. It is also important to make
sure that providing priority for one or more flows docs not make other flows fail.
Fundamental. QoS enables vou to provide better service to certain flows. You can provide
better sen ice bv cither raising the priority ofaflow or limiting the priority ol another flow.
Some of QoS mechanisms are congestion management, congestion avoidance, and link
efficiencv
When you implement QoS. the implementation is split into three major steps:
•
Traffic is identified (voice, signaling, data, and so on).
• Traffic is div ided into classes (real-time traffic, mission-critical traffic, less important
traffic, and soon).
• QoS policy is applied per class, specifying how to serve each class.
Multisite Deployment Implementatii
© 2010 Cisco Systems. Inc
1-33
Multisite Deployment Solution Overview
This topic provides an overview ofsolutions to issues that arise in Cisco Unified
Communications Manager multisite deployments.
Multisite Deployment Solutions
SRST.
Main Site
Cisco Unrfied
Communications
QoS, CAC, RTP
PSTN
Header
Backup,
Compression,
MGCP
Local Media
Resources
Fallback
Remote Site
Manager
The figure illustrates amultisite deployment that incorporates the following solutions to
multisite deployment issues:
• Availability issues are solved by Cisco Unified Survivable Remote Site Telephony (Cisco
Unified SRST) and Media Gateway Control Protocol (MGCP) fallback.
• Quality and bandwidth issues are solved by quality ofservice (QoS), Call Admission
Control (CAC). Real-Time Transport Protocol (RTP)-header compression, and local media
resources.
• Dial plan solutions include access and site codes, as well as digit manipulation.
• Network Address Translation (NAT) and security issues are solved by the deployment ofa
Cisco Unified Border Element.
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>2010 Cisco Systems, Inc.
Lesson 2
Identifying Multisite
Deplovment Solutions
Overview
Amultisite deplounent introduces several issues that do not apply lo single-site deployments.
When implementing Cisco Unified Communications Manager in amultisite environment, you
need to address these issues, fhis lesson provides information on how lo solve issues that arise
neei
in multisite deployments.
Objectives
Upon completing this lesson, you will be able to describe solutions for multisite deplovment
issues.
Thisabilitv includes beingableto meet these objectives:
Describe solutions lo mullisite deployment issues
Describe how QoS solves quality issues inmultisite deployments
Describe solutions to bandwidth limitations in multisite deployments
Describe survivability and availability features in multisite deployments
Describe solutions for dial plan issues in mullisite deployments
Describe how aCisco Unified Border Element can solve NAT and security solutions in
mullisite deployments
Summary
This topic summarizes the keypoints thatwerediscussed in this lesson.
Summary
• Multisite deployment issues include quality issues, bandwidth
issues, availability issues, dial plan issues, and NAT and
security issues.
- When there is congestion, packets have to be bufferedor
they can get dropped.
• Bandwidth in the IP WAN is limited and should be used as
efficiently as possible.
• In a multisite deployment, some services depend on the
availability ofthe IP WAN.
1Amultisite dial plan has to addressoverlapping and
nonconsecutive numbers, variable-length numbering plans
DID ranges, and ISDN TON, and itshould minimize PSTN
costs.
When CiscoUnified Communications Manager and IP
phonesneed to be exposedto the outside, they can be
subject to attacks from the Internet.
References
For additional information, refer to these resources:
• Cisco Systems. Inc. Cisco Unified Communications System 8.x SRND, April 2010.
hup://\vww.eisco.C(>m/en/US/docs/voice_ip_comm/cucni/srnd/8x/uc8x.html
• Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0(1). February 2010.
http://uv\u .cisco.com/en/US/docs/voice.._ip_.comm/cucm/admm/8 0 l/ccmcfg/bccm-801
cm.himl
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implementing Cisco Unified Communications Manager, Pari 2(CIPT2) v8.0
©2010 Cisco Systems, Inc.
Example: NAT Security Issues
fhe figure illustrates the private IP addresses ofCisco Unified Communications Manager
server and the IP phone that is being translated to public IP addresses.
Example: NAT Security Issues
Cisco Unilied Communications Manager, orCisco Unified Communicalions Manager
Express, and IPphones aremade accessible from theInternet by NAT
Company A
Company A
Company B
Company 8
Pnvate IP
Public IP
Public fP
Pnvate IP
10.0.0.0(8
'
Public IP A
Company A
Cisco Unified
Communications
Manager
vateIP Network
"""
Private IPNetwork
100 00 Attacks canbedirected against Unified Communications 1000O
Manager/Cisco Unified Communications Manager
Express and IP phones
In the example, both Companv Aand Company Buse IP network 10.0.0.0/8 internally. For the
companies to communicate over the Internet, the private addresses are translated to public IP
addresses. Companv Auses public IP network A. and Company Buses public IP network B.
All Cisco Unified Communications Manager servers and IP phones arc reachable from the
Internet and communicate with each other.
© 2010 Cisco Systems.
Multisite Deployment Implementation
1-29
NAT and Security Issues
This topic describes NAT and security issues in Cisco Unified Communications Manager
multisite environments.
NAT and Security Issues in Multisite
Environments
• In single-site deployments, Cisco Unified Communications Manager
and IPphonesusually do notrequire access to public IPnetworks:
- NATis not required.
- Not reachablefrom theoutside,
- Notsubjectto attacksfromoutside.
1 In multisite deployments, VPN tunnels can be used:
- Requires gateway configuration ateach site.
- Allows only intersitecommunication.
- Blocks accessto and from outside (unless traffic istunneled).
VPNs cannot be used in some cases:
- Connections to ITSPsor domains underdifferent administration.
- NAT required: Cisco Unified Communications (Manager and IP
phones exposed to the outside.
- Cisco Unified Communications Manager and IP phones are
subject to attacks.
In single-site deployments. Cisco Unified Communications Manager servers and IP phones
usually use private IP addresses, since there is no need to communicate to the outside IP world
NA Tis not configured, and attacks from the outside are not possible at all.
In multisite deployments. IP Security (IPsec) virtual private network (VPN) tunnels can be used
between sites. The VPN tunnels allow only intersite communication; access to the protected
internal networks is not possible from the outside, but only from the other site (through the
tunnel). Therefore, attacks from the outside are blocked at the gateway. To configure IPsec
VPNs, you must configure gateways at each site. Sometimes this configuration is not possible
such as when the two sites are under separate administration, and security policies do not allow
the configuration of IPsec VPNs.
In these cases, or when connecting to apublic service such as an ITSP, you must configure
NAT tor Cisco Unified Communications Manager servers and IP phones. Once Cisco Unified
Communications Manager servers and IP phones are reachable with public IP addresses thev
will be subject to attacks from the outside world, which introduces potential securitv issues '
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©2010 Cisco Systems, Inc.
The ideal solution for alarge deployment would allow an automatic recognition ofroutes.
Internal as well as external (for PSTN backup) numbers should be advertised and learned by
call-routing entities. Adynamic routing protocol for call-routing targets would address
scalabilit} issues in large deployments.
Call control discovcrv (CCD). a feature that is based on the Cisco Service Advertisement
Framework (SAF) provides such functionality. CCD and Cisco SAF are explained in more
detail in a later module of this course.
Multisite Deployment Implementation
C 2010 Cisco Systems. Inc