Implementation & Management of
Cisco Unified Border Element
(CUBE) Enterprise
BRKUCC-2934
Hussain Ali
Technical Marketing Engineer
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BRKUCC-2934
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3
Agenda
• SIP Trunking and CUBE Overview
• SIP Trunking Design & Deployment Models
• CUBE Architecture
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE
• CUBE Management & Troubleshooting
• Futures & Key Takeaways
BRKUCC-2934
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Why does an enterprise need an SBC ?
Enterprise 1
SIP
IP
IP
Enterprise 2
SIP
IP
CUBE
CUBE
Rich Media (Real time Voice, Video, Screenshare etc.. ) Rich Media
BRKUCC-2934
SESSION
CONTROL
SECURITY
INTERWORKING
DEMARCATION
Call Admissions
Control
Trunk Routing
Ensuring QoS
Statistics and Billing
Redundancy/
Scalability
Encryption
Authentication
Registration
SIP Protection
Voice Policy
Firewall Placement
Toll Fraud
SIP - SIP
H.323 - SIP
SIP Normalization
DTMF Interworking
Transcoding
Codec Filtering
Fault Isolation
Topology Hiding
Network Borders
L5/L7 Protocol
Demarcation
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5
Cisco Unified Border Element – Router Integration
An Integrated Network Infrastructure Service
Cisco Unified Border Element
TDM Gateway
 Address Hiding
PSTN Backup
 H.323 and SIP interworking
 DTMF interworking
 SIP security
Voice Policy
 Transcoding
CUBE
Note: An SBC appliance would
have only these features
IP Routing &
MPLS
WAN & LAN
Physical
Interfaces
Unified CM
Conferencing and
Transcoding
FW, IPS,
QoS
SRST
VXML
Note: Some features/components may require additional licensing
BRKUCC-2934
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CUBE (Enterprise) Product Portfolio
ASR 1004/6 RP2
Introduced in July 2012
50-150
ASR 1002-X
ASR 1001-X
50-100
Introduced in July 2013
CPS
3900E Series ISR-G2
(3925E, 3945E)
ASR 1001
Introducing ASR 1001-X
May 2014
Support for ~10,000 sessions
ISR 4451-X
20-35
3900 Series ISR-G2
(3925, 3945)
17
2900 Series
ISR-G2 (2901,
2911, 2921, 2951)
8-12
<5
800/1861 ISR
4
<50
500-600
900-1000
2000-2500
4000
7000-10,000
Active Concurrent Voice Calls Capacity
BRKUCC-2934
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7
12K-14K
14-16K
For Your
Reference
CUBE Session Capacity Summary
Platform
CUBE Sessions
NanoCUBE (8XX and SPIAD Platforms)
15 - 120
2901
100
2911
200
2921
400
2951
600
3925
800
3945
950
3925E
2100
3945E
2500
4451-X (IOS-XE 3.11)
4000
ASR1001-X
10000
ASR1001/1002-X
10000
ASR1004/1006 RP2
16000
BRKUCC-2934
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Introduced in
Oct 2013
Introduced in
July 2013
Introduced in
May 2014
For Your
Reference
CUBE ISR and ASR Licensing
Platform
Cisco 881, 886, 887, 888, 892F, SPIAD
Cisco 2901, 2911, 2921 ISR G2
Cisco 2951, 3925 ISR G2
Cisco 3945, 3925E, 3945E ISR G2
ISR 4451-X
Cisco ASR1000
Single-Use Licenses
FL-NANOCUBE
NEW
FL-CUBEE-5
FL-CUBEE-25
FL-CUBEE-100
FL-CUBEE-5
FL-CUBEE-25
FL-CUBEE-100
FL-CUBEE-500
FL-CUBEE-5
FL-CUBEE-25
FL-CUBEE-100
FL-CUBEE-500
FL-CUBEE-1000
FLASR1-CUBEE-100P
FLASR1-CUBEE-500P
FLASR1-CUBEE-1KP
FLASR1-CUBEE-4KP
FLASR1-CUBEE-16KP
Redundancy Licenses
( 1 SKU for Active/Standby Pair)
N/A
FL-CUBEE-5-RED
FL-CUBEE-25-RED
FL-CUBEE-100-RED
FL-CUBEE-5-RED
FL-CUBEE-25-RED
FL-CUBEE-100-RED
FL-CUBEE-500-RED
FL-CUBEE-5-RED
FL-CUBEE-25-RED
FL-CUBEE-100-RED
FL-CUBEE-500-RED
FL-CUBEE-1000-RED
FLASR1-CUBEE-100R
FLASR1-CUBEE-500R
FLASR1-CUBEE-1K-R
FLASR1-CUBEE-4K-R
FLASR1-CUBEE-16KR
http://www.cisco.com/c/en/us/products/collateral/unified-communications/unified-border-element/order_guide_c07_462222.html
BRKUCC-2934
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9
CUBE Software Release Mapping
ISR G2
ASR
CUBE
Vers.
2900/ 3900
FCS
CUBE Ent
ASR Parity
with ISR
8.7
15.1.4M
Apr 2011
~50%
1.4.2
3.4
15.1(3)S
July 2011
8.8
15.2.1T
July 2011
~70%
1.4.3
3.5
15.2(1)S
Nov 2011
8.9
15.2.2T
Nov 2011
>80%
1.4.4
3.6
15.2(2)S
Mar 2012
Mar 2012
>85%
9.0
3.7
15.2(4)S
July 2012
9.0.1
3.8
15.3(1)S
Oct 2012
9.0.2
3.9
15.3(2)S
Mar 2013
9.0
15.2.3T/
15.2.4M
CUBE
Vers.
IOS XE Release
FCS
9.0.1
15.3.1T
Oct 2012
9.0.2
15.3(2)T
Mar 2013
>95%
>95%
9.5.1
15.3(3)M1
Oct 2013
>95%
9.5.1
3.10.1
15.3(3)S1
Oct 2013
10.0.0
15.4(1)T
Nov 2013
10.0.0
3.11
15.4(1)S
Nov 2013
10.0.1
15.4(2)T
Mar 2014
10.0.1
3.12
15.4(2)S
Mar 2014
10.0.2
15.4(3)M
July 2014
10.0.2
3.13
15.4(3)S
July 2014
10.0.3
15.5(1)T
Nov 2014
>95%
>95%
>95%
>95%
10.0.3
3.14
15.5(1)S
Nov 2014
BRKUCC-2934
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10
Agenda
• SIP Trunking and CUBE Overview
• SIP Trunking Design & Deployment Models
• CUBE Architecture
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE
• CUBE Management & Troubleshooting
• Futures & Key Takeaways
BRKUCC-2934
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11
Cisco Session Management & CUBE:
Essential Elements for Collaboration
• CUBE provides session border control
between IP networks
–
–
–
–
Demarcation
Interworking
Session control
Security
SIP TRUNK TO CUBE
• Cisco SME centralizes
network control
© 2014 Cisco and/or its affiliates. All rights reserved.
Video
3rd Party IP
PBX
Cisco Public
12
Cisco B2B
Cisco Session
Management
IM, Presence,
Voicemail
– Centralizes dial plan
– Centralized applications
– Aggregates PBXs
BRKUCC-2934
CUBE
Mobile
TDM PBX
12
CUBE Deployment Scenarios
TDM
SIP Trunks
for PSTN
Access
SIP
SIP Trunk
H.323
SBC
SP VOIP
Services
CUBE
Standby
Networkbased
Media
Recording
Solution
Partner API
MediaSense
Extending to Video and
High Availability for Audio Calls
CUBE
SIP
SIP
RTP
CUBE
SBC
SP IP
Network
SBC
SP IP
Network
Active
IVR
Integration
for Contact
Centers
CVP
vXML Server
SIP
CUBE
Business to
Business
Telepresence
BRKUCC-2934
Media
Server
SIP
SBC
CUBE
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13
SIP
SP IP
Network
CUBE
The Centralized Model
Characteristics of Centralized
Operational Benefits
• Central Site is the only location with
SIP session connectivity to IP PSTN
• Centralizes Physical
Operations
• Voice services delivered to Branch
Offices over the Enterprise IP WAN
(usually MPLS)
• Centralizes Dial-Peer
Management
• Media traffic hairpins through
central site between SP and
branches
• Centralizes SIP Trunk
Capacity
Challenges
• Increased campus bandwidth, CAC,
latency; media optimization
• HA in campus
• Survivability at branch (PSTN
connection at the branch)
• Emergency services
• Legal/Regulatory
Centralized
IP PSTN
Enterprise
IP WAN
CUBE
Site-SP Media
BRKUCC-2934
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14
The Distributed Model
Characteristics of Distributed
Operational Benefits
Challenges
• Each site has direct connection
for SIP sessions to SP
• Leverages existing branch
routers
• Distributed dial-peer
management
• Takes advantage of SP session
pooling, if offered by SP
• No media hair-pinning thru any
site
• Distributed operational overhead
• Media traffic goes direct from
each branch site to the SP
• Lower latency on voice or video
• IP addressing to Service Provider
from branch
• Built-in Redundancy strategy
Distributed
• Quickest transition from
IP PSTN
existing
TDM
Enterprise
IP WAN
CUBE
CUBE
BRKUCC-2934
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CUBE
15
CUBE
CUBE
Site-SP Media
.. and the Hybrid Model
Characteristics of Hybrid
Benefits
• Connection to SP SIP service is determined on a site by
site basis to be either direct or routed through a regional
site.
• Decision to route call direct or indirect based on various
criteria
• Adaptable to site specific requirements
• Optimizes BW use on Enterprise WAN
• Adaptable to regional SP issues
• Built-in redundancy strategy
• Media traffic goes direct from site to SP or hairpins
through another site, depending on branch configuration.
Hybrid
IP PSTN
Enterprise
IP WAN
CUBE
CUBE
CUBE
BRKUCC-2934
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16
CUBE
CUBE
In-Depth Explanation of SIP Deployment Models
Educate your customer on SIP Deployment Models
New White Paper will be posted by the end of January at the following URL:
www.cisco.com/go/cube
BRKUCC-2934
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17
Agenda
• SIP Trunking and CUBE Overview
• SIP Trunking Design & Deployment Models
• CUBE Architecture
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE
• CUBE Management & Troubleshooting
• Futures & Key Takeaways
BRKUCC-2934
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18
ASR & ISR-G2 Architecture Comparison
ASR (IOS-XE based) Architecture
ISR G2 Architecture
Control Plane
CPU
IOS
RP
Control
Plane
IOS
IOS
I/O
Kernel
ESP
I/O
ISR: Pkt fwd’ing and signaling are handled by the
same CPU

ASR: Pkt fwd’ing and signaling are handled by
different CPUs
‒
ESP must be programmed or instructed by the
control plane to do specific media functions
‒
Performed by Forwarding Plane Interface (FPI)
Data (Forwarding) Plane
BRKUCC-2934
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19
I/O

Msg I/f
I/O
Data Plane
ASR & ISR-G2/4451-X Feature Comparison
General SBC Features
ASR
ISR-G2
4451-X
High Availability Implementation
Redundancy-Group
Infrastructure
HSRP Based
Redundancy-Group
Infrastructure
TDM Trunk Failover/Co-existence
Not Available
Exists
Exists
Media Forking
XE3.8
(Thousands of calls)
15.2.1T
(Upto 1250 calls)
XE3.10
Software MTP registered to CUCM (Including
HA Support)
XE3.6
Exists
Exists
DSP Card
SPA-DSP
PVDM2/PVDM3
PVDM4
Transcoder registered to CUCM
Not Available
Exists via SCCP
Exists via SCCP (XE3.11)
Transcoder Implementation
Local Transcoder
Interface (LTI)
SCCP or LTI (starting IOS
15.2.3T)
SCCP and LTI
Embedded Packet Capture
Exists
Exists
Exists
Web-based UC API
XE3.8
15.2.2T
Exists
Noise Reduction & ASP
Exists
15.2.3T
Exists
Call Progress Analysis
XE3.9
15.3.2T
Exists
CME/SRST and CUBE co-existence
Not Available
Exists
XE3.11
SRTP-RTP Call flows
Exists (NO DSPs needed)
Exists (DSPs required)
Exists (NO DSPs needed)
BRKUCC-2934
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20
Agenda
• SIP Trunking and CUBE Overview
• SIP Trunking Design & Deployment Models
• CUBE Architecture
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE
• CUBE Management & Troubleshooting
• Futures & Key Takeaways
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
21
Transitioning to Centralized SIP Trunking...
Re-purpose your existing Cisco voice gateway’s as Session Border Controllers
BEFORE
AFTER
SIP/H323/MGCP
Media
SIP Trunks
Media
Standby
Enterprise Campus
A
A
High-density Dedicated
Gateways
Enterprise Campus
CUBE
IP PSTN
Active
CUBE
MPLS
MPLS
CUBE with High
Availability
PSTN is now
used only for
emergency
calls over FXO
lines
SRST
CME
SRST
CME
TDM PBX
Enterprise
Branch Offices
BRKUCC-2934
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Enterprise
Branch Offices
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22
TDM PBX
Steps to transitioning...
SIP Trunk
Media
• Step 1 – Configure IP PBX to route
all calls (HQ and branch offices) to
the edge SBC
Standby
A
CUBE
IP PSTN
Active
• Step 2 – Get SIP Trunk details from
the provider
CUBE
Enterprise
Campus
CUBE with High
Availability
• Step 3 – Enable CUBE application
on Cisco routers
MPLS
PSTN is now
used only for
emergency
calls over FXO
lines
SRST
CME
TDM PBX
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• Step 5 – Normalize SIP messages
to meet SIP Trunk provider’s
requirements
• Step 6 – Execute the test plan
Enterprise Branch
Offices
BRKUCC-2934
• Step 4 – Configure call routing on
CUBE (Incoming & Outgoing dialpeers)
23
Also see BRKUCC-2006
Step 1: Configure CUCM to route calls to the edge SBC
SIP Trunk Pointing to CUBE
Standby
A
CUBE
IP PSTN
Active
CUBE
Enterprise
Campus
CUBE with High
Availability
MPLS
• Configure CUCM to route all PSTN
PSTN is now
calls (central and branch) to CUBE via
used only for
a SIP trunk
SRST
emergency
calls over
calls
FXO lines
• Make sure all different patterns of
– local, long distance, international,
emergency,
informational etc.. are
CME
pointing to CUBE
TDM PBX
Enterprise
Branch Offices
BRKUCC-2934
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24
Step 2: Get details from SIP Trunk provider
BRKUCC-2934
Sample
Response
Item
SIP Trunk service provider requirement
1
SIP Trunk IP Address (Destination IP Address for INVITES)
20.1.1.2 or
DNS
2
SIP Trunk Port number (Destination port number for INVITES)
5060
3
SIP Trunk Transport Layer (UDP or TCP)
UDP
4
Codecs supported
G711, G729
5
Fax protocol support
T.38
6
DTMF signaling mechanism
RFC2833
7
Does the provider require SDP information in initial INVITE (Early offer
required)
Yes
8
SBC’s external IP address that is required for the SP to accept/authenticate
calls (Source IP Address for INVITES)
20.1.1.1
9
Does SP require SIP Trunk registration for each DID? If yes, what is the
username & password
No
10
Does SP require Digest Authentication? If yes, what is the username &
password
No
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25
Step 3: Enable CUBE Application on Cisco routers
1. Enable CUBE Application
voice service voip
mode border-element license capacity 200
allow-connections sip to sip
2. Configure any other global settings to meet SP’s requirements
voice service voip
sip
early-offer forced
header-passing
error-passthru
3. Create a trusted list of IP addresses to prevent toll-fraud
BRKUCC-2934
voice service voip
ip address trusted list
ipv4 10.1.1.50
ipv4 20.20.20.20
sip
silent discard-untrusted  Default configuration starting XE 3.10.1 /15.3(3)M1 to
mitigate
TDoS Attack
26
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Cisco Public
Step 4: Configure Call routing on CUBE
Standby
A
CUBE with High
Availability
CUBE
Active
IP PSTN
CUBE
Enterprise
Campus
MPLS
LAN Dial-Peers
WAN Dial-Peers
• Dial-Peer – “static routing” table mapping phone numbers
SRST
PSTN is now
used only for
toemergency
interfaces
callsor
over FXO lines
IP addresses
• LAN Dial-Peers – Dial-peers that are facing towards the IP PBX for sending and receiving
calls to & from the PBX
CME
• WAN Dial-Peers – Dial-peers that are facing towards the SIP Trunk provider for sending & receiving
calls to & from the provider
TDM PBX
Enterprise Branch
Offices
BRKUCC-2934
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27
LAN Dial-Peer Configuration
Inbound Dial-Peer for calls from CUCM to CUBE
dial-peer voice 100 voip
description *** Inbound LAN side dial-peer ***
incoming called-number 9T
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
CUCM sending 9
+ All digits dialed
Outbound Dial-Peer for calls from CUBE to CUCM
dial-peer voice 200 voip
description *** Outbound LAN side dial-peer ***
destination-pattern [2-9].........
session protocol sipv2
session target ipv4:<CUCM_Address>
codec g711ulaw
dtmf-relay rtp-nte
SP will be
sending 10 digits
inbound
Note: If more than 1 CUCM cluster exists, you will have to create multiple such LAN dial-peers with “preference CLI” for CUCM
redundancy/load balancing as the traditional way to accommodate multiple trunks
BRKUCC-2934
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28
WAN Dial-Peer Configuration
Inbound Dial-Peer for calls from SP to CUBE
dial-peer voice 100 voip
description *** Inbound WAN side dial-peer ***
incoming called-number [2-9].........
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
Catch-all for
all inbound
PSTN calls
Outbound Dial-Peer for calls from CUBE to SP
dial-peer voice 200 voip
description *** Outbound WAN side dial-peer ***
translation-profile outgoing Digitstrip
destination-pattern 9[2-9].........
session protocol sipv2
voice-class sip bind control source gig0/1
voice-class sip bind media source gig0/1
session target ipv4:<SIP_Trunk_IP_Address>
codec g711ulaw
dtmf-relay rtp-nte
BRKUCC-2934
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29
Dial-peer for
making long
distance calls
to SP
Note: Separate outgoing DP to be created for Local, International,
Emergency, Informational calls etc.
Step 5: SIP Normalization
SIP profiles is a mechanism to normalize or customize SIP at the network border to
provide interop between incompatible devices
Add user=phone for INVITEs
SIP incompatibilities arise due to:
Incoming
• A device rejecting an unknown header (value
or parameter) instead of ignoring it
INVITE
sip:5551000@sip.com:5060
SIP/2.0
• A device expecting an optional header
value/parameter or can be implemented in
multiple ways
Outgoing
CUBE
INVITE
sip:5551000@sip.com:5060
user=phone SIP/2.0
voice class sip-profiles 100
request INVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"
request REINVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"
• A device sending a value/parameter that must
be changed or suppressed (“normalized”)
before it leaves/enters the enterprise to comply
with policies
Modify a “sip:” URI to a “tel:” URI in INVITEs
Incoming
INVITE
sip:2222000020@9.13.24.6:5060
SIP/2.0
• Variations in the SIP standards of how to
achieve certain functions
Outgoing
CUBE
INVITE
tel:2222000020
SIP/2.0
voice class sip-profiles 100
request INVITE sip-header SIP-Req-URI modify "sip:(.*)@[^ ]+" "tel:\1"
request INVITE sip-header From modify "<sip:(.*)@.*>" "<tel:\1>"
request INVITE sip-header To modify "<sip:(.*)@.*>" "<tel:\1>"
• With CUBE 10.0.1 SIP Profiles
can be applied to inbound SIP
messages as well
More information at www.cisco.com/go/cube > Configure > Configuration Examples and TechNotes
BRKUCC-2934
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30
Normalize Outbound SIP Message (Example 1)
SIP Provider
Requirement
For Call Forward & Transfer scenarios back to PSTN, the Diversion
header should match the registered DID of your network
SIP INVITE that CUBE sends
SIP INVITE that Service Provider expects
Sent:
INVITE sip:2000@9.44.44.4:5060 SIP/2.0
………
User-Agent: Cisco-SIPGateway/IOS-15.2.3.T
………
Diversion: <sip:3000@9.44.44.4>;privacy=off;
reason=unconditional;screen=yes
……...
m=audio 6001 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
……...
Sent:
INVITE sip:2000@9.44.44.4:5060 SIP/2.0
……….
User-Agent: Cisco-SIPGateway/IOS-15.2.3.T
……….
Diversion: <sip:4085266855@sip.abc.com>;
privacy=off;reason=unconditional;screen=yes
……….
m=audio 32278 RTP/AVP 18 8 101
a=rtpmap:0 PCMU/8000
………..
Configure
SIP Profiles
voice class sip-profiles 400
request INVITE sip-header Diversion modify “sip:(.*>)” “sip:4085266855@sip.abc.com>”
request REINVITE sip-header Diversion modify “sip:(.*>)” “sip:4085266855@sip.abc.com>”
Apply to
Dial-peer or
Globally
dial-peer voice 4000 voip
description Incoming/outgoing SP
voice-class sip profiles 400
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
voice service voip
sip
sip profiles 400
31
For Your
Reference
Normalize Inbound SIP Message (Example 2)
CUBE
Requirement
SIP Diversion header must include a user portion
SIP INVITE received by CUBE
SIP INVITE CUBE expects
Sent:
INVITE sip:2000@9.44.44.4:5060 SIP/2.0
………
User-Agent: SP-SBC
………
Diversion: <sip:9.44.44.4>;privacy=off;
reason=unconditional;screen=yes
……...
m=audio 6001 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
……...
Enable Inbound SIP
Profile feature
Sent:
INVITE sip:2000@9.44.44.4:5060 SIP/2.0
……….
User-Agent: SP-SBC
……….
Diversion: <sip:1234@abc.com>;
privacy=off;reason=unconditional;screen=yes
……….
m=audio 32278 RTP/AVP 18 8 101
a=rtpmap:0 PCMU/8000
………..
voice service voip
sip
sip-profiles inbound
Configure Inbound
SIP Profile to add a
dummy user part
voice class sip-profiles 400
request INVITE sip-header Diversion modify “sip:” sip:1234@
Apply to Dial-peer
or Globally
dial-peer voice 4000 voip
description Incoming/outgoing SP
voice-class sip profiles 400 inbound
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
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voice service voip
sip
sip profiles 400 inbound
For Your
Reference
Step 6: Execute the Test Plan
• Inbound and outbound Local, Long distance, International calls for G711 &
G729 codecs (if supported by provider)
• Outbound calls to information and emergency services
• Caller ID and Calling Name Presentation
• Supplementary services like Call Hold, Resume, Call Forward & Transfer
• DTMF Tests
• Fax calls – T.38 and fallback to pass-through (if option available)
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
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33
Agenda
• SIP Trunking and CUBE Overview
• SIP Trunking Design & Deployment Models
• CUBE Architecture
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE
• CUBE Management & Troubleshooting
• Futures & Key Takeaways
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
34
Understanding Dial-Peer matching Techniques:
LAN & WAN Dial-Peers
• LAN Dial-Peers – Dial-peers that are facing towards the IP PBX for sending
and receiving calls to & from the PBX
• WAN Dial-Peers – Dial-peers that are facing towards the SIP Trunk provider for
sending & receiving calls to & from the provider
Inbound LAN Dial-Peer
A
Outbound Calls
SIP Trunk
Outbound WAN Dial-Peer
SP SIP Trunk
IP PSTN
CUBE
Inbound Calls
Inbound WAN Dial-Peer
Outbound LAN Dial-Peer
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
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Understanding Inbound Dial-Peer Matching Techniques
Inbound LAN Dial-Peer
Priority
1
Exact Pattern
match
Match Based on URI of an
incoming INVITE message
A
Host Name/IP
Address
3
Match based on Called
Number
Phone-number of
tel-uri
Match based on Calling
number
4
© 2014 Cisco and/or its affiliates. All rights reserved.
IP PSTN
Cisco Public
Inbound Calls
Inbound WAN Dial-Peer
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
Default Dial-Peer = 0
BRKUCC-2934
SP SIP Trunk
SIP Trunk
CUBE
User portion of URI
2
Outbound Calls
36
Understanding Inbound Dial-Peer Matching Techniques
Inbound LAN Dial-Peer
Priority
A
voice class uri 2001 sip
host ipv4:10.2.1.1
1
A
dial-peer voice 1 voip
incoming uri via 1001
B
dial-peer voice 2 voip
incoming uri request 2001
C
dial-peer voice 3 voip
incoming uri to 2001
D
dial-peer voice 4 voip
incoming uri from 1001
2
dial-peer voice 5 voip
incoming called-number 654321
3
dial-peer voice 6 voip
answer-address 555
4
dial-peer voice 7 voip
destination-pattern 555
BRKUCC-2934
Outbound Calls
voice class uri 1001 sip
host ipv4:10.1.1.1
© 2014 Cisco and/or its affiliates. All rights reserved.
SP SIP Trunk
SIP Trunk
IP PSTN
CUBE
Inbound Calls
Inbound WAN Dial-Peer
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
Cisco Public
37
Understanding Inbound Dial-Peer Matching Techniques
Inbound LAN Dial-Peer
Priority
A
voice class uri 2001 sip
host ipv4:10.2.1.1
1
A
dial-peer voice 1 voip
incoming uri via 1001
B
dial-peer voice 2 voip
incoming uri request 2001
C
dial-peer voice 3 voip
incoming uri to 2001
D
dial-peer voice 4 voip
incoming uri from 1001
2
dial-peer voice 5 voip
incoming called-number 654321
3
dial-peer voice 6 voip
answer-address 555
4
dial-peer voice 7 voip
destination-pattern 555
BRKUCC-2934
Outbound Calls
voice class uri 1001 sip
host ipv4:10.1.1.1
© 2014 Cisco and/or its affiliates. All rights reserved.
SP SIP Trunk
SIP Trunk
IP PSTN
CUBE
Inbound Calls
Inbound WAN Dial-Peer
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
Cisco Public
38
Understanding Inbound Dial-Peer Matching Techniques
Inbound LAN Dial-Peer
Priority
A
voice class uri 2001 sip
host ipv4:10.2.1.1
1
A
dial-peer voice 1 voip
incoming uri via 1001
B
dial-peer voice 2 voip
incoming uri request 2001
C
dial-peer voice 3 voip
incoming uri to 2001
D
dial-peer voice 4 voip
incoming uri from 1001
2
dial-peer voice 5 voip
incoming called-number 654321
3
dial-peer voice 6 voip
answer-address 555
4
dial-peer voice 7 voip
destination-pattern 555
BRKUCC-2934
Outbound Calls
voice class uri 1001 sip
host ipv4:10.1.1.1
© 2014 Cisco and/or its affiliates. All rights reserved.
SP SIP Trunk
SIP Trunk
IP PSTN
CUBE
Inbound Calls
Inbound WAN Dial-Peer
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
Cisco Public
39
Understanding Inbound Dial-Peer Matching Techniques
Inbound LAN Dial-Peer
Priority
A
voice class uri 2001 sip
host ipv4:10.2.1.1
1
A
dial-peer voice 1 voip
incoming uri via 1001
B
dial-peer voice 2 voip
incoming uri request 2001
C
dial-peer voice 3 voip
incoming uri to 2001
D
dial-peer voice 4 voip
incoming uri from 1001
2
dial-peer voice 5 voip
incoming called-number 654321
3
dial-peer voice 6 voip
answer-address 555
4
dial-peer voice 7 voip
destination-pattern 555
BRKUCC-2934
Outbound Calls
voice class uri 1001 sip
host ipv4:10.1.1.1
© 2014 Cisco and/or its affiliates. All rights reserved.
SP SIP Trunk
SIP Trunk
IP PSTN
CUBE
Inbound Calls
Inbound WAN Dial-Peer
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
Cisco Public
40
Understanding Inbound Dial-Peer Matching Techniques
Inbound LAN Dial-Peer
Priority
A
voice class uri 2001 sip
host ipv4:10.2.1.1
1
A
dial-peer voice 1 voip
incoming uri via 1001
B
dial-peer voice 2 voip
incoming uri request 2001
C
dial-peer voice 3 voip
incoming uri to 2001
D
dial-peer voice 4 voip
incoming uri from 1001
2
dial-peer voice 5 voip
incoming called-number 654321
3
dial-peer voice 6 voip
answer-address 555
4
dial-peer voice 7 voip
destination-pattern 555
BRKUCC-2934
Outbound Calls
voice class uri 1001 sip
host ipv4:10.1.1.1
© 2014 Cisco and/or its affiliates. All rights reserved.
SP SIP Trunk
SIP Trunk
IP PSTN
CUBE
Inbound Calls
Inbound WAN Dial-Peer
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
Cisco Public
41
Understanding Inbound Dial-Peer Matching Techniques
Inbound LAN Dial-Peer
Priority
A
voice class uri 2001 sip
host ipv4:10.2.1.1
1
A
dial-peer voice 1 voip
incoming uri via 1001
B
dial-peer voice 2 voip
incoming uri request 2001
C
dial-peer voice 3 voip
incoming uri to 2001
D
dial-peer voice 4 voip
incoming uri from 1001
2
dial-peer voice 5 voip
incoming called-number 654321
3
dial-peer voice 6 voip
answer-address 555
4
dial-peer voice 7 voip
destination-pattern 555
BRKUCC-2934
Outbound Calls
voice class uri 1001 sip
host ipv4:10.1.1.1
© 2014 Cisco and/or its affiliates. All rights reserved.
SP SIP Trunk
SIP Trunk
IP PSTN
CUBE
Inbound Calls
Inbound WAN Dial-Peer
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
Cisco Public
42
Understanding Inbound Dial-Peer Matching Techniques
Inbound LAN Dial-Peer
Priority
A
voice class uri 2001 sip
host ipv4:10.2.1.1
1
A
dial-peer voice 1 voip
incoming uri via 1001
B
dial-peer voice 2 voip
incoming uri request 2001
C
dial-peer voice 3 voip
incoming uri to 2001
D
dial-peer voice 4 voip
incoming uri from 1001
2
dial-peer voice 5 voip
incoming called-number 654321
3
dial-peer voice 6 voip
answer-address 555
4
dial-peer voice 7 voip
destination-pattern 555
BRKUCC-2934
Outbound Calls
voice class uri 1001 sip
host ipv4:10.1.1.1
© 2014 Cisco and/or its affiliates. All rights reserved.
SP SIP Trunk
SIP Trunk
IP PSTN
CUBE
Inbound Calls
Inbound WAN Dial-Peer
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
Cisco Public
43
Understanding Outbound Dial-Peer Matching Techniques
Outbound WAN Dial-Peer
Priority
1
Outbound Calls
Match Based on URI of
incoming INVITE message
& carrier-id target
Exact Pattern
match
A
Host Name/IP
Address
2
3
Phone-number of
tel-uri
Host Name/IP
Address
User portion of URI
4
BRKUCC-2934
Match based on Called
number
© 2014 Cisco and/or its affiliates. All rights reserved.
Phone-number of
tel-uri
Cisco Public
Inbound Calls
Outbound LAN Dial-Peer
Exact Pattern
match
Match based on URI of an
incoming INVITE message
IP PSTN
CUBE
User portion of URI
Match based on Called
Number & carrier-id target
SP SIP Trunk
SIP Trunk
44
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
Understanding Outbound Dial-Peer Matching Techniques
Priority
1
Outbound WAN Dial-Peer
Outbound Calls
voice class uri 2001 sip
host ipv4:10.2.1.1
A
SP SIP Trunk
SIP Trunk
dial-peer voice 1 voip
destination uri 2001
carrier-id target orange
IP PSTN
CUBE
Inbound Calls
Outbound LAN Dial-Peer
2
dial-peer voice 2 voip
destination-pattern 654321
carrier-id target orange
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
voice class uri 2001 sip
host ipv4:10.2.1.1
3
4
BRKUCC-2934
dial-peer voice 3 voip
destination uri 2001
dial-peer voice 4 voip
destination-pattern 654321
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
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45
Understanding Outbound Dial-Peer Matching Techniques
Priority
1
Outbound WAN Dial-Peer
Outbound Calls
voice class uri 2001 sip
host ipv4:10.2.1.1
A
SP SIP Trunk
SIP Trunk
dial-peer voice 1 voip
destination uri 2001
carrier-id target orange
IP PSTN
CUBE
Inbound Calls
Outbound LAN Dial-Peer
2
dial-peer voice 2 voip
destination-pattern 654321
carrier-id target orange
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
voice class uri 2001 sip
host ipv4:10.2.1.1
3
4
BRKUCC-2934
dial-peer voice 3 voip
destination uri 2001
dial-peer voice 4 voip
destination-pattern 654321
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
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46
Understanding Outbound Dial-Peer Matching Techniques
Priority
1
Outbound WAN Dial-Peer
Outbound Calls
voice class uri 2001 sip
host ipv4:10.2.1.1
A
SP SIP Trunk
SIP Trunk
dial-peer voice 1 voip
destination uri 2001
carrier-id target orange
IP PSTN
CUBE
Inbound Calls
Outbound LAN Dial-Peer
2
dial-peer voice 2 voip
destination-pattern 654321
carrier-id target orange
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
voice class uri 2001 sip
host ipv4:10.2.1.1
3
4
BRKUCC-2934
dial-peer voice 3 voip
destination uri 2001
dial-peer voice 4 voip
destination-pattern 654321
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
47
47
Understanding Outbound Dial-Peer Matching Techniques
Priority
1
Outbound WAN Dial-Peer
Outbound Calls
voice class uri 2001 sip
host ipv4:10.2.1.1
A
SP SIP Trunk
SIP Trunk
dial-peer voice 1 voip
destination uri 2001
carrier-id target orange
IP PSTN
CUBE
Inbound Calls
Outbound LAN Dial-Peer
2
dial-peer voice 2 voip
destination-pattern 654321
carrier-id target orange
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
voice class uri 2001 sip
host ipv4:10.2.1.1
3
4
BRKUCC-2934
dial-peer voice 3 voip
destination uri 2001
dial-peer voice 4 voip
destination-pattern 654321
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
48
Understanding Outbound Dial-Peer Matching Techniques
Outbound WAN Dial-Peer
Priority
1
Outbound Calls
Match Based on URI of
incoming INVITE message
& carrier-id target
Exact Pattern
match
A
Host Name/IP
Address
2
Phone-number of
tel-uri
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Match based on URI of an
incoming INVITE message
Host Name/IP
Address
User portion of URI
4
BRKUCC-2934
Inbound Calls
Outbound LAN Dial-Peer
Exact Pattern
match
3
IP PSTN
CUBE
User portion of URI
Match based on Called
Number & carrier-id target
SP SIP Trunk
SIP Trunk
Match based on Called
number
© 2014 Cisco and/or its affiliates. All rights reserved.
Phone-number of
tel-uri
Cisco Public
49
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
Additional Headers for Outbound Dial-Peer Matching
Outbound WAN Dial-Peer
Match Based on URI of incoming INVITE message with
or without carrier-id target
Outbound Calls
A
Match based on CALLED
carrier-id target
Number with or without
Match Based on TO Header of incoming INVITE
Match Based on VIA Header of incoming INVITE
Match based on DIVERSION Header of incoming
INVITE
Match based on REFERRED-BY Header of incoming
INVITE
BRKUCC-2934
IP PSTN
CUBE
Inbound Calls
Match Based on FROM Header of incoming INVITE
Match based on CALLING
SP SIP Trunk
SIP Trunk
Number
© 2014 Cisco and/or its affiliates. All rights reserved.
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50
Outbound LAN Dial-Peer
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";;branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
Supported: timer
Max-Forwards: 70
Subject: BRKUCC-2934 Session
Content-Type: application/sdp
Content-Length: 226
........
Introducing Outbound Dial-peer Provision Policy
• Flexibility to choose how outbound dial-peers are selected
• Dynamically set the priority based on Inbound dial-peers
• Additional Inbound Leg Headers for Outbound Dial-peer Matching
VIA
FROM
TO
DIVERSION
REFERRED-BY
Calling Number
• User-defined outbound dial-peer provision policy on a per incoming call bases
1. A provision policy contains two rules to save the match attributes and its precedence
2. Up to two match attributes can be defined from each rule of a provision policy
3. A provision policy setup will be used to match outbound dial-peers once it is
associated to an incoming VoIP call.
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
51
Dial-peer Provision Policy Configuration
For Your
Reference
1. Define Voice Class Dial-peer Provision Policy
CUBE(config)#voice class dial-peer provision-policy <tag>
CUBE(config-class)# description “Match outbound dial-peer based on this Criteria”
CUBE(config-class)#preference ?
<1-2> Preference order
CUBE(config-class)#preference 1 first-attribute second-attribute
called
Match called number
calling
Match calling number
carrier-id
Match carrier id
diversion Match diversion uri
from
Match from uri
to
Match to uri
uri
Match destination uri
via
Match via uri
referred-by
Match referred-by uri
voice class dial-peer provision-policy <tag>
description “Match outbound dial-peer based on criteria defined here”
preference 1 first-attribute second-attribute
preference 2 first-attribute second-attribute
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
52
Dial-peer Provision Policy Configuration – Cont’d
2. Associate Voice Class Provision Policy to an Incoming Dial-peer
dial-peer voice 1 voip
description Inbound Dial-peer
destination provision-policy <tag>
3. Define Outbound Dial-peer with match patterns based on attributes in a policy
CUBE(config)#dial-peer voice 2 voip
CUBE(config-dial-peer)#description Outbound Dial-peer
CUBE(config-dial-peer)#destination ?
calling
Match destination calling number
e164-pattern-map
Configure voice class to match destination e164-pattern-map
uri
Configure voice class to match destination URI
uri-diversion
voice class uri to match sip diversion header
uri-from
voice class uri to match sip from header
uri-referred-by
voice class uri to match sip referred-by header
uri-to
voice class uri to match sip to header
uri-via
voice class uri to match sip via header
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
53
For Your
Reference
Dial-peer Provision Policy Configuration
– Cont’d
For Your
Reference
Configuring a match command for an outbound dial-peer according to the provision policy rule
attribute configured
Provision Policy Rule Attribute
Outbound Dial-peer Match command
called
destination-pattern pattern
destination e164-pattern-map pattern-map-class-id
calling
destination calling e164-pattern-map pattern-map-class-id
carrier-id
carrier-id target
uri
destination uri uri-class-tag
via
destination uri-via uri-class-tag
to
destination uri-to uri-class-tag
from
destination uri-from uri-class-tag
diversion
destination uri-diversion uri-class-tag
referred-by
destination uri-referred-by uri-class-tag
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
54
Dial-peer Provision Policy Example – Match on FROM
voice class uri 10 sip
user-id 555
dial-peer voice 20201 voip
description "Outbound dialpeer based on FROM"
destination uri-from 10
voice class uri 20 sip
host 10.2.1.1
dial-peer voice 1000 voip
description "Inbound dialpeer. Choose outbound based on DPP 10"
destination provision-policy 10
dial-peer voice 2000 voip
description "Inbound dialpeer. Choose outbound based on DPP 20"
destination provision-policy 20
voice class dial-peer provision-policy 10
description "Match outbound dialpeer on both From AND To Headers"
preference 1 from to
!
voice class dial-peer provision-policy 20
description "Match outbound DP based on FROM first, if no match
select based on TO"
preference 1 from
preference 2 to
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
55
dial-peer voice 20202 voip
description "Outbound dialpeer based on TO"
destination uri-to 20
dial-peer voice 10000 voip
description "Outbound dialpeer based on FROM and TO"
destination uri-from 10
destination uri-to 20
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
........
Dial-peer Provision Policy Example – Match on FROM
voice class uri 10 sip
user-id 555
dial-peer voice 20201 voip
description "Outbound dialpeer based on FROM"
destination uri-from 10
voice class uri 20 sip
host 10.2.1.1
dial-peer voice 1000 voip
description "Inbound dialpeer. Choose outbound based on DPP 10"
destination provision-policy 10
dial-peer voice 2000 voip
description "Inbound dialpeer. Choose outbound based on DPP 20"
destination provision-policy 20
dial-peer voice 20202 voip
description "Outbound dialpeer based on TO"
destination uri-to 20
dial-peer voice 10000 voip
description "Outbound dialpeer based on FROM and TO"
destination uri-from 10
destination uri-to 20
voice class dial-peer provision-policy 10
description "Match outbound dialpeer on both From AND To Headers"
preference 1 from to
Received:
voice class dial-peer provision-policy 20
description "Match outbound DP based on FROM first, if no match
select based on TO"
preference 1 from
preference 2 to
From: "555" <sip:555@10.1.1.1:5060>;tag=1
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
56
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";branch=z9hG4bK-23955-1-0
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
........
Dial-peer Provision Policy Example – Match on FROM
dial-peer voice 20201 voip
description "Outbound dialpeer based on FROM"
destination uri-from 10
voice class uri 10 sip
user-id 555
voice class uri 20 sip
host 10.2.1.1
dial-peer voice 1000 voip
description "Inbound dialpeer. Choose outbound based on DPP 10"
destination provision-policy 10
dial-peer voice 2000 voip
description "Inbound dialpeer. Choose outbound based on DPP 20"
destination provision-policy 20
dial-peer voice 20202 voip
description "Outbound dialpeer based on TO"
destination uri-to 20
dial-peer voice 10000 voip
description "Outbound dialpeer based on FROM and TO"
destination uri-from 10
destination uri-to 20
voice class dial-peer provision-policy 10
description "Match outbound dialpeer on both From AND To Headers"
preference 1 from to
Received:
voice class dial-peer provision-policy 20
description "Match outbound DP based on FROM first, if no match
select based on TO"
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
........
preference 1 from
preference 2 to
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";branch=z9hG4bK-23955-1-0
57
Dial-peer Provision Policy Example – Match on TO
voice class uri 10 sip
user-id 555
dial-peer voice 20201 voip
description "Outbound dialpeer based on FROM"
destination uri-from 10
voice class uri 20 sip
host 10.2.1.1
shutdown
dial-peer voice 1000 voip
description "Inbound dialpeer. Choose outbound based on DPP 10"
destination provision-policy 10
dial-peer voice 2000 voip
description "Inbound dialpeer. Choose outbound based on DPP 20"
destination provision-policy 20
voice class dial-peer provision-policy 10
description "Match outbound dialpeer on both From AND To Headers"
preference 1 from to
voice class dial-peer provision-policy 20
description "Match outbound DP based on FROM first, if no match
select based on TO"
preference 1 from
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
dial-peer voice 10000 voip
description "Outbound dialpeer based on FROM and TO"
destination uri-from 10
destination uri-to 20
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
........
preference 2 to
BRKUCC-2934
dial-peer voice 20202 voip
description "Outbound dialpeer based on TO"
destination uri-to 20
58
Dial-peer Provision Policy Example – Match on TO
voice class uri 10 sip
user-id 555
dial-peer voice 20201 voip
description "Outbound dialpeer based on FROM"
destination uri-from 10
voice class uri 20 sip
host 10.2.1.1
shutdown
dial-peer voice 1000 voip
description "Inbound dialpeer. Choose outbound based on DPP 10"
destination provision-policy 10
dial-peer voice 2000 voip
description "Inbound dialpeer. Choose outbound based on DPP 20"
destination provision-policy 20
voice class dial-peer provision-policy 10
description "Match outbound dialpeer on both From AND To Headers"
preference 1 from to
voice class dial-peer provision-policy 20
description "Match outbound DP based on FROM first, if no match
select based on TO"
preference 1 from
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
dial-peer voice 10000 voip
description "Outbound dialpeer based on FROM and TO"
destination uri-from 10
destination uri-to 20
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
........
preference 2 to
BRKUCC-2934
dial-peer voice 20202 voip
description "Outbound dialpeer based on TO"
destination uri-to 20
59
Dial-peer Provision Policy Example – Match on FROM & TO
voice class uri 10 sip
user-id 555
dial-peer voice 20201 voip
description "Outbound dialpeer based on FROM"
destination uri-from 10
voice class uri 20 sip
host 10.2.1.1
dial-peer voice 1000 voip
description "Inbound dialpeer. Choose outbound based on DPP 10"
destination provision-policy 10
dial-peer voice 2000 voip
description "Inbound dialpeer. Choose outbound based on DPP 20"
destination provision-policy 20
voice class dial-peer provision-policy 10
description "Match outbound dialpeer on both From AND To Headers"
preference 1 from to
voice class dial-peer provision-policy 20
description "Match outbound DP based on FROM first, if no match
select based on TO"
preference 1 from
preference 2 to
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
60
dial-peer voice 20202 voip
description "Outbound dialpeer based on TO"
destination uri-to 20
dial-peer voice 10000 voip
description "Outbound dialpeer based on FROM and TO"
destination uri-from 10
destination uri-to 20
Received:
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";branch=z9hG4bK-23955-1-0
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
........
Dial-peer Provision Policy Example – Match on FROM & TO
voice class uri 10 sip
user-id 555
dial-peer voice 20201 voip
description "Outbound dialpeer based on FROM"
destination uri-from 10
voice class uri 20 sip
host 10.2.1.1
dial-peer voice 1000 voip
description "Inbound dialpeer. Choose outbound based on DPP 10"
destination provision-policy 10
dial-peer voice 2000 voip
description "Inbound dialpeer. Choose outbound based on DPP 20"
destination provision-policy 20
dial-peer voice 20202 voip
description "Outbound dialpeer based on TO"
destination uri-to 20
dial-peer voice 10000 voip
description "Outbound dialpeer based on FROM and TO"
destination uri-from 10
destination uri-to 20
voice class dial-peer provision-policy 10
description "Match outbound dialpeer on both From AND To Headers"
preference 1 from to
Received:
voice class dial-peer provision-policy 20
description "Match outbound DP based on FROM first, if no match
select based on TO"
preference 1 from
preference 2 to
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";branch=z9hG4bK-23955-1-0
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
........
61
Dial-peer Provision Policy Example – Match on FROM & TO
dial-peer voice 20201 voip
description "Outbound dialpeer based on FROM"
destination uri-from 10
shutdown
voice class uri 10 sip
user-id 555
voice class uri 20 sip
host 10.2.1.1
dial-peer voice 1000 voip
description "Inbound dialpeer. Choose outbound based on DPP 10"
destination provision-policy 10
dial-peer voice 2000 voip
description "Inbound dialpeer. Choose outbound based on DPP 20"
destination provision-policy 20
dial-peer voice 20202 voip
description "Outbound dialpeer based on TO"
destination uri-to 20
dial-peer voice 10000 voip
description "Outbound dialpeer based on FROM and TO"
destination uri-from 10
destination uri-to 20
voice class dial-peer provision-policy 10
description "Match outbound dialpeer on both From AND To Headers"
preference 1 from to
Received:
voice class dial-peer provision-policy 20
description "Match outbound DP based on FROM first, if no match
select based on TO"
preference 1 from
preference 2 to
From: "555" <sip:555@10.1.1.1:5060>;tag=1
To: ABC <sip:654321@10.2.1.1:5060>
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
INVITE sip:654321@10.2.1.1 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;x-routetag="cid:orange@10.1.1.1";branch=z9hG4bK-23955-1-0
Call-ID: 1-23955@10.1.1.1
CSeq: 1 INVITE
Contact: sip:555@10.1.1.1:5060
........
62
Destination Server Group
• Supports multiple destinations (session targets) be defined in a group and applied to a
single outbound dial-peer
• Once an outbound dial-peer is selected to route an outgoing call, multiple destinations
within a server group will be sorted in either round robin or preference [default] order
• This reduces the need to configure multiple dial-peers with the same capabilities but
different destinations. E.g. Multiple subscribers in a cluster
voice class server-group 1
hunt-scheme {preference | round-robin}
ipv4 1.1.1.1 preference 5
ipv4 2.2.2.2
ipv4 3.3.3.3 port 3333 preference 3
ipv6 2010:AB8:0:2::1 port 2323 preference 3
ipv6 2010:AB8:0:2::2 port 2222
dial-peer voice 100 voip
description Outbound DP
destination-pattern 1234
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
session server-group 1
* DNS target not supported in server group
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
63
Multiple Destination-Patterns Under Same
Outbound Dial-Peer
Site A
(919)200-2000
Site B
(510)100-1000
Site C
(408)100-1000
G729 Sites
voice class e164-pattern-map 100
e164 919200200.
e164 510100100.
e164 408100100.
dial-peer voice 1 voip
destination e164-pattern-map 100
codec g729r8
session target ipv4:10.1.1.1
A
SIP Trunk
Provides the ability to combine multiple
destination-patterns targeted to the
same destination to be grouped into a
single dial-peer
SP SIP Trunk
IP PSTN
CUBE
Site A
(919)200-2010
Site B
(510)100-1010
Site C
(408)100-1010
voice class e164-pattern-map 100
url flash:e164-pattern-map.cfg
dial-peer voice 1 voip
destination e164-pattern-map 100
codec g711ulaw
session target ipv4:10.1.1.1
G711 Sites
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
64
! This is an example of the contents
of E164 patterns text file
stored in flash:e164-patternmap.cfg
9192002010
5101001010
4081001010
Multiple Incoming Patterns Under Same
Incoming Dial-peer
Site A
(919)200-2000
Site B
(510)100-1000
Site C
(408)100-1000
G729 Sites
voice class e164-pattern-map 100
e164 919200200.
e164 510100100.
e164 408100100.
dial-peer voice 1 voip
description Inbound DP via Calling
incoming calling e164-pattern-map 100
codec g729r8
A
SIP Trunk
Provides the ability to combine multiple
incoming called OR calling numbers on
a single inbound voip dial-peer, reducing
the total number of inbound voip dialpeers required with the same routing
capability
SP SIP Trunk
IP PSTN
CUBE
Site A
(919)200-2010
Site B
(510)100-1010
Site C
(408)100-1010
voice class e164-pattern-map 200
url flash:e164-pattern-map.cfg
dial-peer voice 2 voip
description Inbound DP via Called
incoming called e164-pattern-map 200
codec g711ulaw
G711 Sites
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
65
! This is an example of the contents
of E164 patterns text file
stored in flash:e164-patternmap.cfg
9192002010
5101001010
4081001010
URI Based Dialing Overview
INVITE sip:user@xyz.com
INVITE sip:user@xyz.com
SBC
CUBE
Enterprise
xyz.com
Enterprise
abc.com
Existing CUBE behavior:
• In CUBE URI based routing (user@host), the “user” part must be present and must be an
E164 number
• The outgoing SIP ‘Request-URI’ and ‘To header URI’ are always set to the session target
information of the outbound dial-peer
• For Req-URIs with same user name e.g. hussain@cisco.com, hussain@google.com, two
different dial-peers are configured with the respective session targets
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
66
URI Based Dialing Enhancement –
URI Pass Through
INVITE sip:1234@cisco.com
For Your
Reference
CUBE
INVITE sip:1234@cisco.com
dial-peer voice 100 voip
incoming uri request 1
dial-peer voice 200 voip
session protocol sipv2
destination uri 1
voice-class sip call-route url
session protocol sipv2
session target ipv4:10.1.1.1
voice-class sip requri-passing
voice class uri 1 sip
host cisco.com
• By default, the host portion is replaced with the session target value of the matched
outbound dial-peer
• Enhancement : Outgoing INVITE has same request URI as received in Incoming INVITE.
This can be achieved by configuring ‘requri-passing’ in the outgoing dial-peer or globally.
• Allows for peer-to-peer calling between enterprises using URIs
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
67
URI Based Dialing Enhancement –
‘User’ portion non-E164 format
INVITE sip:hussain@cisco.com
For Your
Reference
CUBE
INVITE sip:hussain@10.1.1.1
dial-peer voice 100 voip
incoming uri request 1
dial-peer voice 200 voip
session protocol sipv2
destination uri 1
voice-class sip call-route url
session protocol sipv2
session target ipv4:10.1.1.1
voice class uri 1 sip
host cisco.com
• By default, alphanumeric/non-E164 users were not allowed
• Enhancement : User part in Incoming INVITE Req-URI can be of Non-E164 format. e.g.
sip:hussain@cisco.com. Outgoing INVITE will have user portion as it is received i.e.
‘hussain’ (unless SIP profiles are applied).
• Useful for video calls
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
68
URI Based Dialing Enhancement –
‘User’ portion absent
INVITE sip:cisco.com
For Your
Reference
CUBE
INVITE sip:cisco.com
dial-peer voice 100 voip
incoming uri request 1
dial-peer voice 200 voip
session protocol sipv2
destination uri 1
voice-class sip call-route url
session protocol sipv2
session target ipv4:10.1.1.1
voice-class sip requri-passing
voice class uri 1 sip
• By default, call is rejected with “400 Bad Request”
host cisco.com
• Enhancement : Incoming INVITE with no user portion (e.g. sip:cisco.com.) is supported. Dial-peer
matching will happen based on ‘host’ portion. Outgoing INVITE Req-URI will not have any user portion in
this case (unless sip-profiles are applied).
• If user portion is present in incoming INVITE ‘To header’, it is retained in outgoing INVITE ‘To Header’
• If ‘voice-class sip requri-passing’ is not configured, INVITE will go out as sip:10.1.1.1
• REFER and 302, both consume and pass-through cases supported as well
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
69
URI Based Dialing Enhancement –
Deriving Target host from Incoming INVITE Req-URI
INVITE sip:hussain@cisco.com
For Your
Reference
CUBE
INVITE sip:hussain@10.1.1.1 Skype
dial-peer voice 100 voip
incoming uri request 1
dial-peer voice 200 voip
session protocol sipv2
destination uri 1
voice-class sip call-route url
session protocol sipv2
Facebook Video
session target sip-uri
voice class uri 1 sip
user hussain
user .*
• For different hosts with the same ‘user’, multiple outgoing dial-peers had to be configured
• Enhancement : To support URIs with the same user portion but with different domains, only one
dial-peer per can be configured. Outgoing dial-peer needs to be configured with ‘session target
sip-uri’ instead of regular session target configuration. This will trigger DNS resolution of the
domain of incoming INVITE Req-URI and dynamically determine the session target IP.
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
70
Destination Dial-peer Group
• Allows grouping of outbound dial-peers based on an incoming dial-peer, reducing
existing outbound dial-peer provisioning requirements
• Eliminates the need to configure extra outbound dial-peers that are sometimes
needed as workarounds to achieve desired call routing outcome
• Multiple outbound dial-peers are saved under a new “voice class dpg <tag>”. The
new “destination dpg <tag>” command line of an inbound voip dial-peer can be
used to reference the new dpg (dial-peer group)
• Once an incoming voip call is handled by an inbound voip dial-peer with an
active dpg, dial-peers of a dpg will then be used as outbound dial-peers for an
incoming call
• The order of outgoing call setups will be the sorted list of dial-peers from a dpg
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
71
Destination Dial-peer Group Configuration
dial-peer voice 1001 voip
description DPG 10000
destination-pattern 1341
session protocol sipv2
session target ipv4:10.1.1.1
!
dial-peer voice 1002 voip
description DPG 10000
destination-pattern 1341
session protocol sipv2
session target ipv4:10.1.1.2
!
dial-peer voice 1003 voip
description DPG 10000
destination-pattern 1341
session protocol sipv2
2. Now session
the DPGtarget
associated
ipv4:10.1.1.3
voice class dpg 10000
description Voice Class DPG for DP Source SJ
dial-peer 1001 preference 1
dial-peer 1002 preference 2
dial-peer 1003
!
dial-peer voice 100 voip
description DP Source SJ w/voice class dpg
incoming called-number 1341
destination dpg 10000
1. Incoming Dial-peer
is first matched
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
with the INBOUND DP is
selected
Cisco Public
72
Audio Transcoding and Transrating
iLBC, iSAC,
Speex
Enterprise
VoIP
SP VoIP
IP Phones:
G.711, G.729 20 ms,
G.722
CUBE
G.729 30 ms
• Transcoding (12.4.20T)
– One voice codec to any other codec E.g. iLBC-G.711 or iLBC-G.729
– Support for H.323 and SIP
– CUCM 7.1.5 or later supports universal Transcoding
• Transrating (15.0.1M)
– Different packetizations of the same codec
–
E.g. G.729 20ms to G.729 30ms
– Support for SIP-SIP calls
– No sRTP support with transrating
dial-peer voice 2 voip
codec g729r8 bytes 30 fixed-bytes
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
!Call volume (gain/loss) adjustment
dial-peer voice 2 voip
audio incoming level-adjustment x
audio outgoing level-adjustment y
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73
• Transcoding: G.711, G.723.1, G.726, G.728,
G.729/a, iLBC, G.722
• Transrating: G.729 20ms ↔ 30ms (AT&T)
Supported Codecs
Packetization
(ms)
G.711 a-law 64 Kbps
10, 20, 30
G.711 µlaw 64 Kbps
10, 20, 30
G.723 5.3/6.3 Kbps
30, 60
G.729, G.729A, G.729B, 10, 20, 30, 40, 50,
G.729AB 8 Kbps
60
G.722—64 Kbps
10, 20, 30
Configuration for SCCP based Transcoding
(ISR-G2/4451-X)
1. Enabling dspfarm services
under voice-card
3. sccp configuration
voice-card 1
dspfarm
dsp services dspfarm
sccp local GigabitEthernet0/0
sccp ccm <CUBE_internal_IP>
identifier 1 version 4.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register CUBE-XCODE
2. telephony-service configuration
telephony-service
sdspfarm units 1
sdspfarm transcode sessions 128
sdspfarm tag 1 CUBE-XCODE
max-ephones 10
max-dn 10
ip source-address
<CUBE_internal_IP> port 2000
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
4. dspfarm profile configuration
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729r8
maximum sessions 10
associate application SCCP
74
For Your
Reference
Configuration for LTI based Transcoding
(ISR-G2/4451-X & ASR)
1. Enabling dspfarm services
under voice-card
Feature Notes:
voice-card 0/1
dspfarm
dsp services dspfarm
• This uses Local Transcoding Interface to
communicate between CUBE and DSPs
• Also available on ISR-G2 starting IOS 15.2.3T
• Can only be used if CUBE invokes the DSP
for media services
• CUCM cannot invoke DSPs using this LTI
interface
2. dspfarm profile configuration
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729abr8
codec g729ar8
codec ilbc
maximum sessions 100
associate application CUBE
BRKUCC-2934
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Mid-call Codec Renegotiation
Transcoder Insert/Drop
G.711
3
CVP
2
Transcoder Inserted
G.711
Call Xfer (signaling only)
Provider supports only
G.729 codec
1
G.729 /
G.711
SP SIP
SIP
CUBE
4
G.729
BRKUCC-2934
G.729
Transcoder Dropped
1
Call arrives on G.729 SIP trunk
2
CVP connects call to speech recognition server that
requires G.711. Since provider does not support G711
CUBE inserts transcoder
3
CVP xfers call to a remote agent that uses G.729
4
CUBE drops xcoder and e2e call becomes G.729 again
© 2014 Cisco and/or its affiliates. All rights reserved.
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Media Forking – Network Based Recording Solution
Dial-peer based
• CUBE sets up a stateful SIP session
with MediaSense server
Cisco Search/Play demo app
-orPartner Application
• After SIP dialog established, CUBE
forks the RTP and sends it for
MediaSense to record
Cisco MediaSense
(authentication disabled w/o UCM)
MediaSens
e
• With XE 3.10.1, Video calls
supported and CUBE HA for audio
calls
SIP
RTP
A
SIP
SIP
SP SIP
RTP
• Call agent
independent
• Configured on a per
Dial-peer level
BRKUCC-2934
CUBE
RTP
media class 1
recorder parameter
media-recording 20
dial-peer voice 20 voip
description dial-peer pointing to MediaSense
Needs to
match
dial-peer voice 1 voip
description dial-peer that needs to be forked
session protocol sipv2
media-class 1
© 2014 Cisco and/or its affiliates. All rights reserved.
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77
session protocol sipv2
session target ipv4:<Mediasense_IP>
Also see BRKUCC-2250
CUCM 10.X Recording
UC Services API
3.
• Selective Recording
• Mobile/SNR/MVA Calls
• Recording Call Preservation
1. Enable HTTP on IOS
ip http server
http client persistent
Gateway/CUBE Recording
Enabled
2. Enable the API on IOS
4.
1.
2.
uc wsapi
source-address [IP_Address_of_CUBE]
3. Enable XMF service within the API
5.
provider xmf
remote-url 1 http://CUCM:8090/ucm_xmf
no shutdown
[1] – [3]: An external call is answered by user with IP phone
[4] – [5]: CUCM sends forking request over HTTP to CUBE,
which sends two media streams towards the Recording Server
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
CUBE Phone Proxy
Phone Registration without VPN
Public
Network Users
TLS/SRTP
SIP/RTP
SIP
SIP Line-Side
SIP LineSide
H.323
ACCESS SIDE
CUBE
CUBE
SIP Trunk
Side
SBC
SP VOIP
Services
CORE SIDE
• Enables B2BUA line side support in CUBE for CUCM
• Allows you to have phones on the Internet connected to a CUBE at the edge of the
enterprise, replacing the need for ASA Phone Proxy by providing Secure RTP and TLS
based communications on the leg away from CUCM
• CUBE Phone Proxy must have a Public IP Address and cannot be behind a NAT
• IP Phones can be behind a NAT
• Access Side : Connection between Phone and CUBE
• Core Side : Connection between CUCM and CUBE
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
79
CUBE SIP Trunk Monitoring with OOD Options message
A
SP SIP Trunk
CUCM SIP Trunk
SP
SIP
CUBE
OOD Options
INVITE
DP 100 =
ACTIVE
200 OK
•
Out-of-dialog OPTIONS message sent
to check the status of the SIP Trunk
INVITE
•
The dial-peer is “busyout” if it does
not receive a response within a
configurable time period
•
For an INVITE that matches a
“busyout” dial-peer, CUBE sends “503
Service Unavailable”
•
If there is a secondary dial-peer
configured, the call will be re-routed
the secondary path
200 OK
200 OK
OOD Options
Timeout – no
response
DP 100 = BUSYOUT
INVITE
OOD Options
503 Service Unavailable
OOD Options
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
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CUBE SIP Trunk Monitoring with OOD Options message
A
SP SIP Trunk
CUCM SIP Trunk
SP SIP
CUBE
dial-peer voice 100 voip
voice-class sip options-keepalive
up-interval 20 down-interval 20 retry 3
OOD Options
200 OK
INVITE
DP 100 =
ACTIVE
Three timers that can be configured:
• up-Interval: OPTIONS keepalive
timer interval for UP endpoint
• down-interval: OPTIONS keepalive
timer interval for DOWN endpoint
• retry: Retry count for OPTIONS
keepalive transmission
INVITE
200 OK
200 OK
OOD Options
Timeout – no
response
DP 100 = BUSYOUT
INVITE
Warning:
• Each dial-peer that has options
message configured sends out a
separate message.
• EEM Script can be used to busyout
other dial-peers
OOD Options
503 Service Unavailable
OOD Options
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
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81
OOD OPTIONS Ping Keepalive Enhancement
A
SP SIP Trunk
CUCM SIP Trunk
SP SIP
CUBE
OOD Options (DP 100)
200 OK
• Network bandwidth and process runtime are
wasted in CUBE and remote targets to sustain
duplicate OOD OPTIONS Ping heartbeat
keepalive connection
DP 100 : Session Target IPv4:1.1.1.1
INVITE
INVITE (DP 100)
200 OK
• Each dial-peer that has OPTIONS message
configured sends out a separate message, even
if the session targets are same
200 OK
OOD Options (DP 200)
• Consolidate SIP OOD Options Ping connections
by grouping SIP dial-peers with same OOD
Options Ping setup
200 OK
DP 200: Session Target IPv4:1.1.1.1
OOD Options (DP 300)
• New CLI : “voice class sip-keepalive-profile
<tag>” is used to define OOD OPTIONS Ping
setup
200 OK
DP 300: Session Target IPv4:1.1.1.1
OOD Options (DP 400)
200 OK
• Consolidated SIP OOD Options Ping connection
will then be established with a target for multiple
SIP dial-peers with the same target and OOD
Options Ping profile setup
DP 400: Session Target IPv4:1.1.1.1
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
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OOD OPTIONS Ping Keepalive Enhancement Configuration
voice class sip-options-keepalive 1
description UDP Options consolidation
down-interval 49
up-interval 180
retry 7
transport udp
dial-peer voice 1 voip
destination-pattern 6666
session protocol sipv2
session target ipv4:10.104.45.253
voice-class sip options-keepalive profile 1
Sample Show command output
CUBE#sh voice class sip-options-keepalive 1
Voice class sip-options-keepalive: 1
Single OOD Option
Ping Group applied
to multiple dial-peers
with same session
targets
AdminStat: Up
Description: UDP Options consolidation
Transport: udp
Sip Profiles: 0
Interval(seconds) Up: 180
Down: 49
Retry: 7
dial-peer voice 2 voip
destination-pattern 5555
session protocol sipv2
session target ipv4:10.104.45.253
voice-class sip options-keepalive profile 1
Peer Tag
Server Group
--------
------------
OOD SessID
OOD Stat
IfIndex
----------
--------
-------
1
4
Active
9
2
4
Active
10
OOD SessID: 4
OOD Stat: Active
Target: ipv4:10.104.45.253
Transport: udp
Sip Profiles: 0
•
With OOD Options Ping Keepalive group, an options ping keepalive connection is established on per remote target base as opposed an options
ping keepalive connection established per dial-peer basis
•
Up to 10,000 “voice class sip-options-keepalive <tag>” can be defined per system
•
Either legacy “sip options-keepalive” or the new “sip options-keepalive profile <tag>” can be configured on a dial-peer
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
83
SIP Trunk to TDM PSTN Failover
• Collocated Cisco Unified Border Element and TDM GW offers:
•
Alternate call routing path (upon congestion or SIP Trunk failure)
•
Easy SIP Trunking migration
SIP Trunk
(Primary)
SBC
IP
SP
VoIP
CUBE
TDM Trunk
(Secondary)
•
•
•
BRKUCC-2934
dial-peer voice 10 voip
description “Primary path to SIP Trunk provider”
destination-pattern 91[2-9]..[2-9]......
session protocol sipv2
session target ipv4:10.10.10.1
voice-class sip options-keepalive
Deployed in small to medium sized
enterprise networks
Deployed at branch locations for
PSTN calls during survivability
mode
Deployed at branch locations for
emergency services
© 2014 Cisco and/or its affiliates. All rights reserved.
dial-peer voice 20 pots
description “Secondary path to PSTN”
destination-pattern 91[2-9]..[2-9]......
preference 2
port 0/0/0:23
Cisco Public
84
SIP Trunking to more than one service provider
SIP SP-1
Standby
A
(10.10.10.2)
CUBE
Active
SIP SP-2
(20.20.20.2)
CUBE
Enterprise
Campus
Large enterprises are deploying more
than one SIP Trunk provider for:
• Alternate call routing
• Load balancing
CUBE with High
Availability
MPLS
SIP SP-1’s
network
interface loopback1
ip address 10.10.10.1 255.255.255.0
SIP SP-2’s
network
SRST
interface loopback2
ip address 20.20.20.1 255.255.255.0
dial-peer voice 10 voip
description “Primary path to SIP SP-1”
destination-pattern 91[2-9]..[2-9]......
CME
session protocol sipv2
session target ipv4:10.10.10.2
voice-class sip options-keepalive
TDM PBX
voice-class sip bind control source-interface loopback1
Enterprise
voice-class sip bind media source-interface loopback1
Branch Offices
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
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85
dial-peer voice 20 voip
description “Secondary path to SIP SP-2”
destination-pattern 91[2-9]..[2-9]......
session protocol sipv2
session target ipv4:20.20.20.2
preference 2
voice-class sip options-keepalive
voice-class sip bind control source-interface loopback2
voice-class sip bind media source-interface loopback2
NOTE: Dual SPs can be used for outbound calls, but cannot
be utilized for inbound calls
CUBE High Availability Options
• Inbox redundancy
•
•
•
ASR1006
ASR 1006
Stateful failover
Local redundancy
Dual Forwarding
plane HW
Dual Control plane
HW (CPU)
ASR(config)#redundancy
ASR-RP2(config-red)#mode sso
ASR-RP2(config-red)#end
• L2 Box-to-Box redundancy
•
•
•
Active
ISR G2/4451-X (Stateful failover)
ASR 1001/2/4/6 (Stateful failover)
Local redundancy (Both routers must
be physically located on the same
Ethernet LAN)
Not supported across data centers
Only 1 RP and 1 ESP in ASR1006
•
•
CUBE
Virtual
IP
Virtual
IP
SIP SP
CUBE
Standby
• Clustering with load balancing
•
•
•
BRKUCC-2934
All platforms
Load balancing by
• SP call agent
• Cisco Unified SIP Proxy
Local and geographical redundancy
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
SIP SP
CUSP
86
CUSP
CUBE HA Design on ISR-G2 for Box-to-Box Redundancy
CUBE-1
Gig0/0 – 10.10.1.11
Gig0/1 – 128.107.60.71
HSRP
Group 0
Keepalives
128.107.60.73
CUBE
HSRP
Group 10
10.10.1.13
10.10.1.10
SP IP
Network
Y.Y.Y.Y
CUCM
Gig0/0 – 10.10.1.12
LAN Virtual IP
CUBE
Gig0/1 – 128.107.60.72
CUBE-2
WAN Virtual IP
•
All signaling is sourced from/to the Virtual IP Address
•
Lower address for both the interfaces (Gig0/0 and Gig0/1) should be on the same platform, which is used as
a tie breaker for the HSRP Active state
•
HSRP Group number should be unique to a pair/interface combination on the same L2
•
Both interfaces of the same group have to be configured with the same priority
•
Call flows requiring DSPs will be preserved in a future release
•
Both the CUBEs must be running on the same type of platform and IOS version and identical configuration
•
Upon failover, the ACTIVE CUBE goes through a reload
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
87
CUBE Configuration on ISR-G2 Box-to-Box Redundancy
CUBE-1
Gig0/0 – 10.10.1.11
Gig0/1 – 128.107.60.71
HSRP
Group 0
Keepalives
128.107.60.73
CUBE
HSRP
Group 10
10.10.1.13
10.10.1.10
Y.Y.Y.Y
CUCM
Gig0/0 – 10.10.1.12
CUBE 1
CUBE-2
Define Redundancy scheme: Creates
interdependency b/w CUBE redundancy & HSRP
voice service voip
mode border-element
allow-connections sip to sip
redundancy
ipc zone default
association 1
no shutdown
protocol sctp
local-port 5000
local-ip 10.10.1.11
remote-port 5000
remote-ip 10.10.1.12
BRKUCC-2934
Gig0/1 – 128.107.60.72
CUBE
LAN Virtual IP
redundancy inter-device
scheme standby SB
SP IP
Network
© 2014 Cisco and/or its affiliates. All rights reserved.
Turn on CUBE Redundancy
IPC configuration :
Allows the ACTIVE
CUBE to tell the
STANDBY about the
state of the calls
Cisco Public
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WAN Virtual IP
CUBE 2
redundancy inter-device
scheme standby SB
voice service voip
mode border-element
allow-connections sip to sip
redundancy
ipc zone default
association 1
no shutdown
protocol sctp
local-port 5000
local-ip 10.10.1.12
remote-port 5000
remote-ip 10.10.1.11
CUBE Configuration on ISR-G2 Box-to-Box Redundancy
CUBE-1
Gig0/0 – 10.10.1.11
Gig0/1 – 128.107.60.71
HSRP
Group 0
Keepalives
128.107.60.73
CUBE
HSRP
Group 10
10.10.1.13
10.10.1.10
SP IP
Network
Y.Y.Y.Y
CUCM
Gig0/0 – 10.10.1.12
CUBE
LAN Virtual IP
CUBE-2
CUBE 1
Gig0/1 – 128.107.60.72
WAN Virtual IP
CUBE 2
interface GigabitEthernet0/0
ip address 10.10.1.11 255.255.255.0
standby version 2
standby 0 ip 10.10.1.13
standby delay minimum 30 reload 60
standby 0 name SB
Inside
interfaces:
HSRP group 0
interface GigabitEthernet0/0
ip address 10.10.1.12 255.255.255.0
standby version 2
standby 0 ip 10.10.1.13
standby delay minimum 30 reload 60
standby 0 name SB
interface GigabitEthernet0/1
ip address 128.107.60.71 255.255.255.0
standby version 2
standby 10 ip 128.107.60.73
standby delay minimum 30 reload 60
Outside
interfaces:
HSRP group 10
interface GigabitEthernet0/1
ip address 128.107.60.72 255.255.255.0
standby version 2
standby 10 ip 128.107.60.73
standby delay minimum 30 reload 60
Configure
Interface
Tracking
track 1 interface Gig0/0 line-protocol
track 2 interface Gig0/1 line-protocol
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
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89
track 1 interface Gig0/0 line-protocol
track 2 interface Gig0/1 line-protocol
CUBE Configuration on ISR-G2 Box-to-Box Redundancy
Configuration on Active and Standby
dial-peer voice 100 voip
description TO SERVICE PROVIDER
destination-pattern 9T
session protocol sipv2
session target ipv4:y.y.y.y
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
!
dial-peer voice 200 voip
description TO CUCM
destination-pattern 555….
session protocol sipv2
session target ipv4:10.10.1.10
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
!
ip rtcp report interval 3000
!
gateway
media-inactivity-criteria all
timer receive-rtcp 5
timer receive-rtp 86400
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
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90
Bind traffic destined to the outside (SP SIP
trunk) to the outside Physical interface.
CUBE HA does not work with loopback
interfaces as they are always up
Bind traffic destined to the inside (CUCM or
IP-PBX) to the inside Physical interface.
CUBE HA does not work with loopback
interfaces as they are always up
Configure media inactivity feature to
clean up any calls that may not
disconnect after a failover
CUBE HA Design on ASR for Box-to-Box Redundancy
CUBE-1
GE 0/0/0 – 10.10.1.1
GE 0/0/1 – 20.20.1.1
redundancy
rii 0
30.30.3.1
GE 0/0/2
30.30.3.2
20.20.1.3
redundancy
rii 2
CUBE
GE 0/0/2
Keepalives
10.10.1.3
10.10.1.10
SP IP
Network
CUCM
GE 0/0/0 – 10.10.1.2
LAN Virtual IP
GE 0/0/1 – 20.20.1.2
CUBE
WAN Virtual IP
CUBE-2
•
All signaling is sourced from/to the Virtual IP Address
•
Lower address for all the interfaces (Gig0/0/0, Gig0/0/1, and Gig0/0/2) should be on the same platform
•
Redundancy Interface Identifier, rii (HSRP Group number) should be unique to a pair/interface combination on the same L2
•
Configuration on both the CUBEs must be identical including physical configuration and must be running on the same type of
platform and IOS version
•
Call flows requiring DSPs will be preserved in a future release
•
Upon failover, starting XE3.11, the ACTIVE CUBE can be moved to PROTECTED state to avoid reload
•
It is mandatory to use separate interface for redundancy (RG Control/data, Gig0/0/2). i.e interface used for traffic cannot be
used for HA keepalives and checkpointing.
•
CUBE B2B HA on ASR is not supported over a crossover cable connection for the RG-control/data link’.
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
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91
CUBE Configuration on ASR Box-to-Box Redundancy
CUBE-1
GE 0/0/0 – 10.10.1.1
For Your
Reference
GE 0/0/1 – 20.20.1.1
redundancy
rii 0
30.30.3.1
GE 0/0/2
30.30.3.2
20.20.1.3
redundancy
rii 2
CUBE
GE 0/0/2
Keepalives
10.10.1.3
10.10.1.10
SP IP
Network
CUCM
GE 0/0/0 – 10.10.1.2
LAN Virtual IP
CUBE 1
redundancy
mode none
application redundancy
group 1
name voice-b2bha
priority 100
control GigabitEthernet 0/0/2 protocol 1
data GigabitEthernet 0/0/2 protocol 1
timers delay 30 reload 60
CUBE-2
Disables software redundancy
For ASR1006: mode rpr
voice service voip
mode border-element
allow-connections sip to sip
redundancy-group 1
BRKUCC-2934
GE 0/0/1 – 20.20.1.2
CUBE
© 2014 Cisco and/or its affiliates. All rights reserved.
Configure RG
Group for use
with CUBE HA
Turn on CUBE
Redundancy
Cisco Public
92
WAN Virtual IP
CUBE 2
redundancy
mode none
application redundancy
group 1
name voice-b2bha
priority 100
control GigabitEthernet 0/0/2 protocol 1
data GigabitEthernet 0/0/2 protocol 1
timers delay 30 reload 60
voice service voip
mode border-element
allow-connections sip to sip
redundancy-group 1
CUBE Configuration on ASR Box-to-Box Redundancy
CUBE-1
GE 0/0/0 – 10.10.1.1
For Your
Reference
GE 0/0/1 – 20.20.1.1
10.10.1.3
30.30.3.1
GE 0/0/2
30.30.3.2
20.20.1.3
redundancy
rii 2
CUBE
GE 0/0/2
Keepalives
redundancy
rii 0
10.10.1.10
SP IP
Network
CUCM
GE 0/0/0 – 10.10.1.2
LAN Virtual IP
CUBE
CUBE-2
CUBE 1
track 1 interface GigabitEthernet 0/0/0
line-protocol
track 2 interface GigabitEthernet 0/0/1
line-protocol
redundancy
application redundancy
group 1
track 1 shutdown
track 2 shutdown
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
GE 0/0/1 – 20.20.1.2
WAN Virtual IP
CUBE 2
Track interfaces
to trigger
switchover
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93
track 1 interface GigabitEthernet 0/0/0
line-protocol
track 2 interface GigabitEthernet 0/0/1
line-protocol
redundancy
application redundancy
group 1
track 1 shutdown
track 2 shutdown
CUBE Configuration on ASR Box-to-Box Redundancy
CUBE-1
GE 0/0/0 – 10.10.1.1
For Your
Reference
GE 0/0/1 – 20.20.1.1
redundancy
rii 0
30.30.3.1
GE 0/0/2
30.30.3.2
20.20.1.3
redundancy
rii 2
CUBE
GE 0/0/2
Keepalives
10.10.1.3
10.10.1.10
SP IP
Network
CUCM
GE 0/0/0 – 10.10.1.2
CUBE
LAN Virtual IP
GE 0/0/1 – 20.20.1.2
CUBE-2
CUBE 1
WAN Virtual IP
CUBE 2
interface GigabitEthernet0/0/0
ip address 10.10.1.1 255.255.255.0
redundancy rii 0
redundancy group 1 ip 10.10.1.3 exclusive
Inside interfaces:
Redundancy Interface
Identifier 0
interface GigabitEthernet0/0/0
ip address 10.10.1.2 255.255.255.0
redundancy rii 0
redundancy group 1 ip 10.10.1.3 exclusive
interface GigabitEthernet0/0/1
ip address 20.20.1.1 255.255.255.0
redundancy rii 2
redundancy group 1 ip 20.20.1.3
Outside interfaces:
Redundancy Interface
Identifier 2
interface GigabitEthernet0/0/1
ip address 20.20.1.2 255.255.255.0
redundancy rii 2
redundancy group 1 ip 20.20.1.3
interface GigabitEthernet 0/0/2
ip address 30.30.3.1 255.255.255.0
Interface for control &
checkpoint data traffic
BRKUCC-2934
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94
interface GigabitEthernet 0/0/2
ip address 30.30.3.2 255.255.255.0
CUBE Configuration on ASR Box-to-Box Redundancy
Configuration on Active and Standby
dial-peer voice 100 voip
description to-SIP-SP
destination-pattern 9T
session protocol sipv2
session target ipv4:y.y.y.y
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
!
dial-peer voice 200 voip
description to-CUCM
destination-pattern 555….
session protocol sipv2
session target ipv4:10.10.1.10
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
!
ip rtcp report interval 3000
!
gateway
media-inactivity-criteria all
timer receive-rtcp 5
timer receive-rtp 86400
BRKUCC-2934
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95
Bind traffic destined to the outside (SP SIP
trunk) to the outside Physical interface to
make sure it uses the virtual IP address as
the source-IP for all calls
Bind traffic destined to the inside (CUCM or
IP-PBX) to the inside Physical interface
Configure media inactivity feature to
clean up any calls that may not
disconnect after a failover
ASR B2B Redundancy : PROTECTED MODE
•
Default failover redundancy behavior in a B2B HA pair is to reload the affected router to avoid out-of-sync
conditions/Split brain
•
Starting XE3.11, an ASR can be configured to transition into PROTECTED mode
•
In PROTECTED mode
o
o
•
Bulk sync request, Call checkpointing, and incoming call processing are disabled
The router in PROTECTED mode needs to be manually reloaded to come out of this state
The PROTECTED mode is enabled with the following CLI
voice service voip
no redundancy-reload !  Default is ‘redundancy-reload’
•
Track for the RG Control/data interface (GE0/0/2) with the same ‘track <id> shutdown’ under redundancy
group needs to be added
track 1 interface GigabitEthernet0/0/0 line-protocol
track 2 interface GigabitEthernet0/0/1 line-protocol
track 3 interface GigabitEthernet0/0/2 line-protocol !  Track for RG Control/data interface
redundancy
application redundancy
group 1
track 1 shutdown
track 2 shutdown
track 3 shutdown
BRKUCC-2934
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96
Multicast MoH to Unicast MoH Conversion- CUBE
Multicast MoH
Hold
A
♬ ♬ ♬
Unicast MoH
CUBE
♬ ♬
SP
SIP
Active Call
ccm-manager music-on-hold
ip multicast-routing distributed
“ip pim dense-mode” under interface
• Extends the ability for enterprises to play Multicast MoH to Service Providers
• CUBE converts Multicast MoH from the MoH server to unicast MoH streamed to
the service provider
• Provides the ability to play Multicast MoH over the WAN from the MoH server at
the HQ to the CUBE at the remote branch (distributed architecture), saving WAN
bandwidth
BRKUCC-2934
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97
Agenda
• SIP Trunking and CUBE Overview
• SIP Trunking Design & Deployment Models
• CUBE Architecture
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE
• CUBE Management & Troubleshooting
• Futures & Key Takeaways
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
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98
CUBE Monitoring
• Network Management Tools can be used to monitor key CUBE
statistics like SIP Trunk status, Trunk utilization, Call Arrival Rate,
Call Success/Failure count, voice quality metrics etc..
• Network Management Tools can send SNMP Queries to CUBE
• CUBE responds to the SNMP queries with real time values
of the monitored objects
• CUBE can also send SNMP Traps to alert the
network management tool of certain events like
SIP Trunk failure, link down, high CPU etc..
Network
Management
Tool
SNMP
Query
SNMP
Response
SIP
H.323 or SIP
CUBE
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
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SBC
99
SP IP
Network
Some Network Management Tools:
-
Cisco Unified Operations Manager
Arcana Networks
Solarwinds
For Your
Reference
CUBE Monitoring
Area
Information
Method
Router Health
CPU, Memory, I/f
 CISCO-PROCESS-MIB, cpmCPUTotal5minRev
 CISCO-MEMORY-POOL-MIB, ciscoMemoryPoolTable
 IF-MIB, IfEntry
SIP Trunk Status
SIP Trunk Status
 SIP OOD Options Ping, CLI dial-peer status
Trunk Utilization




Call Arrival Rate
 CUBE 1.4: CISCO-VOICE-DIAL-CONTROL-MIB, cvCallRateMonitor
Call Success/Failure
 DIAL-CONTROL-MIB, dialCtlPeerStatsSuccessCalls, dialCtlPeerStatsAcceptCalls,
dialCtlPeerStatsFailCalls, dialCtlPeerStatsRefuseCalls
 CISCO-SIP-UA-MIB, cSipStatsErrClient, cSipStatsErrServer, cSipStatsGlobalFail
SIP retries
 CISCO-SIP-UA-MIB, cSipStatsRetry
DSP Availability
 CISCO-DSP-MGMT-MIB, cdspCardResourceUtilization, cdspDspfarmUtilObjects
Transcoding util.
 CUBE 1.4: CISCO-DSP-MGMT-MIB, cdspTotAvailTranscodeSess, cdspTotUnusedTranscodeSess
MTP utilization
 CUBE 1.4: CISCO-DSP-MGMT-MIB, cdspTotAvailMtpSess, cdspTotUnusedMtpSess
Loss, delay, jitter
 CISCO-VOICE-DIAL-CONTROL-MIB, cvVoIPCallActiveTable
IP SLA
 CISCO-RTTMON-RTP-MIB, rttMonJitterStatsTable , rttMonLatestJitterOperTable
Traffic Reports (Calls,
Sessions, Capacity Planning,
Errors)
Media Resources
(DSPs)
Voice Quality
CUBE 1.4: CISCO-VOICE-DIAL-CONTROL-MIB, cvCallVolume
Older CUBE: DIAL-CONTROL-MIB, callActive
CISCO-DIAL-CONTROL-MIB, cCallHistoryTable
CUBE 8.5: SIP RAI Trunk Utilization
More info in CUBE Management and Manageability Specification at:
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/white_paper_c11-613550.html
BRKUCC-2934
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100
Introducing ManageExpress® Border Manager
• Simplified provisioning
and management
• Uniform policies across all SBCs
• Real time 911/211 alerting
and monitoring
• Voice quality monitoring
• Reduce operational costs
• Available on the Cisco price list
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
101
Topology with Real Time Monitoring
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
102
Voice Call Quality Monitoring on CUBE
• Three mechanism exist to monitor call quality statistics
1. End of call statistics in BYE message
2. End of call CDR with 5 critical call parameters (MoSQe, Delay, Jitter, Loss, OoO)
3. Real time export of 30+ AQM via Flexible NetFlow
CDR Example or MIB file: CISCO-VOICE-DIAL-CONTROL-MIB
<MOS-Con>4.4072</MOS-Con>
<round-trip-delay>1 ms</round-trip-delay>
<receive-delay>64 ms</receive-delay>
<voice-quality-total-packet-loss>0.0000 %</ voice-quality-total-packet-loss>
< voice-quality-out-of-order>0.0000 %</ voice-quality-out-of-order>
•
CDR will be sent to Radius server at the end of a call if AAA accounting is
configured
BRKUCC-2934
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103
Audio Quality Monitor using Flexible NetFlow
• AQM uses FNF to export up to 30 voice quality metrics measured by “media monitoring” CLI
• To help the NetFlow collector to process the flow record, AQM also reports call related
information such as calling number, called number, call setup time, etc
• Will be available on ASR starting XE3.14
Configuration to enable VQM Calculation
voice service voip
media monitoring [num] persist
! The max number of channels used for monitoring
media statistics
! Enable media statistics for VQM calculation
dial-peer voice [tag] voip
media monitoring
! Enable media monitoring on this dial-peer, every call leg matching this dial-peer will be monitored
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
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104
For Your
Reference
FNF Configuration
flow record type performance-monitor aqm
match ipv4 source address
match ipv4 destination address
match transport source-port
match transport destination-port
collect application voice number called
collect application voice number calling
collect application voice setup time
collect application voice call duration
collect application voice rx bad-packet
collect application voice rx out-of-sequence
collect application voice codec id
collect application voice play delay current
collect application voice play delay minimum
collect application voice play delay maximum
collect application voice sip call-id
collect application voice router global-call-id
collect application voice delay round-trip
collect application voice delay end-point
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
collect application voice r-factor 1
collect application voice r-factor 2
collect application voice mos conversation
collect application voice mos listening
collect application voice concealment-ratio average
collect application voice jitter configured type
collect application voice jitter configured minimum
collect application voice jitter configured maximum
collect application voice jitter configured initial
collect application voice rx early-packet count
collect application voice rx late-packet count
collect application voice jitter buffer-overrun
collect application voice packet conceal-count
!
105
FNF Configuration – Cont’d
flow exporter aqm-exporter
destination <IP addr>
source FastEthernet8
transport udp 2055
option application-attributes
!
flow monitor type performance-monitor aqm-mon
record aqm
exporter aqm-exporter
cache entries 1000
cache timeout synchronized 10
history size 60 timeout 5
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
For Your
Reference
class-map match-all aqm-class
match application rtp
match application attribute media-type audio
!
policy-map type performance-monitor aqm-policy
class aqm-class
flow monitor aqm-mon
!
interface FastEthernet8
ip address 10.10.10.11 255.255.0.0
load-interval 30
duplex full
speed 100
service-policy type performance-monitor input aqm-policy
service-policy type performance-monitor output aqm-policy
106
For Your
Reference
Viewing AQM
CUBE# show call active voice stats
DSP/TX: PK=0, SG=0, NS=0, DU=0, VO=0
DSP/RX: PK=34, SG=0, CF=1, RX=660, VO=660, BS=0, BP=0, LP=0, EP=0
DSP/PD: CU=69, MI=69, MA=69, CO=0, IJ=0.0000
DSP/PE: PC=0, IC=0, SC=0, RM=0, BO=0, EE=0
DSP/LE: TP=0, TX=0, RP=0, RM=0, BN=0, ER=0, AC=0
DSP/ER: RD=0, TD=0, RC=0, TC=0
DSP/IC: IC=0
DSP/EC: CI=g711alaw, FM=5, FP=1, VS=0, GT=1.0000, GR=1.0000, JD=adaptive, JN=60, JM=40, JX=1000
DSP/KF: KF=0.0000, AV=0.0000, MI=0.0000, BS=0.0000, NB=0, FL=0, NW=0, VR=0.0
DSP/CS: CR=0.0000, AV=0.0000, MX=0.0000, CT=0, TT=0, OK=0, CS=0, SC=0, TS=50, DC=0
DSP/RF: ML=-1.0000, MC=-1.0000, R1=-1, R2=-1, IF=0, ID=0, IE=0, BL=25, R0=93, VR=2.0
DSP/UC: U1=0, U2=0, T1=0, T2=0
DSP/DL: RT=0, ED=0
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
107
AQM viewing through ARCANA’s MEBM
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
108
AQM stats per network segment
BRKUCC-2934
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109
Incremental Metrics are provided throughout the call
BRKUCC-2934
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110
One-on-One Demo of ARCANA’s MEBM®
• One on one demonstrations of ManageExpress Border
Manager: provisioning, management, monitoring and
reporting for CUBE.
• Whisper Suite at St. Regis
• Sign up here: http://bit.ly/arcanalive
BRKUCC-2934
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111
Troubleshooting of Calls
show cube status
Is CUBE Active ?
CUBE-Version : 9.0
SW-Version : 15.2.1T, Platform 2911
HA-Type : none
Licensed-Capacity : 200
debug voip ccapi inout
Oct 26 18:59:01.146: //-1/66A6B1BF8013/CCAPI
cc_api_call_setup_ind_common:
.................
Incoming Dial-peer=1, Progress Indication=NULL(0), Calling IE
Present=TRUE,
.................
Outgoing Dial-peer=100, Params=0x26E8574, Progress
Indication=NULL(0)
Is the call matching
right Dial-peers ?
Are we sending the
right SIP call to SP based
on their requirements ?
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
debug ccsip messages
Received:
INVITE sip:912025552000@14.128.101.24:5060 SIP/2.0
Date: Wed, 26 Oct 2011 18:59:01 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK,
UPDATE, REFER, SUBSCRIBE, NOTIFY
From: "Paul Hewson"
<sip:1500@10.88.156.166>;tag=90d94d92-6ee4-45aa-9f182d09025c1ee4-27352390
................
Cisco Public
112
SIP EO Debug
Example
Sent:
INVITE sip:1000@20.1.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 20.1.1.1:5060;branch=z9hG4bK1216FC
Remote-Party-ID: <sip:2000@20.1.1.1>;party=calling;screen=no;privacy=off
From: <sip:2000@20.1.1.1>;tag=48AE80-CD8
To: <sip:1000@20.1.1.2>
Date: Wed, 22 Jun 2011 12:33:15 GMT
Call-ID: A2F9661D-9C0211E0-803289BC-624E6E32@9.44.44.71
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2734093693-2617381344-2150402492-1649307186
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY, INFO, REGISTER
.........
.........
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
External
Network
10.1.1.1
20.1.1.1
SIP SP
CUBE
B2B User
Agent
 Outbound INVITE message
 Sent with destination number as 1000 and IP address
20.1.1.2 on port 5060
 Calling number is 2000 with source IP address of call is
20.1.1.1
 Cisco-GUID uniquely identifies this call leg
 “c” parameter identifies the IP address (20.1.1.1) that the
peer device should send the media to
 “m” parameter identifies:
 the type of call (audio)
 port number for media (16950)
 payload type for the 1st preferred codec (18 for G729)
 dtmf (101 for RFC2833)
 “a’” parameter identifies all the codecs and other
descriptors for this call leg
v=0
o=CiscoSystemsSIP-GW-UserAgent 2026 314 IN IP4 9.44.44.71
s=SIP Call
c=IN IP4 20.1.1.1
t=0 0
m=audio 16950 RTP/AVP 18 101
c=IN IP4 20.1.1.1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
BRKUCC-2934
Internal
Network
113
SIP EO Debug
Example
Sent:
INVITE sip:1000@20.1.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 20.1.1.1:5060;branch=z9hG4bK1216FC
Remote-Party-ID: <sip:2000@20.1.1.1>;party=calling;screen=no;privacy=off
From: <sip:2000@20.1.1.1>;tag=48AE80-CD8
To: <sip:1000@20.1.1.2>
Date: Wed, 22 Jun 2011 12:33:15 GMT
Call-ID: A2F9661D-9C0211E0-803289BC-624E6E32@9.44.44.71
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2734093693-2617381344-2150402492-1649307186
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY, INFO, REGISTER
.........
.........
Internal
Network
External
Network
10.1.1.1
20.1.1.1
SIP SP
Sent:
INVITE
CUBE
B2B User
Agent
 Outbound INVITE message
 Sent with destination number as 1000 and IP address
20.1.1.2 on port 5060
 Calling number is 2000 with source IP address of call is
20.1.1.1
 Cisco-GUID uniquely identifies this call leg
 Outbound INVITE message
 “c” parameter identifies the IP address (20.1.1.1) that the
peer device should send the media to
 “m” parameter identifies:
 the type of call (audio)
 port number for media (16950)
 payload type for the 1st preferred codec (18 for G729)
 dtmf (101 for RFC2833)
 “a’” parameter identifies all the codecs and other
descriptors for this call leg
v=0
o=CiscoSystemsSIP-GW-UserAgent 2026 314 IN IP4 9.44.44.71
s=SIP Call
c=IN IP4 20.1.1.1
t=0 0
m=audio 16950 RTP/AVP 18 101
c=IN IP4 20.1.1.1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
114
SIP EO Debug
Example
Sent:
INVITE sip:1000@20.1.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 20.1.1.1:5060;branch=z9hG4bK1216FC
Remote-Party-ID: <sip:2000@20.1.1.1>;party=calling;screen=no;privacy=off
From: <sip:2000@20.1.1.1>;tag=48AE80-CD8
To: <sip:1000@20.1.1.2>
Date: Wed, 22 Jun 2011 12:33:15 GMT
Call-ID: A2F9661D-9C0211E0-803289BC-624E6E32@9.44.44.71
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2734093693-2617381344-2150402492-1649307186
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY, INFO, REGISTER
.........
.........
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
10.1.1.1
20.1.1.1
CUBE
B2B User
Agent
 Outbound INVITE message
 Sent with destination number
as 1000 and IP address
20.1.1.2 on port 5060
 Calling number is 2000 with source IP address of call is
20.1.1.1
 Cisco-GUID uniquely identifies this call leg
To: <sip:1000@20.1.1.2>
BRKUCC-2934
External
Network
SIP SP
INVITE sip:1000@20.1.1.2:5060 SIP/2.0
v=0
o=CiscoSystemsSIP-GW-UserAgent 2026 314 IN IP4 9.44.44.71
s=SIP Call
c=IN IP4 20.1.1.1
t=0 0
m=audio 16950 RTP/AVP 18 101
c=IN IP4 20.1.1.1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Internal
Network
 “c”with
parameter
identifies the
IP address (20.1.1.1) that the
Sent
destination
number
peer device should send the media to
as 1000
and IP address
“m” parameter identifies:
20.1.1.2
5060
 theon
typeport
of call
(audio)
 port number for media (16950)
 payload type for the 1st preferred codec (18 for G729)
 dtmf (101 for RFC2833)
 “a’” parameter identifies all the codecs and other
descriptors for this call leg
115
SIP EO Debug
Example
Internal
Network
Sent:
INVITE sip:1000@20.1.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 20.1.1.1:5060;branch=z9hG4bK1216FC
Remote-Party-ID: <sip:2000@20.1.1.1>;party=calling;screen=no;privacy=off
From: <sip:2000@20.1.1.1>;tag=48AE80-CD8
To: <sip:1000@20.1.1.2>
Date: Wed, 22 Jun 2011 12:33:15 GMT
Call-ID: A2F9661D-9C0211E0-803289BC-624E6E32@9.44.44.71
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2734093693-2617381344-2150402492-1649307186
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY, INFO, REGISTER
.........
.........
SIP SP
10.1.1.1
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
CUBE
20.1.1.1
B2B User
Agent
 Outbound INVITE message
 Sent with destination number as 1000 and IP address
20.1.1.2 on port 5060
 Calling number is
2000 with source IP
address of call is 20.1.1.1
 Cisco-GUID uniquely identifies this call leg
From: <sip:2000@20.1.1.1>;tag=48AE80-CD8
v=0
o=CiscoSystemsSIP-GW-UserAgent 2026 314 IN IP4 9.44.44.71
s=SIP Call
c=IN IP4 20.1.1.1
t=0 0
m=audio 16950 RTP/AVP 18 101
c=IN IP4 20.1.1.1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
External
Network
 “c” parameter identifies the IP address (20.1.1.1) that the
peer device should send the media to

“m” parameter
identifies:
 Calling
number
is 2000
 the type of call (audio)
with
source
address
 port
number IP
for media
(16950)of
st
type for the 1 preferred codec (18 for G729)
call payload
is 20.1.1.1
 dtmf (101 for RFC2833)
 “a’” parameter identifies all the codecs and other
descriptors for this call leg
116
SIP EO Debug
Example
Internal
Network
Sent:
INVITE sip:1000@20.1.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 20.1.1.1:5060;branch=z9hG4bK1216FC
Remote-Party-ID: <sip:2000@20.1.1.1>;party=calling;screen=no;privacy=off
From: <sip:2000@20.1.1.1>;tag=48AE80-CD8
To: <sip:1000@20.1.1.2>
Date: Wed, 22 Jun 2011 12:33:15 GMT
Call-ID: A2F9661D-9C0211E0-803289BC-624E6E32@9.44.44.71
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2734093693-2617381344-2150402492-1649307186
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY, INFO, REGISTER
.........
.........
External
Network
SIP SP
10.1.1.1
Cisco-Guid: 2734093693-2617381344-2150402492-1649307186
B2B User
v=0
o=CiscoSystemsSIP-GW-UserAgent 2026 314 IN IP4 9.44.44.71
s=SIP Call
c=IN IP4 20.1.1.1
t=0 0
m=audio 16950 RTP/AVP 18 101
c=IN IP4 20.1.1.1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
CUBE
20.1.1.1
Agent
 Outbound INVITE message
 Sent with destination number as 1000 and IP address
20.1.1.2 on port 5060
 Calling number is
2000 with source IP address of call is 20.1.1.1
 Cisco-GUID uniquely
identifies this call leg
 “c” parameter identifies the IP address (20.1.1.1) that the
peer device should send the media to
 “m” parameter identifies:
 the type of
call (audio)
 Cisco-GUID
uniquely
 port number for media (16950)
identifies
this
call
leg1st preferred codec (18 for G729)
 payload
type
for the
 dtmf (101 for RFC2833)
 “a’” parameter identifies all the codecs and other
descriptors for this call leg
117
SIP EO Debug
Example
Sent:
INVITE sip:1000@20.1.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 20.1.1.1:5060;branch=z9hG4bK1216FC
Remote-Party-ID: <sip:2000@20.1.1.1>;party=calling;screen=no;privacy=off
From: <sip:2000@20.1.1.1>;tag=48AE80-CD8
To: <sip:1000@20.1.1.2>
Date: Wed, 22 Jun 2011 12:33:15 GMT
Call-ID: A2F9661D-9C0211E0-803289BC-624E6E32@9.44.44.71
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2734093693-2617381344-2150402492-1649307186
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY, INFO, REGISTER
.........
.........
Internal
Network
External
Network
10.1.1.1
20.1.1.1
SIP SP
CUBE
B2B User
Agent
 Outbound INVITE message
 “c” parameter
the IP
address
 Sent withidentifies
destination number
as 1000
and IP address
20.1.1.2 on port 5060
(20.1.1.1)
that the peer device should
 Calling number is
send the2000
media
to IP address of call is 20.1.1.1
with source
 Cisco-GUID uniquely identifies this call leg
c=IN IP4 20.1.1.1
 “c” parameter identifies the IP address
(20.1.1.1) that the peer device should
send the media to
 “m” parameter identifies:
 the type of call (audio)
 port number for media (16950)
 payload type for the 1st preferred codec (18 for G729)
 dtmf (101 for RFC2833)
 “a’” parameter identifies all the codecs and other
descriptors for this call leg
v=0
o=CiscoSystemsSIP-GW-UserAgent 2026 314 IN IP4 9.44.44.71
s=SIP Call
c=IN IP4 20.1.1.1
t=0 0
m=audio 16950 RTP/AVP 18 101
c=IN IP4 20.1.1.1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
c=IN IP4 20.1.1.1
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
118
SIP EO Debug
Example
Internal
Network
Sent:
INVITE sip:1000@20.1.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 20.1.1.1:5060;branch=z9hG4bK1216FC
Remote-Party-ID: <sip:2000@20.1.1.1>;party=calling;screen=no;privacy=off
From: <sip:2000@20.1.1.1>;tag=48AE80-CD8
To: <sip:1000@20.1.1.2>
Date: Wed, 22 Jun 2011 12:33:15 GMT
Call-ID: A2F9661D-9C0211E0-803289BC-624E6E32@9.44.44.71
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2734093693-2617381344-2150402492-1649307186
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY, INFO, REGISTER
.........
.........
v=0
m=audio
16950 RTP/AVP 18 101
SIP SP
10.1.1.1
© 2014 Cisco and/or its affiliates. All rights reserved.
CUBE
20.1.1.1
B2B User
Agent
 Outbound INVITE
message
 m” parameter
identifies:
 Sent with destination number as 1000 and IP address
 the type
ofoncall
20.1.1.2
port (audio)
5060

Calling
number
is
 port number for media (16950)
2000 with source IP address
of call is 20.1.1.1
 payload
type uniquely
for theidentifies
1st preferred
 Cisco-GUID
this call leg
codec (18 for G729)
 dtmf (101
for RFC2833)
“c” parameter
identifies the IP address
o=CiscoSystemsSIP-GW-UserAgent 2026 314 IN IP4 9.44.44.71
s=SIP Call
c=IN IP4 20.1.1.1
t=0 0
m=audio 16950 RTP/AVP 18 101
c=IN IP4 20.1.1.1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
BRKUCC-2934
External
Network
Cisco Public
119
(20.1.1.1) that the peer device should
send the media to
 “m” parameter identifies:
 the type of call (audio)
 port number for media (16950)
 payload type for the 1st preferred
codec (18 for G729)
 dtmf (101 for RFC2833)
 “a’” parameter identifies all the codecs and other
descriptors for this call leg
SIP EO Debug
Example
Internal
Network
Sent:
INVITE sip:1000@20.1.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 20.1.1.1:5060;branch=z9hG4bK1216FC
Remote-Party-ID: <sip:2000@20.1.1.1>;party=calling;screen=no;privacy=off
From: <sip:2000@20.1.1.1>;tag=48AE80-CD8
To: <sip:1000@20.1.1.2>
Date: Wed, 22 Jun 2011 12:33:15 GMT
Call-ID: A2F9661D-9C0211E0-803289BC-624E6E32@9.44.44.71
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2734093693-2617381344-2150402492-1649307186
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY, INFO, REGISTER
.........
.........
External
Network
SIP SP
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
10.1.1.1
CUBE
20.1.1.1
B2B User
Agent
 Outbound INVITE message
 Sent with destination number as 1000 and IP address
20.1.1.2 on port 5060
 Calling number is
2000 with source IP address of call is 20.1.1.1
 Cisco-GUID uniquely identifies this call leg
 “c” parameter identifies the IP address
(20.1.1.1) that the peer device should
send the media to
 “a’” parameter
identifies
all the codecs
 “m” parameter
identifies:
 the type of call
and other descriptors
for(audio)
this call leg
 port number for media (16950)
 payload type for the 1st preferred codec (18 for G729)
 dtmf (101 for RFC2833)
 “a’” parameter identifies all the codecs
and other descriptors for this call leg
v=0
o=CiscoSystemsSIP-GW-UserAgent 2026 314 IN IP4 9.44.44.71
s=SIP Call
c=IN IP4 20.1.1.1
t=0 0
m=audio 16950 RTP/AVP 18 101
c=IN IP4 20.1.1.1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
120
New CUBE Serviceability Features
Call Arrival Rate
Example:
show call history stats cps
Histogram for Call rate
Histogram for Concurrent calls
Histogram for Call duration
Histogram for SIP message rate
High/Low watermark for Call Rate
High/Low watermark for Concurrent calls
High/Low watermark for SIP message rate
1122222357676678753222211111122247545789774322213311112245654598843333222
10
9
*
*
8
*
**
***
7
* * ***
*
*****
*
##*
6
********
*
*****
** *##*
5
*########*
#* *####*
*######*
4
*########*
*#***####**
*########*
3
**########**
*#########**
**
*########*****
2
******#########*****
****##########**** **
***########********
1 *######################################################################*
0....5....1....1....2....2....3....3....4....4....5....5....6....6....7..
0
5
0
5
0
5
0
5
0
5
0
5
0
Call switching rate / CPS (last 72 hours)
* = maximum calls/s
# = average calls/s
Histogram for Call Failure Rate
High/Low watermark for Call Failure Rate
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
121
New CUBE Serviceability Features
Total Number of Active Calls
 A single call can have multiple calllegs. To determine the total number
of active calls from call-legs is
challenging
 CLI added to display the value of
current number of active
(connected) calls on CUBE
 The table defines the relation
between call-legs and number of
active calls
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Router# show call active total-calls
Total Number of Active Calls : 10
Call Flow
Call-legs
Connected
call
Basic call (audio/video)
2
1
Transferred call (Refer
handling)
3
2
Transcoded call (SCCP)
4
1
Calls after rotary/hunt
2+x
1
Forwarded calls (CUBE
handling)
3
1
Forked call (media forking)
3
2
Forked call (signaling forking)
2
1
Cisco Public
122
Avoiding Non-call-context Debug Logs
• Many times SIP debugs contain unrelated debugs that are not useful in
debugging issues related to call failures
• Starting CUBE 10.0.1, non-call-context debugs will not be printed when
debug ccsip is issued
• If a message is not part of any call, that debug will not be printed
• Affected messages: OPTIONS, REGISTER, SUBSCRIBE/NOTIFY
• To see the above messages in debugs, issue the following command
debug ccsip non-call
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
123
Debugging Made Easier
Categorize Debugs based on Functionality
 Categorization based on
Functionality
1.
2.
3.
4.
5.
6.
7.
Audio/video/sdp/control
Configuration /sip-transport
CAC
DTMF/FAX/Line-side
Registration
Sdp - passthrough
Sip-profile/SRTP/transcoder
Router# debug ccsip feature < audio | cac | config |
control | dtmf | fax | line | misc | misc-features | parse |
registration | sdp-negotiation | sdp-passthrough | sipprofiles | sip-transport | srtp | supplementary-services |
transcoder | video >
Example: enabling DTMF and audio debugs only with default log level is considered.
DTMF(32) debug code
CUBE#sh debugging
CCSIP SPI: SIP info debug tracing is enabled (filter is OFF)
CCSIP SPI: audio debugging for ccsip info is enabled (active)
CCSIP SPI: dtmf debugging for ccsip info is enabled (active)
Audio(2) debug code
May 21 17:54:53.377: //444/5FE632EB8479/SIP/Info/verbose/32/sipSPI_ipip_store_channel_info: dtmf negotiation done, storing
negotiated dtmf = 0,
May 21 17:54:53.377: //444/5FE632EB8479/SIP/Info/info/2/sipSPIUpdateCallEntry:
Call 444 set InfoType to SPEECH
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
124
Debugging Made Easier
Categorize Debugs based on Functionality
|----------------------------------------------| show cube debug category caodes values.
|----------------------------------------------| Indx | Debug Name
|
Value
|----------------------------------------------| 01 | SDP Debugs
|
1
| 02 | Audio Debugs
|
2
| 03 | Video Debugs
|
4
| 04 | Fax Debugs
|
8
| 05 | SRTP Debugs
|
16
| 06 | DTMF Debugs
|
32
| 07 | SIP Profiles Debugs |
64
| 08 | SDP Passthrough Deb |
128
| 09 | Transcoder Debugs
|
256
| 10 | SIP Transport Debugs |
512
| 11 | Parse Debugs
|
1024
| 12 | Config Debugs
|
2048
| 13 | Control Debugs
|
4096
| 14 | Mischellaneous Debugs|
8192
| 15 | Supp Service Debugs |
16384
| 16 | Misc Features Debugs|
32768
| 17 | SIP Line-side Debugs |
65536
| 18 | CAC Debugs
|
131072
| 19 | Registration Debugs |
262144
|-----------------------------------------------
CUBE# show cube debug category codes
 This CLI is used to collect the
predefined debug features category
codes , which helps in analysis of
debugs manually.
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
125
Agenda
• SIP Trunking and CUBE Overview
• SIP Trunking Design & Deployment Models
• CUBE Architecture
• Transitioning to SIP Trunking using CUBE
• Advanced features on CUBE
• CUBE Management & Troubleshooting
• Futures & Key Takeaways
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
126
IP Trunk Evolution – Cutting edge designs
Cloud Connected Audio
Media Manipulation & Optimization
Improved quality of speech
by Noise Cancellation,
Acoustic shock prevention
Customer
Network
Speech corrupted with
background noise
A
SIP Trunk to
Webex
IP Cloud
SIP Trunk SP
Cisco WebEx
Collaboration Cloud
CUBE
Network based recording
SecureLogix
Application Layer
Voice Policy:
Media
Sense
UC
Application
Network
A
SIP Trunk SP
CUBE
Platform
BRKUCC-2934
conne
ction
Integration of Voice Policies
Partner
Application
Cisco
MediaSense
Cisco
peerin
WebEx
iPOP
g
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
127
 Centralized voice policy
creation/distribution
 Protection from external
harassing calls
 Service Abuse control
by internal users
 Enterprise-wide UC
reporting & analytics
 Compliance & Data
Leakage prevention
Key Takeaways
• It is a manageable transition from existing TDM based networks to SIP
networks using these network design techniques
• Enterprise SBC (Cisco Unified Border Element - CUBE) is an essential
component of a UC solution providing;
– Security, Session Management, Interworking, Demarcation
– An unmatched set of features and functionality
– Proven interoperability with 3rd party PBX vendors and different service providers
around the world (more than 160 countries)
– Integrated & revolutionary platforms
• Now is the time to deploy SIP Trunking in either a Centralized or a Distributed
solution to save money, simplify your topology and setup your infrastructure for
future services
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
128
Recommended Reading
BRKUCC-2934
VoDs on CUBE
https://learningnetworkstore.cisco.com
Email – dsladden@cisco.com
for Special Discount Code
Recommended e-Learning
Course on SIP
http://cisco.thesipschool.com
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
129
Participate in the “My Favorite Speaker” Contest
Promote Your Favorite Speaker and You Could be a Winner
• Promote your favorite speaker through Twitter and you could win $200 of Cisco
Press products (@CiscoPress)
• Send a tweet and include
– Your favorite speaker’s Twitter handle <@hussainshoaib>
– Two hashtags: #CLUS #MyFavoriteSpeaker
• You can submit an entry for more than one of your “favorite” speakers
• Don’t forget to follow @CiscoLive and @CiscoPress
• View the official rules at http://bit.ly/CLUSwin
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
Cisco Public
130
Complete Your Online Session Evaluation
• Give us your feedback and you
could win fabulous prizes. Winners
announced daily.
• Complete your session evaluation
through the Cisco Live mobile app
or visit one of the interactive kiosks
located throughout the convention
center.
Don’t forget: Cisco Live sessions will be available
for viewing on-demand after the event at
CiscoLive.com/Online
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
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131
Continue Your Education
• Demos in the Cisco Campus
• Walk-in Self-Paced Labs
• Table Topics
• Meet the Engineer 1:1 meetings
BRKUCC-2934
© 2014 Cisco and/or its affiliates. All rights reserved.
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132