Configuring the Sonus SBC 2000 with
Cisco Unified Call Manager 10.5 for Verizon
Deployment
Application Notes Rev 1.0
P/N 550 - 06690
Last Updated: October 26, 2015
Revision History
Revision
Date
Revised By
Comments
1.0
10/26/2015
Technical Publications
Final Document
Copyright © 2015, Sonus and/or its affiliates. All rights reserved.
Contents
1 Document Overview .............................................................................. 4
1.1
Audience ................................................................................................................................. 4
1.2
Requirements .......................................................................................................................... 4
1.3
Reference Configuration ......................................................................................................... 6
2 Interworking Architecture ....................................................................... 6
2.1
Cisco Functionality .................................................................................................................. 6
2.2
Cisco Components .................................................................................................................. 6
Cisco Unified Communication Manager .............................................................................. 6
3 Configuring Sonus SBC 2000 Series ..................................................... 8
3.1
External peer side SBC configuration ..................................................................................... 8
Node Interface ..................................................................................................................... 8
Node Ports .......................................................................................................................... 9
Logical Interface ................................................................................................................ 10
SIP Profile ......................................................................................................................... 10
Media Profile ..................................................................................................................... 12
Media List .......................................................................................................................... 13
SIP Server Tables ............................................................................................................. 13
Signaling Group ................................................................................................................. 14
Calling Routing Table ........................................................................................................ 16
Transformation Table ........................................................................................................ 17
Message Manipulation ...................................................................................................... 18
3.2
Internal Peer Side Configuration ........................................................................................... 21
Node Interface ................................................................................................................... 21
Node Ports ........................................................................................................................ 21
Logical Interface ................................................................................................................ 22
SIP Profile ......................................................................................................................... 22
Media Profile ..................................................................................................................... 24
Media List .......................................................................................................................... 25
SIP Server Tables ............................................................................................................. 26
Signaling Group ................................................................................................................. 27
Calling Routing Table ........................................................................................................ 28
Transformation Table ........................................................................................................ 29
4 Cisco Unified Communication Manager(CUCM) configuration ............ 30
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4.1
CUCM 10.5 Configuration Settings ....................................................................................... 30
Login to CUCM .................................................................................................................. 30
SIP Trunk Security Profile ................................................................................................. 30
SIP Profile ......................................................................................................................... 32
Create a New Trunk .......................................................................................................... 35
Create a New Route Group ............................................................................................... 37
Create a New Route List ................................................................................................... 38
Create a New Route Pattern ............................................................................................. 39
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1 Document Overview
This document provides a configuration guide for Sonus SBC 1000/2000 Series (Session Border Controller)
when connecting Verizon SIP trunk and the Cisco Unified Call Manger 10.5 (CUCM 10.5).
The Sonus SBC 1000 and SBC 2000 are Session Border Controllers that connect disparate SIP trunks, SIP
PBXs, and communication applications within an enterprise. The SBC can also be used as a SIP routing and
integration engine.
The Sonus SBC is the point of connection between the Verizon SIP trunk and the CUCM 10.5.
1.1 Audience
This technical document is intended for telecommunication engineers with the purpose of configuring the Sonus
SBC 1000 and SBC 2000 and aspects of the SIP trunk group together with the CUCM 10.5 product. There will
be steps that require navigating the third-party and Sonus SBC Web browser user interface or WebUI.
Understanding the basic concepts of IP/Routing and SIP/RTP is also necessary to complete the configuration
and for troubleshooting, if necessary.
This configuration guide is offered as a convenience to Sonus customers. The specifications and information
regarding the product in this guide are subject to change without notice. All statements, information, and
recommendations in this guide are believed to be accurate but are presented without warranty of any kind,
express or implied, and are provided “AS IS”. Users must take full responsibility for the application of the
specifications and information in this guide.
Technical support on SBC 1000 and SBC 2000 can be obtained through the following:


Phone: +1 888-391-3434 (Toll-free) or +1 978-614-8589 (Direct)
Web: http://www.sonus.net/company/maintenance/log-trouble-tickets
1.2 Requirements
The following equipment and software was used for the sample configuration provided:
Sonus Equipment
Type
Version
SBC 2000
SBC 2000
4.1.1 Build 371
Tenor AFM200
Analog VoIP Gateway
P108-09-26
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3rd Party Equipment
Type
Version
Cisco Unified Call
Manager
IP-PBX
10.5.1.11005-2
Cisco IP Phone 7942
SIP Phone
9.3.1.57
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1.3 Reference Configuration
A simulated enterprise site consists of the following elements: CUCM 10.5 and an SBC 2000 system running
software version 4.1.1. Verizon SIP trunks were used to connect the SBC to the Cisco UCM.
Cisco 10.5
Sonus
SBC 2000
Verizon
Internal IP Network
2 Interworking Architecture
2.1 Cisco Functionality

General Cisco functionality provides the UCM application

All the SIP Phones will register with Cisco Unified Communication Manager
2.2 Cisco Components
Cisco Unified Communication Manager
Cisco Unified Communications Manager software is the call-processing component of the Cisco Unified
Communications system. Cisco Unified Communications Manager extends enterprise telephony features and
capabilities to packet telephony network devices such as IP phones, media processing devices, voice over IP
(VoIP) gateways, and multimedia applications. Additional services such as unified messaging, multimedia
conferencing, collaborative contact centers, and interactive multimedia response systems are made possible
through Cisco Unified Communications Manager open telephony APIs. Cisco Unified Communications Manager
offers a suite of integrated voice applications and utilities, including the Cisco Unified Communications Manager
Attendant Console, an ad-hoc conferencing application, the Cisco Unified Communications Manager Bulk
Administration Tool, the Cisco Unified Communications Manager CDR (call detail record) Analysis and Reporting
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Tool, the Cisco Unified Communications Manager Real-Time Monitoring Tool, and the Cisco Unified
Communications Manager Assistant application.
The dial plan feature in Unified Communications Manager enable you to:
•
Route calls based on the physical location context of the caller.
•
Represent calling and called party numbers in a global form such as that described by the International
Telecommunications Union's E.164 recommendation.
•
Present calls to users in a format based on local dialing habits.
•
Present calls to external networks (for example, the PSTN) in a manner compatible with the local
requirements for calling party number, called party number, and their respective numbering types.
•
Derive the global form of the calling party number on incoming calls from gateways, based on the calling
number digits and the numbering type.
For additional information, go to:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/tsd_products_support_general_information.html.
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3 Configuring Sonus SBC 2000 Series
In this section, all settings used in the call testing are shown as seen in the WebUI or web browser user
interface. For more detailed information on the parameters and the WebUI, please refer to the Administration
and Configuration guides for the SBC 1000 and SBC 2000 in the documentation pages at:
https://support.sonus.net/display/ALLDOC/Sonus+Product+Documentation
Internal/Private
External/Public
Signaling Group: From/To CM10
Call Routing: From CM10
Signaling Group: From/To Verizon
Call Routing: From VZ
CUCM 10.5
10.35.177.232:5060
216.110.2.231:5060
CUCM Signaling Group
Verizon
SIP over UDP
SIP over TCP
SIP over UDP
10.35.180.109:5060 Signaling Group: From/To CM10
Call Routing: From CM10
63.65.248.30:5070
Tenor
FXS to FAX
10.160.186.162:5060
Signaling Group: From/To Tenor-FAX
Call Routing: From Tenor-FAX
Fax
Figure 3.1
SBC 2000 SIP Trunk Diagram
3.1 External peer side SBC configuration
Node Interface
The Sonus SBC 1000/2000 WebUI allows the configuration of the Identification information, Physical Data Layer,
and Networking Layer for the ports. Use the associated Logical Interface to configure the IP address or other IP
parameters. Those settings can be modified using the Modify Ethernet IP task found under the Tasks tab.
Shown below are the settings for the Ethernet connections (SIP signaling/RTP) between the Sonus SBC 2000
and the Public internet (SIP Trunk) to Verizon.
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Node Ports
Figure 3.2
External (public) Network Node Port
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Logical Interface
Figure 3.3
External Logical Interface
SIP Profile
SIP Profiles control how the Sonus SBC 1000/2000 communicates with SIP devices. These control important
characteristics such as: session timers, SIP Header customization, SIP timers, MIME payloads, and option tags.
The default SIP profile used for the SBC 2000 for this testing effort is shown below.
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Figure 3.4
SIP Profile
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Media Profile
Media Profiles specify the individual voice and fax compression codecs and their associated settings for
inclusion into a Media List. Different codecs provide varying levels of compression, allowing the reduction in
bandwidth requirements at the expense of voice quality. Listed below are the media profiles of the voice codecs
used for the SBC 2000 in this testing effort and is for reference only.
Figure 3.5
Figure 3.6
Voice Codec G.729
Voice Codec G.711u
Figure 3.7
Fax Codec
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Media List
The Media List shows the selected voice and fax compression codecs and their associated settings.
Figure 3.8
Media List
SIP Server Tables
SIP Server Tables contain information about the SIP devices connected to the Sonus SBC 1000/2000. The
entries in the tables provide information about the IP addresses, ports, and protocols used to communicate with
each SIP server. The entries also contain links to counters that are useful for troubleshooting.
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Figure 3.9
SIP Server Table
Signaling Group
Signaling Groups allow telephony channels to be grouped together for the purposes of routing and shared
configuration. These groups are the entity to which calls are routed, as well as the location from which Call
Routes are selected. These are also the location from which Tone Tables and Action Sets are selected. In the
case of SIP, this will specify protocol settings and link to server, media and mapping tables.
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Figure 3.10 Signaling Group
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Calling Routing Table
Call Routing allows calls to be carried between signaling groups, thus allowing calls to be carried between ports,
and between protocols (like ISDN to SIP). Routes are defined by Call Routing Tables, which allow for flexible
configuration of which calls will be carried, and also how the calls are translated. These tables are one of the
central connection points of the system, linking Transformation Tables, Message translations, Cause Code
Reroutes, Media Lists and the three types of Signaling Groups (ISDN, SIP and CAS).
Figure 3.11 Call Routing Table
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Transformation Table
Transformation Tables facilitate the conversion of names, numbers and other fields when routing a call. For
example, transformations can convert a public PSTN number into a private extension number, or into a SIP
address (URI). Every entry in a Call Routing Table requires a Transformation Table, and are sequentially
selected from there. In addition, Transformation tables will be configurable as a reusable pool that Action
Sets can reference.
Figure 3.12 Transformation Table – Match And Change CLI
Figure 3.13 Transformation Table – Match And Change Redirecting Number
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Figure 3.14 Transformation Table – Pass-Through
Message Manipulation
Message manipulation allows to change/add/delete some parameters in SIP header and SDP body which are
necessarily for correct interoperability between SBC and other platforms.
Figure 3.15 Message Manipulation – PAI Remove
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Figure 3.16 Message Manipulation – Port Remove
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Figure 3.17 Message Manipulation – IP Address Change
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3.2 Internal Peer Side Configuration
Node Interface
The Sonus SBC 1000/2000 WebUI allows the configuration of the Identification information, Physical Data Layer,
and Networking Layer for the ports. Use the associated Logical Interface to configure the IP address or other IP
parameter. Those settings can be modified using the Modify Ethernet IP task found under the Tasks tab. Shown
below are the settings for the Ethernet connections (SIP signaling/RTP) between the Sonus SBC 2000 and the
CUCM SIP trunk.
Node Ports
Figure 3.18
Internal (Private) Network Node Port
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Logical Interface
Figure 3.19
Internal Logical Interface
SIP Profile
SIP Profiles control how the Sonus SBC 1000/2000 communicates with SIP devices. These control important
characteristics such as: session timers, SIP Header customization, SIP timers, MIME payloads, and option tags.
The default SIP profile used for the SBC 2000 for this testing effort is shown below.
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Figure 3.20 SIP Profile
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Media Profile
Media Profiles specify the individual voice and fax compression codecs and their associated settings for
inclusion into a Media List. Different codecs provide varying levels of compression, allowing one to reduce
bandwidth requirements at the expense of voice quality. Listed below are the media profiles of the voice codecs
used for the SBC 2000 in this testing effort and is for reference only.
Figure 3.21 Voice Codec G.729
Figure 3.22 Voice Codec G.711u
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Media List
The Media List shows the selected voice and fax compression codecs and their associated settings.
Figure 3.23 Media List
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SIP Server Tables
SIP Server Tables contain information about the SIP devices connected to the Sonus SBC 1000/2000. The
entries in the tables provide information about the IP addresses, ports, and protocols used to communicate with
each SIP server. The entries also contain links to counters that are useful for troubleshooting.
Figure 3.24 SIP Server Table
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Signaling Group
Signaling Groups allow telephony channels to be grouped together for the purposes of routing and shared
configuration. These groups are the entity to which calls are routed, as well as the location from which Call
Routes are selected. These are also the location from which Tone Tables and Action Sets are selected. In the
case of SIP, this will specify protocol settings and link to server, media and mapping tables.
Figure 3.25 Signaling Group
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Calling Routing Table
Call Routing allows calls to be carried between signaling groups, thus allowing calls to be carried between ports,
and between protocols (like ISDN to SIP). Routes are defined by Call Routing Tables, which allow for flexible
configuration of which calls will be carried, and also how the calls are translated. These tables are one of the
central connection points of the system, linking Transformation Tables, Message translations, Cause Code
Reroute, Tables, Media Lists and the three types of Signaling Groups (ISDN, SIP and CAS).
Figure 3.26 Call Routing Table
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Transformation Table
Transformation Tables facilitate the conversion of names, numbers and other fields when routing a call. For
example, transformations can convert a public PSTN number into a private extension number, or into a SIP
address (URI). Every entry in a Call Routing Table requires a Transformation Table, and are sequentially
selected from there. In addition, Transformation tables will be configurable as a reusable pool that Action
Sets can reference.
Figure 3.27 Transformation Table – Match CLD
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4 Cisco Unified Communication Manager(CUCM)
configuration
4.1 CUCM 10.5 Configuration Settings
The CUCM 10.5 was configured per the details provided in the Cisco Configuration and Administration Guide.
This guide is available online at the following location:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/im_presence/configAdminGuide/10_5_2/CUP0_BK_
CEB3E82E_00_config-admin-guide-imp-1052.html
In order to connect Cisco CUCM 10.5 PBX to the SBC 2000, the following objects must be created and properly
associated.
1.
2.
3.
4.
Trunk Group (TG)
Route Group (RG)
Route List (RL)
Route Pattern (RP)
Login to CUCM
Login to the Administration Portal of the Communication Manager, type in valid credentials and login.
Figure 4.1
CUCM Administration Page
SIP Trunk Security Profile
From the menu bar:
1. Select System > Security Profile > SIP Trunk Security Profile.
2. Select the Device Security Mode.
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3. Select the Transport Type.
This certification utilized UDP between Cisco and the Sonus SBC.
Figure 4.2
SIP Trunk Security Profile
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SIP Profile
From the menu bar, select Device > Device Settings > SIP Profile.
Figure 4.3
SIP Profile
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Figure 4.4
SIP Profile – Second Part
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Figure 4.5
SIP Profile – Third Part
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Create a New Trunk
From the menu bar:
1. Select Device > Trunk.
2. Click the “Add New” button.
3. Select a Trunk Type.
4. Select a Device Protocol.
5. Click Next.
Figure 4.6
Figure 4.7
New SIP Trunk
SIP Trunk Configuration - Second Part
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Figure 4.8
Figure 4.9
SIP Trunk Configuration – Third Part
SIP Trunk Configuration – Fourth Part
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Create a New Route Group
From the menu bar:
1. Select Call Routing > Route-Hunt > Route Group.
2. Click Add New.
(Note that the TG must already be added and will be displayed as an Available Devices under the Find
Devices to Add to Route Group area.)
3. Select the appropriate TG.
4. Click Add to Route.
5. Enter a Route Group Name (refer to CUCM 10.5 guide for more detail).
Figure 4.10 New Route Group
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Create a New Route List
From the menu bar:
1. Select Call Routing > Route-Hunt > Route List.
2. Click Add New.
3. Enter the Name.
4. Enter a Description.
5. Click Add Route Group.
6. On the next screen, select the Route Group that was just created (refer to CUCM 10.5 guide for more
detail).
Figure 4.11 New Route List
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Create a New Route Pattern
From the menu bar:
1. Select Call Routing > Route-Hunt > Route Pattern.
2. Click Add New.
3. Enter the Route Pattern.
4. Click Save.
Figure 4.12 New Route Pattern
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Figure 4.13 New Route Pattern – Second Part
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