Configuring the Sonus SBC 2000 with Cisco Unified Call Manager 10.5 for Verizon Deployment

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1 Configuring the Sonus SBC 2000 with Cisco Unified Call Manager 10.5 for Verizon Deployment Application Notes Rev 1.0 P/N Last Updated: October 26, 2015 Revision History Revision Date Revised By Comments /26/2015 Technical Publications Final Document

2 Contents 1 Document Overview Audience Requirements Reference Configuration Interworking Architecture Cisco Functionality Cisco Components... 6 Cisco Unified Communication Manager Configuring Sonus SBC 2000 Series External peer side SBC configuration... 8 Node Interface... 8 Node Ports... 9 Logical Interface SIP Profile Media Profile Media List SIP Server Tables Signaling Group Calling Routing Table Transformation Table Message Manipulation Internal Peer Side Configuration Node Interface Node Ports Logical Interface SIP Profile Media Profile Media List SIP Server Tables Signaling Group Calling Routing Table Transformation Table Cisco Unified Communication Manager(CUCM) configuration of 40

3 4.1 CUCM 10.5 Configuration Settings Login to CUCM SIP Trunk Security Profile SIP Profile Create a New Trunk Create a New Route Group Create a New Route List Create a New Route Pattern of 40

4 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 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: (Toll-free) or (Direct) Web: Requirements The following equipment and software was used for the sample configuration provided: Sonus Equipment Type Version SBC 2000 SBC Build 371 Tenor AFM200 Analog VoIP Gateway P of 40

5 3rd Party Equipment Type Version Cisco Unified Call Manager IP-PBX Cisco IP Phone 7942 SIP Phone of 40

6 1.3 Reference Configuration A simulated enterprise site consists of the following elements: CUCM 10.5 and an SBC 2000 system running software version Verizon SIP trunks were used to connect the SBC to the Cisco UCM. Cisco 10.5 Sonus SBC 2000 Internal IP Network Verizon 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 6 of 40

7 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: 7 of 40

8 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: Internal/Private Signaling Group: From/To CM10 Call Routing: From CM10 External/Public Signaling Group: From/To Verizon Call Routing: From VZ CUCM : :5060 CUCM Signaling Group SIP over TCP SIP over UDP Verizon :5060 Signaling Group: From/To CM10 Call Routing: From CM10 SIP over UDP :5070 Tenor :5060 Signaling Group: From/To Tenor-FAX Call Routing: From Tenor-FAX FXS to 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. 8 of 40

9 Node Ports Figure 3.2 External (public) Network Node Port 9 of 40

10 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. 10 of 40

11 Figure 3.4 SIP Profile 11 of 40

12 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 Voice Codec G.729 Figure 3.6 Voice Codec G.711u Figure 3.7 Fax Codec 12 of 40

13 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. 13 of 40

14 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. 14 of 40

15 Figure 3.10 Signaling Group 15 of 40

16 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 16 of 40

17 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 17 of 40

18 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 18 of 40

19 Figure 3.16 Message Manipulation Port Remove 19 of 40

20 Figure 3.17 Message Manipulation IP Address Change 20 of 40

21 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 21 of 40

22 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. 22 of 40

23 Figure 3.20 SIP Profile 23 of 40

24 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 24 of 40

25 Media List The Media List shows the selected voice and fax compression codecs and their associated settings. Figure 3.23 Media List 25 of 40

26 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 26 of 40

27 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 27 of 40

28 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 28 of 40

29 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 29 of 40

30 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: 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. Trunk Group (TG) 2. Route Group (RG) 3. Route List (RL) 4. 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. 30 of 40

31 3. Select the Transport Type. This certification utilized UDP between Cisco and the Sonus SBC. Figure 4.2 SIP Trunk Security Profile 31 of 40

32 SIP Profile From the menu bar, select Device > Device Settings > SIP Profile. Figure 4.3 SIP Profile 32 of 40

33 Figure 4.4 SIP Profile Second Part 33 of 40

34 Figure 4.5 SIP Profile Third Part 34 of 40

35 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 New SIP Trunk Figure 4.7 SIP Trunk Configuration - Second Part 35 of 40

36 Figure 4.8 SIP Trunk Configuration Third Part Figure 4.9 SIP Trunk Configuration Fourth Part 36 of 40

37 From the menu bar: Create a New Route Group 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 37 of 40

38 From the menu bar: Create a New Route List 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 38 of 40

39 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 39 of 40

40 Figure 4.13 New Route Pattern Second Part 40 of 40

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