Configuring the Sonus SBC 1000/2000 series with LYNC 2013 for TELUS Deployment over Public Internet
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1 Configuring the Sonus SBC 1000/2000 series with LYNC 2013 for TELUS Deployment over Public Internet Application Notes V 1.0 Last Updated: January 14, Sonus Networks, Inc. All Rights Reserved
2 Contents 1 Document Overview Audience Requirements Reference configuration Tested Features Configuring Sonus SBC 1000 Series External Side Configuration... 7 Node Interface... 7 Node Ports... 8 Logical Interface... 8 SIP Profile... 9 Media Profile Media list Remote Authorization Table Contact Registrant Table SIP Server Tables Signaling Group Calling Routing Table Transformation Table Message Manipulation Internal Side Configuration Node Interface Node Ports Logical Interface SIP Profile Media Profile Media List SIP Server Tables Signaling Group Call Routing Table Transformation Table Internal Side Fax Configuration Node Interface Sonus Networks 2
3 Node Ports Logical Interface SIP Profile Media Profile Media list SIP Server Tables Signaling Group Call Routing Table Transformation Table Microsoft Lync 2013 Configuration Lync 2013 configuration settings Addition of the SBC to the Lync Server Lync Routing to the SBC Lync Specific Configurations Call Transfer Sonus Networks 3
4 1 Document Overview This document describes the steps required to configure the Sonus Session Border Controller (SBC) 1000/2000 series when connecting the TELUS SIP trunk over public internet and Microsoft Lync 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 TELUS SIP trunk over public internet and the Lync Audience This technical document is intended for telecommunication engineers with the purpose of configuring the Sonus SBC 1000/2000 series and aspects of the SIP trunk group together with the Lync 2013 product. There will be steps that require navigating a third-party and Sonus SBC Web browser user interface (WebUI). Understanding the basic concepts of IP/Routing and SIP/RTP are 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 for the SBC 1000/2000 series can be obtained through the following: Phone: (Toll-free) or (Direct) Web: 4
5 1.2 Requirements The following equipment and software was used for the sample configuration provided: Sonus Equipment Type Version SBC 1000 SBC 1000/2000 series Build 388 Tenor AFM200 Analog VoIP Gateway P VentaFax Fax Software I 3rd Party Equipment Type Version Lync 2013 IP-PBX Polycom CX500 phone SIP Phone TELUS SIP Trunking Service Release 2 Platform Components TELUS Equipment Version Oracle AP6300 Session Border Controller Genband EXPERiUS Application Server Genband C20 Call Session Controller MR 3 P 2 MCP CVM17 5
6 1.3 Reference configuration The simulated enterprise site consists of Microsoft Lync 2013 and an SBC 1000 system running software version The TELUS SIP trunk over public internet was used to connect the SBC 1000 to the Lync Lync 2013 Sonus SBC 1000 Internal IP Network Telus Figure 1.1 Network Diagram 1.4 Tested Features The testing was executed with the TELUS test plan, and the following features were tested for PSTN, BVOIP and Mobile clients: Basic originated and terminated calls Basic inbound/outbound call Basic inbound/outbound call with privacy Hold and resume Call Transfer (Blind transfer) Call Transfer (Consult transfer) Call Forwarding Unconditional Call Forwarding Busy Call Forwarding Don t Answer Voic Conference call Long calls FAX DTMF 6
7 2 Configuring Sonus SBC 1000 Series In this section, all settings used in the call testing are shown in the web browser user interface (WebUI). For more detailed information on the parameters and the WebUI, please refer to the Administration and Configuration guides for the SBC 1000/2000 series at the following link: Internal/Private Signaling Group: From/To Lync 2013 Call Routing: From Lync to Telus External/Public Signaling Group: From/To Telus Call Routing: From Telus Lync : :5060 fe.lync2013.sonusnet.com:5068 SIP over TCP SIP over UDP SIP over UDP Public Internet Telus :5060 Tenor :5060 Signaling Group: Telus Fax Call Routing: From Lync to Telus FXS to FAX Fax Figure 2.1 SBC 1000 SIP Trunk Diagram 2.1 External Side Configuration Node Interface The SBC 1000/2000 series 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. The following sections are the settings for the Ethernet connections (SIP signaling/rtp) between the SBC 1000 and the SIP Trunk to TELUS over public internet. 7
8 Node Ports Figure 2.2 External Network Node Port Logical Interface Figure 2.3 External Logical Interface 8
9 SIP Profile SIP Profiles control how the SBC 1000/2000 series communicates with SIP devices. The SIP Profiles control important characteristics such as session timers, SIP Header customization, SIP timers, MIME payloads, and option tags. The following figure shows the SIP Profile used for the SBC 1000 for this testing effort. Figure 2.4 SIP Profile 9
10 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. The following figures show the media profiles of the voice codecs used for the SBC 1000 in this testing effort and are for reference only. Figure 2.5 Voice codec G.729 Figure 2.6 Voice codec G.711u Figure 2.7 Voice codec G.711a Figure 2.8 Fax Codec 10
11 Media list The Media List in the following figure shows the selected voice and fax compression codecs and their associated settings. Figure 2.9 Media list 11
12 Remote Authorization Table Figure 2.10 Remote Authorization Table Contact Registrant Table Figure 2.11 Contact Registrant table 12
13 SIP Server Tables SIP Server Tables contain information about the SIP devices connected to the SBC 1000/2000 series. The table entries in the following figure 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 2.12 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. This is also the location where 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. 13
14 Figure 2.13 Signaling Group 14
15 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 to determine 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 2.14 Call Routing Table 15
16 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 2.15 Transformation Table Message Manipulation Message manipulation can change/add/delete some parameters in SIP headers and SDP body that are necessary for correct interoperability between SBC and other platforms. Condition Rule Table Match DID Range Match assigned TELUS DID range. Figure 2.16 Condition Rule Table 16
17 Message Rule Tables Delete PAID This message rule deletes PAID if previous condition is met. Figure 2.17 Message Manipulation Delete PAID Message Rule Tables Create PAID This message rule creates PAID if previous condition is met. Figure 2.18 Message Manipulation Create PAID 17
18 Message Rule Tables Delete Diversion This message rule deletes Diversion Header. Figure 2.19 Message Manipulation Delete Diversion Message Rule Tables Delete Referred-by This message rule deletes Referred-by header. Figure 2.20 Message Manipulation Delete Referred-by 18
19 2.2 Internal Side Configuration Node Interface The SBC 1000/2000 series 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. The following sections show the settings for the Ethernet connections (SIP signaling/rtp) between the SBC 1000 and the Lync 2013 SIP trunk. Node Ports Figure 2.21 Internal Network Node Port 19
20 Logical Interface Figure 2.22 Internal Logical Interface 20
21 SIP Profile SIP Profiles control how the Sonus SBC 1000/2000 series communicates with SIP devices. These control important characteristics such as session timers, SIP Header customization, SIP timers, MIME payloads, and option tags. The following figure shows the default SIP profile used for the SBC 1000 for this testing effort. Figure 2.23 SIP Profile 21
22 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. The following figures show the media profiles of the voice codecs used for the SBC 1000 in this testing effort and are for reference only. Figure 2.24 Voice Codec G.711a Figure 2.25 Voice Codec G.711u 22
23 Media List The Media List in the following figure shows the selected voice and fax compression codecs and their associated settings. Figure 2.26 Media List 23
24 SIP Server Tables SIP Server Tables contain information about the SIP devices connected to the Sonus SBC 1000/2000 series. The table entries in the following figure 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 2.27 SIP Server Table 24
25 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. This is also the location where 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 2.28 Signaling Group 25
26 Call 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 to determine which calls will be carried and also how the calls are translated. These tables are one of the central connection points of the system Figure 2.29 Call Routing Table 26
27 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 is sequentially selected from there. In addition, Transformation Tables are configurable as a reusable pool that Action Sets can reference. Transformation Table Remove + From Redirecting Number Figure 2.30 Transformation Table Remove + From Redirecting Number Transformation Table Match All Redirecting Numbers Figure 2.31 Transformation Table Match All Redirecting Numbers 27
28 Transformation Table Copy Redirecting Number Into SG User Value 1 Figure 2.32 Transformation Table Copy Redirecting Number Into SG User Value 1 Transformation Table Match All Called numbers Figure 2.33 Transformation Table Match All Called numbers 28
29 Transformation Table Match All Called numbers Figure 2.34 Transformation Table Remove + From Calling Number Transformation Table Match All Calling Numbers Figure 2.35 Transformation Table Match All Calling Numbers 29
30 2.3 Internal Side Fax Configuration Node Interface The SBC 1000/2000 series 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. The following sections are the settings for the Ethernet connections (SIP signaling/rtp) between the SBC 1000 and the Fax SIP trunk. Node Ports Figure 2.36 Internal Network Node Port 30
31 Logical Interface Figure 2.37 Internal Logical Interface 31
32 SIP Profile SIP Profiles control how the Sonus SBC 1000/2000 series communicates with SIP devices. These control important characteristics such as session timers, SIP Header customization, SIP timers, MIME payloads, and option tags. The following figure shows the default SIP profile used for the SBC 2000 for this testing effort. Figure 2.38 SIP Profile 32
33 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. The following figures show the media profiles of the voice codecs used for the SBC 1000 in this testing effort and are for reference only. Figure 2.39 Voice Codec G.729 Figure 2.40 Voice Codec G.711u Figure 2.41 Voice Codec G.711a Figure 2.42 Fax Codec 33
34 Media list The Media List in the following figure shows the selected voice and fax compression codecs and their associated settings. Figure 2.43 Media List 34
35 SIP Server Tables SIP Server Tables contain information about the SIP devices connected to the SBC 1000/2000 series. The table entries in the following figure 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 2.44 SIP Server Table 35
36 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. This is also the location where 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 2.45 Signaling Group 36
37 Call 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. Figure 2.46 Call Routing Table 37
38 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. Transformation Table Remove + From Redirecting Number Figure 2.47 Transformation Table Remove + From Redirecting Number Transformation Table Match All Redirecting Numbers Figure 2.48 Transformation Table Match All Redirecting Numbers 38
39 Transformation Table Remove + From Redirecting Number Figure 2.49 Transformation Table Copy Redirecting Number Into SG User Value 1 Transformation Table Match All Called Numbers Figure 2.50 Transformation Table Match All Called numbers Transformation Table Remove +1 From Calling Number Figure 2.51 Transformation Table Remove + From Calling Number 39
40 Transformation Table Match All Calling Numbers Figure 2.52 Transformation Table Match All Calling Numbers 40
41 3 Microsoft Lync 2013 Configuration 3.1 Lync 2013 configuration settings This section assumes that the Lync Server components have been installed along with Lync users. The user should be familiar with Lync Server Topology Builder, Lync Server Control Panel and Lync Server management Shell. This section does not cover the basic installation of Lync Server Addition of the SBC to the Lync Server Login to the Administration Portal of the Communication Manager. The Lync Server topology needs to be modified by adding the SBC as a Gateway device. The Gateway device is the interface to the TELUS SIP Trunk. 1. Open Lync Server Topology builder. 2. Load the current topology. 3. Expand the topology. 4. Right click the PSTN Gateways link in the left hand pane. 5. Select New IP/PSTN Gateway from the menu as shown and follow thru with the process. 41
42 Figure 3.1 Create New Gateway 42
43 Figure 3.2 Define the FQDN of the Gateway 43
44 Figure 3.3 Enable IPV4/IPV6 44
45 Figure 3.4 Define Trunk Port and Protocol 45
46 Lync Routing to the SBC Figure 3.5 Add Routing 46
47 3.2 Lync Specific Configurations Call Transfer Call Transfer via REFER Method Microsoft Lync Server 2013 needs additional configuration in order to enable SIP REFER Method. Refer support needs to be set to Enable sending refer to gateway under Trunk Configuration profile assigned to the appropriate SBC trunk. Figure 3.6 Refer Support Call Transfer via Re-Invite Method Transferring a call to another phone number is supported via the RFC3261 method. No special flag is required to be set for this method. Under the Trunk Configuration profile on Lync 2013, ensure that the Refer support is set to none. 47
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