Abstract. These Application Notes provide information for the setup, configuration, and verification of the call flows tested on this solution.

Transcription

1 Avaya Solution & Interoperability Test Lab Application Notes for Radvision Scopia XT 5000 Series Endpoint with Multi Avaya Aura Communication Manager R6.2 FP1 and Multi Avaya Aura Session Manager R6.2 FP1 Integration Issue 1.0 Abstract These Application Notes describe the configuration steps required to integrate Radvision Scopia XT 5000 Series endpoints with a Multi Avaya Aura Communication Manager and Avaya Aura Session Manager infrastructure. In the interoperability testing, calls were made between Avaya Aura 6.2 FP1 infrastructure and Radvision Scopia XT 5000 Series Endpoint registered to Scopia Management Enhanced Communication Server (ECS) Gatekeeper. These Application Notes provide information for the setup, configuration, and verification of the call flows tested on this solution. 1 of 54

2 Table of Contents 1. Introduction Interoperability Testing Test Description and Coverage Test Results and Observations Reference Configuration Equipment and Software Validated Configure Avaya Aura Communication Manager Verify License Verify Off-PBX Telephones Capacities Verify SIP Trunk Capacity Verify AAR Access Code is Configured Verify Private Networking is Enabled Configure System Parameters Configure Trunk-to-Trunk Transfers Administer CPN/ANI/ICLID PARAMETERS Administer Dialplan Administer Uniform Dialplan Administer Private Numbering Administer Public-Unknown-Numbering Administer AAR Administer Audio Codec Administer Network Region Administer Node Names Administer Signaling Groups Administer Signaling Group for H.323 Trunk Administer Signaling Group for SIP Trunk Administer Trunk Groups Administer H.323 Trunk Group Administer SIP Trunk Administer Route Patterns Administer H.323 Route Pattern Administer SIP Route Pattern Administer Incoming Call Handling Treatment Save Translations Configure Avaya Aura Session Manager Define SIP Domains Define Locations Define SIP Entities of 54

3 Add Session Manager SIP Entity Add Communication Manager SIP Entity Define Entity Links Define Routing Policy Define Dial Patterns Synchronize Changes with Avaya Aura Communication Manager Configure Scopia Management Add Location Add Communication Manager as Gatekeeper Add ECS Gatekeeper Add Radvision Scopia XT 5000 Endpoint Verification Steps Verify Avaya Aura Session Manager Configuration Verify Radvision Scopia Configuration Verify Radvision Scopia XT 5000 Endpoint Registration Verify Communication Manager Configuration Conclusion Additional References Avaya Aura Communication Manager Release 6.2 and Radvision SCOPIA Release 7.7 and 8.0 Interoperability Day 180 Solution Quick Setup, Issue Scopia Solution Guide Version of 54

4 1. Introduction These Application Notes describe the configuration steps required to integrate Radvision Scopia XT 5000 Series endpoint within a multi Avaya Aura Communication Manager and Avaya Aura Session Manager infrastructure. The Radvision Scopia XT Series endpoints incorporate state-of-the-art video technology for high definition (HD) conferencing. The Radvision Scopia XT 5000 Series endpoint provides HD 1080p video at 60 frames per second (fps) capabilities for video conferencing. The XT 5000 also has the capability to locally host HD continuous presence videoconferences with its built-in MCU however this is out of scope for this document. Scopia Management delivers web-based management control and scheduling for video conferencing deployments. Scopia Management provides an all-inclusive solution for managing video conferencing, infrastructure and call control applications such as gatekeepers and SIP agents. Scopia Management can be deployed within a distributed environment to provide scalability and redundancy. Scopia Management also includes an internal ECS H.323 Gatekeeper. The Scopia ECS H.323 gatekeeper delivers comprehensive, intelligent backbone management of multimedia communication networks. ECS provides full H.323 Gatekeeper functionality. 2. Interoperability Testing The following sections describe the test scenarios that were used to verify the functionality of point-to-point audio and video calls with multiple Avaya Aura endpoints and Radvision Scopia XT 5000 Series endpoints across Multi Avaya Aura Communication Manager and Avaya Aura Session Manager servers. The bi-directional audio and video calls include basic Make Call, Release Call, Answer Call, Call Hold and Retrieve, Calls with Long Duration and the promotion of Audio Calls to Video Calls. In addition, various features including IP Shuffling, Audio Codec Negotiation and Video Codec Negotiation were tested Test Description and Coverage This section provides an overview of the test cases performed after the installation and configuration of Radvision Scopia XT 5000 Series endpoint within a Multi Avaya Aura Communication Manger and Multi Avaya Aura Session Manager Infrastructure. Calls: Place and Answer Audio/Video calls between Radvision Scopia XT 5000 Series endpoints and Avaya Flare Experience for Windows. Additional testing was completed with the following Avaya SIP endpoints: 96x1 Avaya Desktop Phone Series, Avaya Flare Experience for ipad, Avaya Desktop Video Device (ADVD) and one-x Communicator (Computer mode) however this Application Note focuses on Avaya Flare Experience for Windows. o Make Call 4 of 54

5 o Release Call o Answer Call o Hold/Un-Hold Call o Video Start/Stop o Audio/Video Mute o Long Duration o Promote Audio Call to Video Call 2.2. Test Results and Observations Overall, testing of the sample configuration was completed with the following observations: Radvision Scopia XT Series Endpoints (XT 5000, XT 4200 & XT 1200) do not support G.729 audio codec. Currently, G.711, G.722, G.722.1C, G.719, AAC-LD audio codecs are supported. Scopia Management does not strip the routing prefix administered with Scopia Management when routing call to Avaya Aura Communication Manager. The inccall-handling-trmt form on Communication Manager is used to strip the leading prefix in order to ensure proper routing of calls to Avaya Aura endpoints. When Flare Experience on Windows originates a video call to the Radvision XT5000 the Flare Experience s video is muted. The Flare Experience needs to un-mute the video in order to get two-way video. Interconnectivity between Avaya Aura core and Radvision Scopia solution consists of an H.323 trunk between Avaya Aura Communication Manager and Radvision Scopia Management internal ECS Gatekeeper. Point-to-point calls between Avaya Aura H.323 and/or Avaya Aura SIP endpoints and Radvision Scopia endpoints connect through the H.323 trunk. 5 of 54

6 3. Reference Configuration The diagram below Figure 1 shows the tested configuration. The sample configuration includes two separate Sites, SIL Site A and SIL Site B. SIL Site A includes Avaya Aura Communication Manager Duplex Server running as a Evolution Server installed on Avaya Aura System Platform S8800 server with a G650 Media Gateway. SIL Site A also has Avaya Aura System Manager installed on Avaya Aura System Platform S8800 server along with a single Avaya Aura Session Manager on S8800 server. Scopia Management is installed on a Microsoft Windows 2008 Enterprise Server R2 Operating System as a single server deployment. Scopia Management server also contains the internal Radvision ECS Gatekeeper. Radvision Scopia XT 5000 endpoint at this location is registered with the Radvision ECS Gatekeeper. SIL Site B deployment includes Avaya Aura Communication Manager Simplex Server running as a Evolution Server installed on Avaya Aura System Platform S8800 server with a G450 Media Gateway. SIL Site B also includes a single Avaya Aura Session Manager on S8800 server. The Communication Manager and Session Manager at this location are managed by the SIL Site A Avaya Aura System Manager. Note: IP addresses have been partially hidden for security. Figure 1: Compliance Testing Configuration 6 of 54

7 3.1. Equipment and Software Validated The following equipment and software were used for the sample configuration provided: Equipment/Software Release/Version Avaya Aura Session Manager on Avaya Release 6.2 FP S8800 server Avaya Aura System Manager on System Release 6.2 FP Platform S8800 server System Platform Avaya Aura Communication Manager Release R016x Evolution Server on System Platform S8800 System Platform server Scopia Management Release Scopia ECS Release Radvision Scopia XT5000 (H.323) Release Avaya Flare Experience on Windows (SIP) Release Build 85 7 of 54

8 4. Configure Avaya Aura Communication Manager This section provides the procedures for configuring Communication Manager. The procedures include the following areas: Verify License Configure System Parameters Administer Dial Plan Administer Uniform Dial Plan Administer Private Numbering Administer Public Numbering Administer AAR (Auto Alternate Routing) Administer Codecs Administer Network Regions Administer Node Names Administer Signaling Groups Administer Trunk Groups Administer Route Patterns Administer Incoming Call Treatment NOTE: the following procedures must be performed on both Communication Manager servers at both sites Verify License This section describes the procedures to verify the correct system capacities and licensing have been configured. If there is insufficient capacity or a required feature is not available, contact an authorized Avaya sales representative to make the appropriate changes. 8 of 54

9 Verify Off-PBX Telephones Capacities On Page 1 of the display system-parameters customer-options command, verify an adequate number of Off-PBX Stations (OPS) Telephones are administered for the system as shown below. display system-parameters customer-options Page 1 of 11 OPTIONAL FEATURES G3 Version: V16 Software Package: Enterprise Location: 2 System ID (SID): 1 Platform: 28 Module ID (MID): 1 USED Platform Maximum Ports: Maximum Stations: Maximum XMOBILE Stations: Maximum Off-PBX Telephones - EC500: Maximum Off-PBX Telephones - OPS: Maximum Off-PBX Telephones - PBFMC: Maximum Survivable Processors: Verify SIP Trunk Capacity On Page 2 of the display system-parameters customer-options command, verify an adequate number of SIP Trunk Members are administered for the system as shown below. display system-parameters customer-options Page 2 of 11 OPTIONAL FEATURES IP PORT CAPACITIES USED Maximum Administered H.323 Trunks: Maximum Concurrently Registered IP Stations: Maximum Administered Remote Office Trunks: Maximum Concurrently Registered Remote Office Stations: Maximum Concurrently Registered IP econs: Max Concur Registered Unauthenticated H.323 Stations: Maximum Video Capable Stations: Maximum Video Capable IP Softphones: Maximum Administered SIP Trunks: of 54

10 Verify AAR/ARS Routing is Enabled To simplify the dialing plan for calls between SIP endpoints and other types of stations, verify the following AAR/ARS features are enabled on the system. On Page 3 of system-parameters customer-options command, verify the following features are enabled. ARS? Verify y is displayed. ARS/AAR Partitioning? Verify y is displayed. ARS/AAR Dialing without FAC? Verify y is displayed. display system-parameters customer-options Page 3 of 11 OPTIONAL FEATURES Answer Supervision by Call Classifier? y Change COR by FAC? n ARS? y Computer Telephony Adjunct Links? y ARS/AAR Partitioning? y Cvg Of Calls Redirected Off-net? y ARS/AAR Dialing without FAC? y DCS (Basic)? y ASAI Link Core Capabilities? y DCS Call Coverage? y ASAI Link Plus Capabilities? y DCS with Rerouting? y Verify AAR Access Code is Configured To enable Communication Manager to route calls to SIP endpoints, verify an Automatic Alternative Routing (AAR) access code has been defined for the system. On Page 1 of change feature-access-codes command, verify a value has been defined in the Auto Alternate Routing (AAR) Access Code field. In the sample configuration, 8 was used. change feature-access-codes Page 1 of 11 FEATURE ACCESS CODE (FAC) Attendant Access Code: Auto Alternate Routing (AAR) Access Code: 8 Auto Route Selection (ARS) - Access Code 1: 9 Access Code 2: Automatic Callback Activation: *08 Deactivation: * Verify Private Networking is Enabled On Page 5 of display system-parameters customer-options command, verify the Private Networking feature is set to y. display system-parameters customer-options Page 5 of 11 OPTIONAL FEATURES Private Networking? y Processor and System MSP? y Processor Ethernet? y Uniform Dialing Plan? y Usage Allocation Enhancements? y Wideband Switching? y 10 of 54

11 4.2. Configure System Parameters Configure Trunk-to-Trunk Transfers Use the change system-parameters features command to enable trunk-to-trunk transfers. This feature is needed when an incoming call to a SIP station is transferred to another SIP station. For simplicity, the Trunk to Trunk Transfer field on Page 1 was to all to enable all trunk-to-trunk transfers on a system wide basis. Note: Enabling this feature poses significant security risk by increasing the risk of toll fraud, and must be used with caution. To minimize the risk, a COS could be defined to allow trunkto-trunk transfers for specific trunk group(s). For more information regarding how to configure Communication Manager to minimize toll fraud, see Reference [8] in Section 9. change system-parameters features Page 1 of 20 FEATURE-RELATED SYSTEM PARAMETERS Self Station Display Enabled? n Trunk-to-Trunk Transfer: all Automatic Callback with Called Party Queuing? n Automatic Callback - No Answer Timeout Interval (rings): Administer CPN/ANI/ICLID PARAMETERS Indicates whether to replace restricted numbers with administrable strings for incoming and outgoing calls assigned to the specified trunk group. If enabled, the display is replaced regardless of the service type of the trunk. Applies to BRI, PRI, H.323, and SIP trunks. Available only if the Group Type is isdn or sip. Use the change system-parameters features command to set the value for the CPN/ANI/ICLID Parameters. Set the CPN/ANI/ICLID Replacement for Restricted Calls and CPN/ANI/ICLID Replacement for Unavailable Calls parameters as desired. For the sample configuration, these parameters were set to SILFST CM4. change system-parameters features Page 9 of 20 FEATURE-RELATED SYSTEM PARAMETERS CPN/ANI/ICLID PARAMETERS CPN/ANI/ICLID Replacement for Restricted Calls: SILFST CMA CPN/ANI/ICLID Replacement for Unavailable Calls: SILFST CMA DISPLAY TEXT Identity When Bridging: principal User Guidance Display? n Extension only label for Team button on 96xx H.323 terminals? n INTERNATIONAL CALL ROUTING PARAMETERS Local Country Code: International Access Code: SCCAN PARAMETERS Enable Enbloc Dialing without ARS FAC? n CALLER ID ON CALL WAITING PARAMETERS Caller ID on Call Waiting Delay Timer (msec): of 54

12 4.3. Administer Dialplan This section describes the dial plan analysis screen for SIL SITE A Communication Manager. Use the Dial Plan analysis to provide the information that Communication Manager software uses to interpret the digits that a user dials. The user can determine the beginning digits and total length for each type of call that Communication Manager needs to interpret. The Dialed String beginning with the number 5204 and with a Total Length of 5 digits was used to administer the extension range used for the Scopia XT endpoints. Fill in the indicated fields as shown below and use default values for remaining fields. Dialed String Enter dialed string pattern of Scopia XT endpoints Total Length Enter extension range length Call Type Enter udp change dialplan analysis Page 1 of 12 DIAL PLAN ANALYSIS TABLE Location: all Percent Full: 5 Dialed Total Call Dialed Total Call Dialed Total Call String Length Type String Length Type String Length Type ext 72 2 aar 2 4 ext 76 7 udp 2 5 ext 77 8 udp 41 5 ext udp 42 5 ext 9 1 fac 43 5 ext * 3 fac udp # 4 dac 5 7 ext 4.4. Administer Uniform Dialplan Use the Uniform Dial Plan (UDP) feature to share a common dial plan among a group of Communication Servers. The UDP applies both interserver dialing and intraserver dialing. The UDP provides the dial plans from 3 to 18 digits. UDP provides extension-to-extension dialing between two or more private-switching systems. Communication Manager uses the UDP to route a call off the local server. The user dials an extension that consists of 3 to 18 digits. The software converts the extension that the user dials into a private network number or a public network number. Using the command change uniform-dialplan n where n is the first digit of the extension range that will be used by the Radvision Scopia XT endpoints. The dialed string beginning with the Matching Pattern 5204 and with a Total Length of 5 digits was used to administer the extension range used for the Radvision Scopia XT 5000 endpoints. Fill in the indicated fields as shown below and use default values for remaining fields. Matching Pattern Enter digit pattern of Radvision Scopia XT endpoints Len Enter extension length 12 of 54

13 Del Net Enter number of digits to delete if applicable Enter aar change uniform-dialplan 5 Page 1 of 2 UNIFORM DIAL PLAN TABLE Percent Full: 0 Matching Insert Node Pattern Len Del Digits Net Conv Num aar n aar n aar n aar n aar n 4.5. Administer Private Numbering For the sample configuration, private numbering was used and all Scopia XT endpoint extension numbers were unique within the private network. However, in many customer networks, it may not be possible to define unique extension numbers for all users within the private network. For these types of networks, additional administration may be required as described in Reference [6] in Section 9. Use the change private-numbering n command, where n is the leading digit of the private number. Fill in the indicated fields as shown below and use default values for remaining fields. Ext Len Enter length of extension numbers. In the sample configuration, 5 was used. Ext Code Enter leading digit9s) from extension number. In the sample configuration 5 is used. Total Len Enter the Total Length of digits. change private-numbering 5 Page 1 of 2 NUMBERING - PRIVATE FORMAT Ext Ext Trk Private Total Len Code Grp(s) Prefix Len Total Administered: Maximum Entries: Administer Public-Unknown-Numbering Although the trunk is set to private, the calling number will be transformed into a public number when the called number is determined to be public by the settings of the AAR and Route Pattern for the called number. For this adaptation, an entry in the public unknown numbering for the calling number is required. Use the change public-unknown-numbering n command, where n is the leading digit of the public number. Fill in the indicated fields as shown below and use default values for remaining fields. 13 of 54

14 Ext Len Ext Code Total CPN Len Enter length of extension numbers. In the sample configuration, 5 was used. Enter leading digit9(s) from extension number. In the sample configuration 5 is used. Enter the Total Length of digits. change public-unknown-numbering 5 Page 1 of 2 NUMBERING - PUBLIC/UNKNOWN FORMAT Total Ext Ext Trk CPN CPN Len Code Grp(s) Prefix Len Total Administered: Maximum Entries: Note: If an entry applies to a SIP connection to Avaya Aura(R) Session Manager, the resulting number must be a complete E.164 number. Communication Manager automatically inserts a '+' digit in this case Administer AAR Communication Manager uses AAR routing information to route calls within the company over its own private network. Communication Manager converts the number that the user dials. The software then analyzes the number, and routes the call as a private-network call. Use the change aar analysis n command, where n is the leading digit of the dialed number. Fill in the indicated fields as shown below and use default values for remaining fields. Dialed String Enter leading digit(s) from Scopia XT extensions range. In the sample configuration, 5204 was used. Min Enter minimum number of digits that must be dialed. In the sample configuration, 5 was used. Max Enter maximum number of digits that must be dialed. In the sample configuration, 5 was used. Route Pattern Enter Route Pattern defined in Section Call Type Enter unku. change aar analysis 5 Page 1 of 2 AAR DIGIT ANALYSIS TABLE Location: all Percent Full: 0 Dialed Total Route Call Node ANI String Min Max Pattern Type Num Reqd p2 aar n aar n unku n unku n p2 aar n p2 unku n p2 unku n 14 of 54

15 4.8. Administer Audio Codec Use the change ip-codec-set x command where x is the number used to identify the codec set. Enter the following values noted below and use default values for remaining fields. Audio Codec Enter G.711MU. change ip-codec-set 2 Page 1 of 2 Codec Set: 2 IP Codec Set Audio Silence Frames Packet Codec Suppression Per Pkt Size(ms) 1: G.711MU n : 3: 4: On Page 2, enter the following values: Allow Direct-IP Multimedia? Enter y. Maximum Call Rate for Direct-IP Multimedia: Enter Maximum Call Rate for Priority Direct-IP Multimedia: Enter change ip-codec-set 2 Page 2 of 2 IP Codec Set Allow Direct-IP Multimedia? y Maximum Call Rate for Direct-IP Multimedia: 1920:Kbits Maximum Call Rate for Priority Direct-IP Multimedia: 1920:Kbits Mode Redundancy FAX relay 0 Modem off 0 TDD/TTY US 3 Clear-channel n 0 15 of 54

16 4.9. Administer Network Region Use the change ip-network-region n command where n is an available network region. Enter the following values and use default values for remaining fields. Authoritative Domain: Enter the correct SIP domain for the configuration. For the sample configuration, dr.avaya.com was used. Name: Codec Set: Enter descriptive name. Enter the number of the IP codec set configured in Section 4.8. change ip-network-region 4 Page 1 of 20 IP NETWORK REGION Region: 4 Location: Authoritative Domain: dr.avaya.com Name: ScopiaRadvision H323 Stub Network Region: n MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 2 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? n UDP Port Max: 3329 DIFFSERV/TOS PARAMETERS Call Control PHB Value: 46 Audio PHB Value: 46 Video PHB Value: P/Q PARAMETERS Call Control 802.1p Priority: 6 Audio 802.1p Priority: 6 Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS H.323 IP ENDPOINTS RSVP Enabled? n H.323 Link Bounce Recovery? y Idle Traffic Interval (sec): 20 Keep-Alive Interval (sec): 5 Keep-Alive Count: Administer Node Names Using the change node-names ip command, add node Name and IP Addresses for the SM- 100 interfaces for both Session Managers and the Scopia Management server, if not previously added. Note: IP addresses have been partially hidden for security. change node-names ip Page 1 of 2 IP NODE NAMES Name IP Address procr X.X scopiadtsvr1 X.X silasm3 X.X silasm4 X.X of 54

17 4.11. Administer Signaling Groups Administer Signaling Group for H.323 Trunk Use the add signaling-group n command, where n is an available signaling group number to create a H.323 signaling group. In the sample configuration, trunk group 4 and signaling group 4 were used for connecting to Session Manager. On Page 1, enter the following values and use default values for remaining fields. Group Type: Enter h.323 IP Video Enter y. Priority Video Enter y. Trunk Group for Channel Selection After completing Section come back to this signaling group and insert the trunk group number here. Near-end Node Name: Enter procr node name from Section Far-end Node Name: Enter node name for Scopia Management ECS server defined in Section Near-end Listen Port: Verify 1719 is used. Far-end Listen Port: Verify 1719 is used. Far-end Network Region Enter network region defined in Section 4.9 LRQ Required? Enter y. H.323 Station Outgoing Direct Media? Enter y add signaling-group 4 Page 1 of 6 SIGNALING GROUP Group Number: 4 Group Type: h.323 SBS? n Remote Office? n Max number of NCA TSC: 0 Q-SIP? n Max number of CA TSC: 0 IP Video? y Priority Video? y Trunk Group for NCA TSC: Trunk Group for Channel Selection: 4 TSC Supplementary Service Protocol: a X-Mobility/Wireless Type: NONE Network Call Transfer? n T303 Timer(sec): 10 H.245 DTMF Signal Tone Duration(msec): Near-end Node Name: procr Far-end Node Name: scopiadtsvr1 Near-end Listen Port: 1719 Far-end Listen Port: 1719 Far-end Network Region: 4 LRQ Required? y Calls Share IP Signaling Connection? n RRQ Required? n Bypass If IP Threshold Exceeded? n H.235 Annex H Required? n DTMF over IP: out-of-band Direct IP-IP Audio Connections? y Link Loss Delay Timer(sec): 90 IP Audio Hairpinning? n Enable Layer 3 Test? y Interworking Message: PROGress H.323 Station Outgoing Direct Media? y DCP/Analog Bearer Capability: 3.1kHz 17 of 54

18 Administer Signaling Group for SIP Trunk Use the add signaling-group n command, where n is an available signaling group number to create SIP signaling groups. In the sample configuration, trunk group 10 and signaling group 10 were used for connecting to Session Manager. On Page 1, enter the following values and use default values for remaining fields. Group Type: Enter sip Transport Method: Enter tls. (Note: tcp can be used as well, check with your Network Administrator). IP Video Enter y Priority Video Enter y Peer Detection Enabled? Enter y Peer Server: Use default value Note: default value is replaced with SM after SIP trunk to Session Manager is established Near-end Node Name: Enter procr node name from Section Far-end Node Name: Enter node name of SIL Site A Session Manager defined in Section 4.9 Near-end Listen Port: Verify 5061 is used Far-end Listen Port: Verify 5061 is used Far-end Domain: Enter domain name for Authoritative Domain field as defined in Section 4.9. add signaling-group 10 Page 1 of 2 SIGNALING GROUP Group Number: 10 Group Type: sip IMS Enabled? n Transport Method: tls Q-SIP? n IP Video? y Priority Video? y Enforce SIPS URI for SRTP? y Peer Detection Enabled? y Peer Server: SM Prepend '+' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers? y Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers? n Near-end Node Name: procr Far-end Node Name: silasm3 Near-end Listen Port: 5061 Far-end Listen Port: 5061 Far-end Network Region: Far-end Secondary Node Name: Far-end Domain: dr.avaya.com Bypass If IP Threshold Exceeded? n Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y Session Establishment Timer(min): 3 IP Audio Hairpinning? n Enable Layer 3 Test? y Initial IP-IP Direct Media? y H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6 18 of 54

19 4.12. Administer Trunk Groups Administer H.323 Trunk Group Add the corresponding trunk groups controlled by the signaling group defined Section using the add trunk-group n command where n is an available trunk group number. Fill in the indicated fields as shown below. Default values can be used for the remaining fields. NOTE: Don t forget after adding this trunk group to go back to the signaling group and add it to the Trunk Group for Channel Selection field. Group Type: Group Name: TAC: Direction: Carrier Medium: Service Type: Member Assignment: Signaling Group: Number of Members: Enter isdn. Enter a descriptive name. Enter an available trunk access code. Enter two-way. Enter H.323. Enter tie. Enter auto. Enter the number of the signaling group added in Section Enter the number of members in the H.323 trunk. Note: once the add trunk-group command is completed, trunk members will be automatically generated based on the value in the Number of Members field. add trunk-group 4 Page 1 of 21 TRUNK GROUP Group Number: 4 Group Type: isdn CDR Reports: y Group Name: ScopiaRadvision H323 Trunk COR: 1 TN: 1 TAC: #004 Direction: two-way Outgoing Display? n Carrier Medium: H.323 Dial Access? n Queue Length: 0 Service Type: tie Busy Threshold: 255 Night Service: Auth Code? n Member Assignment Method: auto Signaling Group: 4 Number of Members: of 54

20 On Page 3, fill in the indicated fields as shown below. Default values can be used for the remaining fields. Send Name: Enter y. Send Calling Number: Enter y. Format: Enter private. Replace Restricted Numbers? Enter y. Replace Unavailable Numbers? Enter y. Send Connected Number? Enter y. display trunk-group 4 Page 3 of 21 TRUNK FEATURES ACA Assignment? n Measured: none Internal Alert? n Maintenance Tests? y Data Restriction? n NCA-TSC Trunk Member: Send Name: y Send Calling Number: y Used for DCS? n Send EMU Visitor CPN? n Suppress # Outpulsing? n Format: private UUI IE Treatment: service-provider Network Call Redirection: none Send UUI IE? y Send UCID? n Send Codeset 6/7 LAI IE? y Replace Restricted Numbers? y Replace Unavailable Numbers? y Send Connected Number: y Hold/Unhold Notifications? n Modify Tandem Calling Number: no Show ANSWERED BY on Display? y DSN Term? n 20 of 54

21 Administer SIP Trunk Add the corresponding trunk groups controlled by the signaling groups defined Section using the add trunk-group n command where n is an available trunk group number. Fill in the indicated fields as shown below. Default values can be used for the remaining fields. Group Type: Enter sip. Group Name: Enter a descriptive name. TAC: Enter an available trunk access code. Direction: Enter two-way. Outgoing Display? Enter y. Service Type: Enter tie. Signaling Group: Section Number of Members: Enter the number of the signaling group added in Enter the number of members in the SIP trunk (must be within limits configured in Section 4.1.2). Note: once the add trunk-group command is completed, trunk members will be automatically generated based on the value in the Number of Members field. add trunk-group 10 Page 1 of 21 TRUNK GROUP Group Number: 10 Group Type: sip CDR Reports: y Group Name: SIP TG to silasm4 COR: 1 TN: 1 TAC: #010 Direction: two-way Outgoing Display? y Dial Access? n Night Service: Queue Length: 0 Service Type: tie Auth Code? n Member Assignment Method: auto Signaling Group: 10 Number of Members: 6 On Page 3, fill in the indicated fields as shown below. Default values can be used for the remaining fields. Numbering Format: Enter private. add trunk-group 10 Page 3 of 21 TRUNK FEATURES ACA Assignment? n Measured: none Maintenance Tests? y Numbering Format: private UUI Treatment: service-provider Replace Restricted Numbers? n Replace Unavailable Numbers? n Modify Tandem Calling Number: no Show ANSWERED BY on Display? y DSN Term? n SIP ANAT Supported? N 21 of 54

22 4.13. Administer Route Patterns Administer H.323 Route Pattern This section describes the Route Pattern screen. Calls originating from Avaya Aura H.323 endpoints to the Scopia Elite MCU must be routed via the H.323 Trunk Group. Use the change route-pattern n command, where n is the associated route pattern. Enter a descriptive name for Pattern Name. The sample configuration uses to Scopia ECS. Enter the H.323 Trunk Group created in Section for the field Grp No. The sample configuration uses 4. Enter the appropriate Facility Restriction Level for the field FRL. In the sample configuration FRL 0 was used which is the least restrictive. Use default values for remaining fields. change route-pattern 4 Page 1 of 3 Pattern Number: 4 Pattern Name: to Scopia ECS SCCAN? n Secure SIP? n Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC No Mrk Lmt List Del Digits QSIG Dgts Intw 1: 4 0 n user 2: n user 3: n user 4: n user Administer SIP Route Pattern This section describes the Route Pattern screen. Calls originating from Avaya Aura SIP endpoints to the Scopia Elite MCU must be routed via the SIP Trunk Group. This routepattern will be used during Partition Group Routing. Use the change route-pattern x command, where x is the associated route pattern. Enter a descriptive name for Pattern Name. The sample configuration uses Route to silasm4. Enter the SIP Trunk Group created in Section for the field Grp No. The sample configuration uses 10. Enter the appropriate Facility Restriction Level for the field FRL. In the sample configuration FRL 0 was used which is the least restrictive. Use default values for remaining fields. change route-pattern 4 Page 1 of 3 Pattern Number: 4 Pattern Name: Route to silasm4 SCCAN? n Secure SIP? n Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC No Mrk Lmt List Del Digits QSIG Dgts Intw 1: 10 0 n user 2: n user 3: n user 4: n user Administer Incoming Call Handling Treatment The Incoming Call Handling Treatment form on Communication Manager specifies unique call treatment for different incoming calls on any trunk group. This screen is available only if digit handling is administered as overlap on the in side, and the direction is outgoing. 22 of 54

23 Incoming Call Handling Treatment Table entries with a predefined service or feature always appear before entries with a user-defined service or feature. Use the change inc-call-handling-trmt trunk-group n, where n is the assigned SIP trunk group in Section Fill in the indicated fields as shown below. In this sample configuration our extension lengths are 5 and an extra digit is inserted when leaving the Radvision Scopia ECS so our Number Len is 6. This Number Len will be determined by the length of your extensions plus the number of digits inserted by the Radvision Scopia ECS. Default values can be used for the remaining fields. Number Len: Enter 6. Number Digits: Enter 45. Del: Enter 1. change inc-call-handling-trmt trunk-group 4 Page 1 of 30 INCOMING CALL HANDLING TREATMENT Service/ Number Number Del Insert Per Call Night Feature Len Digits CPN/BN Serv tie tie tie Save Translations Configuration of Communication Manager Evolution Server is complete. Use the save translation command to save these changes. Note: After making a change on Communication Manager which alters the dial plan or numbering plan, synchronization between Communication Manager and System Manager must be completed. See Section 5.7 for steps on performing System Manger synchronization. 23 of 54

24 5. Configure Avaya Aura Session Manager This section describes the procedures for configuring the secondary Avaya Aura Session Manager to support registrations of SIP endpoints. These application notes assume other administration activities have already been completed such as defining the network connection between System Manager and Session Manager, defining the Session Manager instance and defining Communication Manager as a Managed Element. For more information on these additional actions, see References [1] through [5] in Section 9. In addition, these application notes assume the Flare Experience for Windows SIP users have already been defined. See References [9] through [11] in Section 9 for more information of the setup and configuration of Flare Experience for Windows. The following administration activities will be described: Define SIP Domain Define Locations Define SIP Entities Define Entity Links Define Routing Policies Define Dial Patterns Synchronize changes with Communication Manager Note: Some administration screens have been abbreviated for clarity. Configuration is accomplished by accessing the browser-based GUI of Avaya Aura System Manager, using the URL where <ip-address> is the IP address of Avaya Aura System Manager. Login with the appropriate credentials. 24 of 54

25 5.1. Define SIP Domains Expand Elements Routing and select Domains from the left navigation menu. Click New (not shown). Enter the following values and use default values for remaining fields. Name Enter the Authoritative Domain Name specified in Section 4.9. For the sample configuration, dr.avaya.com was used. Type Select sip from drop-down menu. Notes Add a brief description. [Optional]. Click Commit to save. The screen below shows the SIP Domain defined for the sample configuration Define Locations Locations are used to identify logical and/or physical locations where SIP Entities or SIP endpoints reside, for purposes of bandwidth management or location-based routing. Expand Elements Routing and select Locations from the left navigation menu. Click New (not shown). In the General section, enter the following values and use default values for remaining fields. Name: Enter a descriptive name for the location. Notes: Add a brief description. [Optional]. Click Commit to save. The screen below shows the Location used for Session Manager in the sample configuration. Note: screen has been abbreviated for clarity. 25 of 54

26 Repeat the steps to define a Location for the secondary site Define SIP Entities A SIP Entity must be added for Avaya Aura Communication Manager and Avaya Aura Session Manager. To add a SIP Entity, expand Elements Routing and select SIP Entities from the left navigation menu (not shown) Add Session Manager SIP Entity Click New (not shown). In the General section, enter the following values and use default values for the remaining fields. Name Enter an identifier for new SIP Entity. In the sample configuration, silasm4 was used. FQDN or IP Address Enter IP address of the signaling interface for Session Manager. Type Select Session Manager. Location Select the applicable Location defined for the Session Manager from Section 5.2 Notes Enter a brief description. [Optional]. In the SIP Link Monitoring section: SIP Link Monitoring Select Use Session Manager Configuration. Click Commit to save SIP Entity definition. The following screen shows the SIP Entity defined for Session Manager. Note: IP addresses have been partially hidden for security Add Communication Manager SIP Entity Click New (not shown). In the General section, enter the following values and use default values for remaining fields. Name: Enter an identifier for new SIP Entity. In the sample configuration, CM_PROCR was used. 26 of 54

27 FQDN or IP Address: Type: Location: Notes: Enter IP address of the PROCR interface for Communication Manager. Select CM. Select the applicable Location defined Session Manager from Section 5.2 Enter a brief description. [Optional]. In the SIP Link Monitoring section: SIP Link Monitoring: Select Use Session Manager Configuration. Click Commit to save SIP Entity definition. The following screen shows the SIP Entity defined for Communication Manager. Note: IP addresses have been partially hidden for security Define Entity Links A SIP trunk between Session Manager and Communication Manager is described by an Entity Link. In the sample configuration, a SIP Entity Link was added between Avaya Aura Communication Manager and Avaya Aura Session Manager. To add an Entity Link, expand Elements Routing and select Entity Links from the left navigation menu. Click New (not shown). Enter the following values. Name Enter an identifier for the link to Communication Manager. In the sample configuration, silasm4_cm4_procr_5061_tls was used. SIP Entity 1 Select the SIP Entity for Session Manager defined Section from the drop-down menu. SIP Entity 2 Select the SIP Entity for Communication Manager defined in Section from the drop-down menu. 27 of 54

28 Protocol After selecting both SIP Entities, verify TLS is selected as the required Protocol. Port Verify Port for both SIP entities is Policy Select Trusted from the drop-down menu. Click Commit to save Entity Link definition. The following screen shows the Entity Link defined between Avaya Aura Communication Manager Evolution Server and Avaya Aura Session Manager Define Routing Policy Routing policies describe the conditions under which calls will be routed to non-sip stations on Communication Manager Evolution Server. Note: Since the SIP users are registered to Session Manager, a routing policy does not need to be defined for calls to SIP endpoints supported by Communication Manager Evolution Server. To add a routing policy, expand Elements Routing and select Routing Policies. Press New (not shown). In the General section, enter the following values. Name: Enter an identifier for Communication Manager Evolution Server. Disabled: Leave unchecked. Retries: Retain default value of 0. Notes: Enter a brief description. [Optional]. 28 of 54

29 In the SIP Entity as Destination section, click Select. The SIP Entity List page opens (not shown). Select the SIP Entity defined for Communication Manager and click Select. The selected SIP Entity displays on the Routing Policy Details page. Use default values for remaining fields. Press Commit to save Routing Policy definition. Note: IP addresses have been partially hidden for security. Note: the routing policy defined in this section is an example used in the sample configuration. Other routing policies may be appropriate for different customer networks Define Dial Patterns Dial Patterns determine which routing policy will be applied. To add a Dial Pattern, expand Elements Routing and select Dial Patterns. Click New (not shown). In the General section, enter the following values. Pattern: Enter pattern of 52 Min: Enter the value 5 29 of 54

30 Max: Enter the value 5 SIP Domain: Retain the default value -ALL- Notes: Enter a brief description. [Optional]. In the Originating Locations and Routing Policies section, click Add. The Originating Location page opens. In the Originating Location section, select Apply The Selected Routing Policies to All Originating Locations. 30 of 54

31 In the Routing Policies section, select the Routing Policy created in Section 5.5. Click Select. Click Commit to save (not shown). 31 of 54

32 5.7. Synchronize Changes with Avaya Aura Communication Manager After completing these changes in System Manager, perform an on demand synchronization. Navigate to Elements Inventory Synchronization Communication System. On the Synchronize CM Data and Configure Options page, expand the Synchronize CM Data/Launch Element Cut Through table and select the row associated with Communication Manager as shown below. Click to select Incremental Sync data for selected devices option. Press Now to start the synchronization. Use the Refresh button in the table header to verify status of the synchronization. Verify synchronization successfully completes by verifying the status in the Sync Status column shows Completed. Note: Depending on the number of administration changes made, synchronization might take several minutes to complete. 32 of 54

33 6. Configure Scopia Management This section describes the procedures for configuring the Scopia Management for integration with Avaya Aura. These instructions assume other administration activities have already been completed such as network configuration and product licensing for both Radvision Scopia XT 5000 endpoints and Scopia Management. These instructions also assume the software and firmware installation is complete. For more information on these additional actions, see References [14] through [18] in Section 9. The following administration activities will be described: Define Locations Define Gatekeepers (Avaya Aura Communication Manager and ECS) Define Radvision Scopia XT 5000 Endpoint Note: Some administration screens have been abbreviated for clarity. Configuration is accomplished by accessing the browser-based GUI of Scopia Management, using the URL where <ip-address> is the IP address of Scopia Management. Log in with the appropriate credentials Add Location Navigate to Settings Topology Locations. Click Add (not shown). Enter the following value and use default values for remaining fields. Location Name Enter a descriptive name for the Location For the sample configuration, SIL LAB SITE A was used. Click Add. 33 of 54

34 Repeat these steps to add a location for the secondary site. For the sample configuration SIL LAB SITE B was used. Repeat these steps to add a location for Radvision resources. For the sample configuration RADVISION RESOURCES was used. 34 of 54

35 6.2. Add Communication Manager as Gatekeeper Navigate to Devices Devices by Type Gatekeepers. Click Add (not shown). Enter the following values and use default values for remaining fields. Name: Enter a descriptive name for Communication Manager For the sample configuration, CM4 was used. IP Address: Enter the IP Address of SIL LAB SITE A Communication Manager. This is the same address as defined in Section Model: Select Other Model from the drop down list. Location: Select the location defined in Section 6.1 from the drop down list. Strip Local Zone Prefix(es) Ensure this box is checked. Zone Prefix: Enter the prefix used to identify the zone used for Communication Manager. The sample configuration uses the prefix 4 for SIL LAB SITE A. Note: IP addresses have been partially hidden for security. Click OK. Repeat this step to add SIL LAB SITE B Communication Manager with a Zone Prefix of 8 35 of 54

36 6.3. Add ECS Gatekeeper The ECS Gatekeeper will be configured during the base install of Scopia Management. The configuration is displayed to show the lab configuration. Navigate to Devices Devices by Type Gatekeepers. Click Add (not shown). Enter the following values and use default values for remaining fields. Name: Enter a descriptive name for ECS Gatekeeper For the sample configuration, local_gatekeeper was used IP Address: Enter the IP Address of the ECS Gatekeeper. The sample configuration used X.X Model: Select RADVISION ECS from the drop down list. Location: Select the location defined in Section 6.1 from the drop down list. The sample configuration uses RADVISION RESOURCES. Strip Local Zone Prefix(es) Ensure this box is checked. Zone Prefix: Enter the prefix used to identify the zone used for Communication Manager. The sample configuration uses the prefix 4 for SIL LAB SITE A. Note: IP addresses have been partially hidden for security. Click OK. 36 of 54

37 6.4. Add Radvision Scopia XT 5000 Endpoint Navigate to Endpoints and click Add. Click Add Manually (not shown). Enter the following values and use default values for remaining fields. Name: Enter a descriptive name for the Radvision Scopia XT 5000 endpoint. For the sample configuration, XT5000SIL was used. Description: Enter a descriptive name for the Radvision Scopia XT 5000 endpoint. [Optional]. Type: Retain the default value Single Codec Endpoint. Protocol: Select IP (H.323) from the drop down list. E.164/IP Address: Enter the Registered To: Location: Select the ECS local gatekeeper created in Section 6.3 from the drop down list. The sample configuration uses local_gatekeeper (X.X.10.20). Select the location created in Section 6.1 for RADVSION RESOURCES. Note: IP addresses have been partially hidden for security. Click OK. 37 of 54

38 7. Verification Steps 7.1. Verify Avaya Aura Session Manager Configuration Step 1: Verify Avaya Aura Session Manager is operational Log in to System Manager. Expand Elements Session Manager and select Dashboard to verify the overall system status for Session Manager. Specifically, verify the status of the following fields as shown below: Tests Pass Security Module Service State Data Replication Step 2: Verify SIP Entity Link Status Navigate to Elements Session Manager System Status SIP Entity Monitoring to view more detailed status information for the specific SIP Entity Links used for calls. Select the SIP Entity for Communication Manager Evolution Server from the All Monitored SIP Entities table (not shown) to open the SIP Entity, Entity Link Connection Status page. In the All Entity Links to SIP Entity: CM4_PROCR table, verify the Conn. Status for both links is Up for Communication Manager. 38 of 54

39 Note: IP addresses have been partially hidden for security Verify Radvision Scopia Configuration Verify Radvision Scopia XT 5000 Endpoint Registration Step 1: Verify Radvision Scopia XT 5000 Endpoint is registered with Scopia Management. Log in to Scopia Management. Navigate to Endpoints Endpoints by Location All. Verify the Radvision Scopia XT 5000 endpoint is registered. Specifically, verify the status of the endpoint shows a green status indicator as shown below: Note: IP addresses have been partially hidden for security. 39 of 54

40 Verify Communication Manager Configuration Status SIP Trunk and Signaling Groups Verify the status of one of the SIP Trunk group by using the status trunk-group n command where n is one of the trunk group numbers administered in Section Verify the trunk group is in-service/idle as indicated in the Service State: field shown below: status trunk 10 Page 1 TRUNK GROUP STATUS Member Port Service State Mtce Connected Ports Busy 0011/001 T00001 in-service/idle no 0011/002 T00002 in-service/idle no 0011/003 T00003 in-service/idle no 0011/004 T00004 in-service/idle no 0011/005 T00005 in-service/idle no 0011/006 T00006 in-service/idle no 0011/007 T00007 in-service/idle no 0011/008 T00008 in-service/idle no 0011/009 T00009 in-service/idle no 0011/010 T00010 in-service/idle no 0011/011 T00011 in-service/idle no 0011/012 T00012 in-service/idle no 0011/013 T00013 in-service/idle no 0011/014 T00014 in-service/idle no Verify the status of one of the SIP signaling groups by using the status signaling-group n command where n is one of the signaling group numbers administered in Section Verify the signaling group is in-service as indicated in the Group State: field shown below: status signaling-group 11 STATUS SIGNALING GROUP Group ID: 11 Group Type: sip Group State: in-service 40 of 54