Application Notes for Configuring Scopia XT5000 Series as a SIP Endpoint with Avaya Aura Session Manager Issue 1.0

Transcription

1 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Scopia XT5000 Series as a SIP Endpoint with Avaya Aura Session Manager Issue 1.0 Abstract These Application Notes describe the configuration steps required to integrate Radvision Scopia XT5000 Series as a SIP Endpoint with Avaya Aura Session Manager. In the interoperability testing, calls utilize Scopia XT5000 Series as a SIP Endpoint registered with Session Manager and Avaya Flare Experience for Windows. These Application Notes provide information for the setup, configuration, and verification of the call flows tested on this solution. 1 of 48

2 1. Introduction These Application Notes describe the configuration steps required to integrate Scopia XT5000 Series endpoint with an Avaya SIP-enabled enterprise solution. The Scopia XT5000 Series endpoint is registered with Avaya Aura Session Manager as a SIP endpoint. The Avaya SIP-enabled enterprise solution consists of consists of Avaya Aura Communication Manager, Avaya Aura System Manager and Avaya Aura Session Manager. Avaya Aura Communication Manager is configured as an evolution server and acts as the telephony application server for Session Manager. The role of the Avaya Aura Session Manager in the reference architecture is to act as a Registrar for Avaya SIP endpoints and providing a centralized dial-plan for least-cost and time-of-day based routing. Avaya Aura System Manager provides a web-based interface for the provisioning and maintenance of Avaya Aura Session Manager. The Scopia XT Series endpoints incorporate state-of-the-art video technology for high definintion (HD) conferencing. The Scopia XT5000 Series endpoint provides HD 1080p video at 60 frames per second (fps) capabilities for video conferencing. 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 Scopia XT5000 Series as a SIP Endpoint registered with Session Manager and Avaya Flare Experience for Windows. The bi-directional audio and video calls include basic Make Call, Release Call, Answer Call, Call Hold and Retrieve, Video Start/Stop and Calls with Long Duration.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 Scopia XT5000 Series endpoint registered with Avaya Aura Session Manager using the SIP Protocol. Calls: Place/Answer Audio/Video calls between Scopia XT5000 Series endpoints and Avaya Flare Experience for Windows endpoints. o Make Call o Release Call o Answer Call o Hold/Un-Hold Call o Video Start/Stop o Audio/Video Mute o Long Duration 2 of 48

3 2.2. Test Results and Observations Overall, testing of the sample configuration is complete with the following observations: XT Series Endpoints (XT5000, XT4200 & XT1200) do not support G.729 audio codec. Currently, G.711, G.722, G.722.1C, G.719, AAC-LD audio codecs are supported. 3. Reference Configuration The diagram below Figure 1 shows the tested configuration. The sample configuration 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. The sample configuration 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. The Scopia XT5000 endpoint at this location is registered with Avaya Aura Session Manager using the SIP Protocol. Note: the IP address scheme depicted within are ficticious. The IP address scheme is used for demonstration. 3 of 48

4 Figure 1: Compliance Testing Configuration 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 XT5000 (SIP) Release Avaya Flare Experience on Windows (SIP) Release Build 85 4 of 48

5 4. Configure Avaya Aura Communication Manager This section provides the procedures for configuring Communication Manager. The procedures include the following areas: Verify License Enable Trunk-to-Trunk transfers Administer Dial Plan Administer Uniform Dial Plan Administer Private Numbering Administer Public Unknown Numbering Administer Node Names Administer Network Regions Administer IP codec set Administer AAR Administer Signaling Groups Administer SIP Trunk 4.1. 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 features is not available, contact an authorized Avaya sales representative to make the appropriate changes 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 Off-PBX Telephones - PVFMC: Maximum Off-PBX Telephones - SCCAN: 0 0 Maximum Survivable Processors: of 48

6 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: 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 6 of 48

7 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 7 of 48

8 4.2. 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 set 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 trunk-to-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 Dialplan This section describes the Dial Plan Analysis screen. Use the Dial Plan feature 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 5 and with a Total Length of 5 digits was used to administer the extension range used for the Scopia XT SIP 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 SIP endpoints Total Length Enter extension range length Call Type Enter ext 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 41 5 ext udp 42 5 ext 9 1 fac 43 5 ext * 3 fac udp # 4 dac 5 5 ext 8 of 48

9 4.4. Administer Uniform Dialplan Use the Uniform Dial Plan (UDP) feature to share a common dial plan among a group of Communication Manager 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 x where x is the first digit of the extension range that will be used Scopia XT SIP endpoints. The dialed string beginning with the Matching Pattern 5 and with a Total Length of 5 digits was used to administer the extension range used for the Scopia XT SIP endpoints. Fill in the indicated fields as shown below and use default values for remaining fields. Matching Pattern Enter digit pattern of Scopia XT endpoints Len Enter extension length Del Enter number of digits to delete if applicable Net 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 x command, where x is the length 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, 9 of 48

10 Ext Code 5 was used. Enter leading digit(s) from extension number. In the sample configuration, 5 is used. 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 unknownnumbering for the calling number is required. Use the change public-unknown-numbering n command, where n is the length 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 digit(s) from extension number. In the sample configuration, 5 is used. 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. 10 of 48

11 4.7. Administer AAR Communication Manager uses AAR routing information to route calls over the private network. Communication Manager software 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 length of the private 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 extension range. In the sample configuration, 5 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 aar 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 4.8. Administer Node Names Using the change node-names ip command, add the node-name and IP Addresses for the SM- 100 interface and Scopia Management server, if not previously added. change node-names ip Page 1 of 2 IP NODE NAMES Name IP Address procr scopiadtsvr silasm of 48

12 4.9. 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: Audio Codec Silence Suppression Frames Per Pkt Enter 2. Packet Size (ms) Enter 20. Enter G.711MU and G.729 as supported types. Retain the default value n. 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 : G.729 n : 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 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. Name: Enter descriptive name. Authoritive Domain: Enter dr.avaya.com. Codec Set: Enter the number of the IP codec set configured in Section of 48

13 Intra-region IP-IP Direct Audio: Inter-region IP-IP Direct Audio: Enter yes. Enter yes. 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 Signaling Group for SIP Trunk Use the add signaling-group x command, where x is an available signaling group number to create SIP signaling group. 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 IMS Enabled? Enter n. Transport Method: Enter tls. 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 4.8. Far-end Node Name: Enter node name for Session Manager defined in Section 4.8. Near-end Listen Port: Verify 5061 is used. Far-end Listen Port: Verify 5061 is used. 13 of 48

14 Far-end Network Region: Enter network region defined in Section 4.10 Far-end Domain: Enter domain name for Authoritative Domain field defined in Section 4.10 Initial IP-IP Direct Media? Enter y. 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: silasm4 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 Initial IP-IP Direct Media? y H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6 14 of 48

15 4.12. Administer SIP Trunk Add the corresponding trunk group controlled by the signaling group defined Section 4.11 using the add trunk-group x command where x 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: Enter the number of the signaling group added in Section Number of Members: 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: 5 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 2, set the Preferred Minimum Session Refresh Interval: field to add trunk-group 10 Page 2 of 21 Group Type: sip TRUNK PARAMETERS Unicode Name: auto Redirect On OPTIM Failure: 5000 SCCAN? n Digital Loss Group: 18 Preferred Minimum Session Refresh Interval(sec): 1200 Disconnect Supervision - In? y Out? y XOIP Treatment: auto Delay Call Setup When Accessed Via IGAR? n 15 of 48

16 On Page 3, fill in the indicated fields as shown below. Default values can be used for the remaining fields. Numbering Format: Enter private. Show ANSWERED BY on Display? Enter y. 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 On Page 4, fill in the indicated field as shown below. Default values can be used for the remaining fields. Support Request History? Enter y. add trunk-group 10 Page 4 of 21 PROTOCOL VARIATIONS Mark Users as Phone? n Prepend '+' to Calling/Alerting/Diverting/Connected Number? n Send Transferring Party Information? n Network Call Redirection? n Send Diversion Header? n Support Request History? y Telephone Event Payload Type: Convert 180 to 183 for Early Media? n Always Use re-invite for Display Updates? n Identity for Calling Party Display: P-Asserted-Identity Block Sending Calling Party Location in INVITE? n Accept Redirect to Blank User Destination? n Enable Q-SIP? N 16 of 48

17 4.13. Administer SIP Route Pattern This section describes the Route Pattern screen. Calls originating from Scopia XT5000 SIP endpoint will be routed via the SIP Trunk Group. 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 4.12 for the field Grp No. The sample configuration uses 10. Enter 0 for the field FRL. Use default values for remaining fields. change route-pattern 10 Page 1 of 3 Pattern Number: 10 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 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. 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 instructions assume other administration activities have already been completed such as defining the network connection between System Manager and Session Manager, defining 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. The following administration activities will be described: Define SIP Domain Define Locations Define SIP Entities Define Entity Links, which describe the SIP trunk parameters used by Session Manager when routing calls between SIP Entities Define Routing Policies and Dial Patterns which control routing between SIP Entities 17 of 48

18 Add new SIP Users Synchronize changes with Communication Manager Perform Data Replication with Session 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. Log in with the appropriate credentials. 18 of 48

19 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 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. 19 of 48

20 5.2. 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 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). 20 of 48

21 Add Session 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, 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 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. 21 of 48

22 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. FQDN or IP Address: Enter IP address of the PROCR interface for Communication Manager. Type: Select CM. Location: Select the applicable Location defined for Communication 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 Communication Manager. 22 of 48

23 5.4. 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 server. 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. from the drop-down menu. 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. 23 of 48

24 5.5. 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. Click 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]. 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. Click Commit to save Routing Policy definition. 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. The following screen shows the Routing Policy for Communication Manager Evolution Server. 24 of 48

25 5.6. 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 51 Min: Enter the value 5 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. 25 of 48

26 In the Routing Policies section, select the Routing Policy created in Section 5.5. Click Select. Click Commit to save (not shown). 26 of 48

27 5.7. Add SIP User To add new SIP users, expand Users User Management and select Manage Users from left navigation menu. Step 1: Click New (not shown). Enter values for the following required attributes for a new SIP user in the Identity section and use default values for remaining fields. Last Name: Enter last name of user. First Name: Enter first name of user. Login Name: Enter extension number@<domain> where <domain> matches the domain defined in Section 5.1. Authentication Type: Verify Basic is selected. Password: Enter password used to log into System Manager. Confirm Password: Repeat value entered above. Localized Display Name: Enter display name for user [Optional]. The screen below shows results from Step 1 for a new SIP user. 27 of 48

28 Click Commit & Continue to save changes from Step 1. Step 2: Select Communication Profile tab and enter the value the endpoint will use to register to Session Manager in the Communication Profile Password and Confirm Password fields. Verify there is a default entry identified as the Primary profile as shown below: If an entry does not exist, select New and enter values for the following required attributes: Name: Enter Primary. Default: Verify has been entered. 28 of 48

29 Step 3: Expand Communication Address sub-section and select New to define a Communication Address for the new user. Enter values for the following required attributes: Type: Select Avaya SIP from drop-down menu. Fully Qualified Address: Enter same extension number as used for Login Name in Step 1. Domain: Verify value matches Domain name defined in Section 5.1. Press Add to save the Communication Address. 29 of 48

30 Step 4: Scroll down to the Session Manager Profile section and enter to expand section. These application notes assume the Application and Application sequence has already been defined for Communication Manager. Enter the following values as noted below. Use default values for the remaining fields. Primary Session Manager Origination Application Sequence Termination Application Sequence Home Location Select silasm4 from drop-down menu. Select Application Sequence for Communication Manager. Select Application Sequence for Communication Manager. Select the applicable Location for the user defined in Section 5.2 Note: After selecting the values for the Session Manager Profile section, verify the CS 1000 Endpoint Profile section is not selected as shown below. 30 of 48

31 Step 5: Scroll down to the CM Endpoint Profile section and enter to expand section. Enter the following values and use defaults for remaining fields: System Select Managed Element defined for Communication Manager Evolution Server. Profile Type Select Endpoint. Use Existing Endpoints Leave unchecked to automatically create new endpoint when a new user is created. Else, enter if endpoint is already defined. Extension Enter same extension number used for Login Name in Step 1. Template Select template 9640SIP_DEFAULT_CM_6_3. Security Code Enter numeric value used to register the SIP endpoint. Note: this field should match the value entered for the Communication Profile Password field in Step 2. Port Select IP from drop down menu. Voice Mail Number Enter Pilot Number for Avaya Modular Messaging or Avaya Aura Messaging if installed. Else, leave field blank. Delete Station on Unassign Enter to automatically delete station when of Endpoint from User or un-assigned from user [Optional]. on Delete User The screen below shows the results from Step 5 when adding a new SIP user in the sample configuration. 31 of 48

32 Click Endpoint Editor. Navigate to Feature Options tab. Within the IP Video as outlined below. Default values can be used for the remaining fields. section, check Click Done (not shown) to save definition of the endpoint. Click Commit (not shown) to save definition of the new user. Repeat the steps noted within this section to create a user for Avaya Flare Experience on Windows endpoint. See Reference [13] in Section 9 for further information. The sample configuration uses extension for Avaya Flare Experience on Windows SIP endpoint. 32 of 48

33 Step 6: 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. 33 of 48

34 6. Administer Scopia XT5000 Endpoint This section describes steps needed to configure and connect Scopia XT5000 Endpoint to Session Manager. These instructions assume other administration activities have already been completed such as firmware installation, network configuration and applicable licensing in place. For more information regarding how to configure Scopia XT5000 Endpoint, see Reference [15] and [16] in Section 9. The following administration activities will be described: SIP Protocol configuration Note: Some administration screens have been abbreviated for clarity. Configuration is accomplished by accessing the browser-based GUI of Scopia XT5000 Endpoint, using the URL where <ip-address> is the IP address of Scopia XT5000 Endpoint. Log in with the appropriate credentials. 34 of 48

35 Navigate to Administrator Settings Protocols SIP Enter the following values and use defaults for remaining fields: User Enter the SIP User Name created Section 5.7 Authentication Name Enter the SIP User Extension created Section 5.7 Authentication Password Enter the SIP User Extension password created in Section 5.7 Listening Port Retain the default value of Transport Outbound Call Retain the default value of TCP. Use SIP Registrar Select Yes from the drop down list Register DNS Name Enter the IP Address of the signaling interface of Session Manager as used in Section Use SIP Proxy Select Yes from the drop down list Proxy DNS Name Enter the IP Address of the signaling interface of Session Manager as used in Section Proxy Model Select Avaya from the drop down list Click SAVE to save definition of the SIP Protocol configuration. 35 of 48

36 7. Verification Steps 7.1. Verify Avaya Aura Session Manager Configuration Step 1: Verify Avaya Aura Session Manager is Operational Expand Elements Session Manager and select Dashboard to verify the overall system status of Session Manager. Specifically, verify the status of the following fields as shown below: Tests Pass Security Module Service State Data Replication 36 of 48

37 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. 37 of 48

38 Step 3: Verify Registrations of SIP Endpoints Navigate to Elements Session Manager System Status User Registrations to verify the SIP endpoints have successfully registered with Session Manager. 38 of 48

39 7.2. Verify Avaya Aura Communication Manager Configuration Verify the status of the SIP Trunk group by using the status trunk x command, where x is trunk group administered in Section Verify the trunk group is in-service/idle as shown below: status trunk 10 Page 1 TRUNK GROUP STATUS Member Port Service State Mtce Connected Ports Busy 0010/001 T00001 in-service/idle no 0010/002 T00002 in-service/idle no 0010/003 T00003 in-service/idle no 0010/004 T00004 in-service/idle no 0010/005 T00005 in-service/idle no 0010/006 T00006 in-service/idle no 0010/007 T00007 in-service/idle no 0010/008 T00008 in-service/idle no 0010/009 T00009 in-service/idle no 0010/010 T00010 in-service/idle no 0010/011 T00011 in-service/idle no 0010/012 T00012 in-service/idle no 0010/013 T00013 in-service/idle no 0010/014 T00014 in-service/idle no Verify the status of the SIP signaling group by using the status signaling-group x command, where x is the signaling group number administered in Section Verify the signaling group is in-service as indicated in the Group State: field shown below: status signaling-group 10 STATUS SIGNALING GROUP Group ID: 10 Group Type: sip Group State: in-service Use the SAT command, list trace tac #, where tac # is the trunk access code for the trunk group defined in Section 4.12 to trace trunk group activity for the SIP trunk between Avaya Aura Session Manager and Avaya Aura Communication Manager. For example, the trace below illustrates a call from a Scopa XT5000 SIP endpoint using extension to an Avaya Flare Experience on Windows SIP endpoint on extension over trunk group of 48

40 Avaya Flare Experience on Windows Endpoint answers the inbound call. Avaya Flare Experience on Windows SIP Endpoint then un-mutes video. Both parties now have audio and video call established. Avaya Flare Experience on Windows SIP Endpoint terminates the call. list trace tac #010 Page 1 LIST TRACE time data 11:42:26 TRACE STARTED 03/28/2013 CM Release String cold :42:33 SIP<INVITE sip:51015@dr.avaya.com:5060;transport=tcp SIP/2. 11:42:33 SIP<0 11:42:33 Call-ID: 64ba c :42: :42:33 dial 51015# 11:42:33 term station cid 0x10d5 11:42:33 SIP>INVITE sip:51015@dr.avaya.com SIP/2.0 11:42:33 Call-ID: d396e21a2b55155b8cc00 11:42:33 SIP<INVITE sip:51015@dr.avaya.com SIP/2.0 11:42:33 Call-ID: d396e21a2b55155b8cc00 11:42:33 SIP>INVITE sip:51015@dr.avaya.com SIP/2.0 11:42:33 Call-ID: d396e21a2b55155b8cc00 11:42:33 SIP<SIP/ Trying 11:42:33 Call-ID: d396e21a2b55155b8cc00 11:42:33 SIP>SIP/ Trying 11:42:33 Call-ID: d396e21a2b55155b8cc00 11:42:33 SIP<SIP/ Ringing 11:42:33 Call-ID: d396e21a2b55155b8cc00 11:42:33 SIP>SIP/ Ringing 11:42:33 Call-ID: d396e21a2b55155b8cc00 11:42:33 SIP>SIP/ Ringing 11:42:33 Call-ID: 64ba c :42: :42:38 SIP<SIP/ OK 11:42:38 Call-ID: d396e21a2b55155b8cc00 11:42:38 SIP>SIP/ OK 11:42:38 Call-ID: d396e21a2b55155b8cc00 11:42:38 SIP>PUBLISH sips:51023@dr.avaya.com SIP/2.0 11:42:38 Call-ID: 09e7a6bd396e21a5b55155b8cc00 11:42:38 SIP>PUBLISH sips:51015@dr.avaya.com SIP/2.0 11:42:38 Call-ID: 09e7a6bd396e21a8b55155b8cc00 11:42:38 SIP>SIP/ OK 11:42:38 Call-ID: 64ba c :42: :42:38 active station cid 0x10d5 11:42:38 SIP<SIP/ OK 11:42:38 Call-ID: 09e7a6bd396e21a5b55155b8cc00 11:42:38 SIP<ACK sip:51015@ :5061;transport=tls;epv=%3csip 11:42:38 SIP<:51015%40dr.avaya.com%3bgr%3d fb1f654aaf5cade6fa 11:42:38 SIP<27db7e%3e SIP/2.0 11:42:38 Call-ID: 64ba c :42: :42:38 SIP>ACK sip:51015@ :5061;transport=tls;gsid= :42:38 SIP>cd60-97cd-11e2-96d a2 SIP/2.0 11:42:38 Call-ID: d396e21a2b55155b8cc00 40 of 48

41 11:42:38 SIP<SIP/ OK 11:42:38 Call-ID: 09e7a6bd396e21a8b55155b8cc00 11:42:38 SIP<INFO 11:42:38 SIP<p:51015%40dr.avaya.com%3bgr%3d fb1f654aaf5cade6f 11:42:38 SIP<a27db7e%3e SIP/2.0 11:42:38 Call-ID: 64ba c :42: :42:38 SIP>SIP/ OK 11:42:38 Call-ID: 64ba c :42: :42:38 SIP>INFO 11:42:38 SIP>8cd60-97cd-11e2-96d a2 SIP/2.0 11:42:38 Call-ID: d396e21a2b55155b8cc00 11:42:38 SIP<INFO 11:42:38 SIP<p:51015%40dr.avaya.com%3bgr%3d fb1f654aaf5cade6f 11:42:38 SIP<a27db7e%3e SIP/2.0 11:42:38 Call-ID: 64ba c :42: :42:38 SIP>SIP/ OK 11:42:38 Call-ID: 64ba c :42: :42:38 SIP>INFO 11:42:38 SIP>8cd60-97cd-11e2-96d a2 SIP/2.0 11:42:38 Call-ID: d396e21a2b55155b8cc00 11:42:42 SIP<INVITE 11:42:42 SIP<sip:51023%40dr.avaya.com%3bgr%3dce6a196e727f2e3e448d62f 11:42:42 SIP<38a8d9d58%3e SIP/2.0 11:42:42 Call-ID: d396e21a2b55155b8cc00 11:42:42 SIP>SIP/ Trying 11:42:42 Call-ID: d396e21a2b55155b8cc00 11:42:42 SIP>INVITE 11:42:42 SIP>58cd60-97cd-11e2-96d a2 SIP/2.0 11:42:42 Call-ID: 64ba c :42: :42:42 SIP>SIP/ OK 11:42:42 Call-ID: d396e21a2b55155b8cc00 11:42:42 SIP<ACK 11:42:42 SIP<:51023%40dr.avaya.com%3bgr%3dce6a196e727f2e3e448d62f38a 11:42:42 SIP<8d9d58%3e SIP/2.0 11:42:42 Call-ID: d396e21a2b55155b8cc00 11:42:42 SIP>ACK 11:42:42 SIP>d60-97cd-11e2-96d a2 SIP/2.0 11:42:42 Call-ID: 64ba c :42: :42:48 SIP<BYE 11:42:48 SIP<:51023%40dr.avaya.com%3bgr%3dce6a196e727f2e3e448d62f38a 11:42:48 SIP<8d9d58%3e SIP/2.0 11:42:48 Call-ID: d396e21a2b55155b8cc00 11:42:48 SIP>SIP/ OK 11:42:48 Call-ID: d396e21a2b55155b8cc00 11:42:48 SIP>BYE 41 of 48

42 The following screen shows the corresponding SIP trace output from the perspective of Session Manager. Login to the CLI interface of Session Manager and launch the tracesm utility with SIP and Call Processiong options enabled CM4_PROCR SM :30: INVITE--> (8) T:51015 U: :30: <--Trying-- (8) 100 Trying 11:30:14,152 Remote host is not trusted Host not trusted 11:30:14,152 Originating Location found Location: :30: <--Proxy A- (8) 407 Proxy Authentication Required 11:30: ACK---> (8) 11:30: INVITE--> (8) T:51015 U: :30:14,198 Remote host is not trusted Host not trusted 11:30: <--Trying-- (8) 100 Trying 11:30:14,199 Originating Location found Location: :30:14,199 Try routing to determine if eme Location: :30:14,199 Request Dial Pattern route for: Location: :30:14,199 Dial Pattern route parameters URI Domain: dr.avaya.com Location: :30:14,199 Dial Pattern route parameters URI Domain: avaya.com Location: :30:14,199 Dial Pattern route parameters URI Domain: null Location: :30:14,199 Dial Pattern route parameters URI Domain: dr.avaya.com Location: null 11:30:14,199 Dial Pattern route parameters URI Domain: avaya.com Location: null 11:30:14,199 Dial Pattern route parameters URI Domain: null Location: null 11:30:14,199 Dial Pattern found for: Pattern: 51 11:30:14,199 Route Policy found Pattern: 51 RoutePolicyList: to CM4_PROCR 11:30:14,199 Trying to Authenticate User: Realm: dr.avaya.com 11:30:14,200 Route found for: SIPEntity: CM4_PROCR 11:30:14,201 Request Adaptation Adapter: Presence Buddy List 11:30:14,201 Applied egress Adaptation 11:30:14,201 Routing SIP request SipEntity: CM4_PROCR EntityLink: silasm4->tls: :30:14,203 No hostname resolution required Routing to: sip: ;transport=tls;lr;phase=imsorig 11:30: INVITE--> (8) T:51015 F:51023 U:51015 P:imsorig 11:30: <--Trying-- (8) 100 Trying 11:30: <--PUBLISH- (9) 11:30:14,216 Request Adaptation Adapter: Presence Buddy List 11:30:14,216 Applied ingress Adaptation 11:30:14,216 Originating Location found Location: of 48

43 11:30: OK--> (9) 200 OK (PUBLISH) 11:30: <--INVITE-- (10) T:51015 F:51023 U:51015 P:origdone 11:30: Trying--> (10) 100 Trying 11:30:14,227 Request Adaptation Adapter: Presence Buddy List 11:30:14,228 Applied ingress Adaptation 11:30:14,228 Originating Location found Location: :30:14,228 Try routing to determine if eme Location: :30:14,228 Request Dial Pattern route for: Location: :30:14,228 Dial Pattern route parameters URI Domain: dr.avaya.com Location: :30:14,228 Trying Dial Pattern route Domain: dr.avaya.com Location: :30:14,228 Dial Pattern route parameters URI Domain: avaya.com Location: :30:14,228 Dial Pattern route parameters URI Domain: null Location: :30:14,228 Trying Dial Pattern route Domain: null Location: :30:14,228 Dial Pattern route parameters URI Domain: dr.avaya.com Location: null 11:30:14,228 Trying Dial Pattern route Domain: dr.avaya.com Location: null 11:30:14,228 Dial Pattern route parameters URI Domain: avaya.com Location: null 11:30:14,228 Dial Pattern route parameters URI Domain: null Location: null 11:30:14,228 Trying Dial Pattern route Domain: null Location: null 11:30:14,228 Dial Pattern found for: Pattern: 51 11:30:14,228 Route Policy found Pattern: 51 RoutePolicyList: to CM4_PROCR 11:30:14,229 Route found for: SIPEntity: CM4_PROCR 11:30:14,229 Entity Link found SIPEntity: CM4_PROCR EntityLink: silasm4->tls, bidirid=null, deny=false: :30:14,230 Request Adaptation Adapter: Presence Buddy List 11:30:14,230 Applied egress Adaptation 11:30:14,230 Routing SIP request SipEntity: CM4_PROCR EntityLink: silasm4->tls: :30:14,232 No hostname resolution required Routing to: sip: ;transport=tls;lr;phase=imsterm 11:30: INVITE--> (10) T:51015 F:51023 U:51015 P:imsterm 11:30: <--INVITE-- (10) T:51015 F:51023 U:51015 P:termdone 11:30: Trying--> (10) 100 Trying 11:30:14,239 Request Adaptation Adapter: Presence Buddy List 11:30:14,239 Applied ingress Adaptation 11:30:14,239 Originating Location found Location: :30: <--Trying-- (10) 100 Trying 11:30:14,240 Try routing to determine if eme Location: :30:14,240 Request Dial Pattern route for: Location: :30:14,240 Dial Pattern route parameters URI Domain: dr.avaya.com Location: :30:14,240 Trying Dial Pattern route Domain: dr.avaya.com Location: :30:14,240 Dial Pattern route parameters URI Domain: avaya.com Location: of 48