Avaya 1000 Service Guide

Configuring Avaya 10x0 Series SIP Video Endpoints with Avaya Aura
TM
Aura
Communication Manager Feature Server Release
TM
Session Manager Release 6.0 and Avaya
6.0 – Issue 1.0
Abstract
These Application Notes describe the configuration of the Avaya 10x0 Series SIP Video Endpoints with Avaya AuraTM Session Manager and Avaya AuraTM Communication Manager as a Feature Server.
Avaya AuraTM Session Manager provides SIP proxy/routing functionality, routing SIP
sessions across a TCP/IP network with centralized routing policies and registrations for SIP endpoints.
Avaya AuraTM Communication Manager operates as a Feature Server for the SIP
endpoints which communicate with Avaya AuraTM Session Manager over SIP trunks.
These Application Notes provide information for the setup, configuration, and verification of the call flows tested on this solution.
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Table of Contents:
1. Introduction ............................................................................................................. 4
1.1.Equipment and Software Validated......................................................................... 5
2. Configuring Avaya AuraTM Communication Manager Feature Server ..................... 5
2.1.Verify System Capabilities and Licensing ............................................................... 5
2.1.1.SIP Trunk Capacity Check ................................................................................. 6
2.1.2.AAR/ARS Routing Check ................................................................................... 6
2.1.3.Enable Private Numbering .................................................................................. 7
2.2.Add Node Name of Avaya AuraTM Session Manager ............................................. 7
2.3.Configure Codec Type ............................................................................................ 8
2.4.Configure IP Network Region ................................................................................. 8
2.5.Add SIP Signaling Group ........................................................................................ 9
2.6.Add SIP Trunk Group ........................................................................................... 10
2.7.Administering Numbering Plan ............................................................................. 11
2.8.Configure Stations ................................................................................................ 12
2.9.Configure Off-PBX-Telephone Station-Mapping ................................................... 14
2.10.Save Translations ................................................................................................. 14
3. Configure Avaya Aura™ Session Manager .......................................................... 14
3.1.Administer SIP Domains ....................................................................................... 15
3.2.Define Locations ................................................................................................... 16
3.3.Add Avaya AuraTM Communication Manager Feature Server ............................... 17
3.3.1.Define a SIP Element for Avaya AuraTM Communication Manager Feature
Server ............................................................................................................... 17
3.3.2.Define Element Link for Avaya AuraTM Communication Manager Feature Server
......................................................................................................................... 18
3.3.3.Define Routing Policy for Avaya AuraTM Communication Manager Feature
Server ............................................................................................................... 18
3.3.4.Define Applications for Avaya AuraTM Communication Manager Feature Server
......................................................................................................................... 19
3.3.5.Define Application Sequences for Avaya AuraTM Communication Manager
Feature Server ................................................................................................. 20
3.3.6.Define Avaya AuraTM Communication Manager Feature as an Administrable
Entity ................................................................................................................ 21
3.3.7.Add SIP Users .................................................................................................. 23
4. Configure Avaya 10x0 Video Endpoint ................................................................. 27
5. Verification Steps .................................................................................................. 34
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5.1.Verify Avaya Aura
TM
Session Manager Configuration ........................................... 34
5.1.1.Verify Avaya AuraTM Session Manager is Operational ..................................... 34
5.1.2.Verify SIP Link Status ....................................................................................... 36
5.1.3.Verify Registrations of SIP Endpoints ............................................................... 37
5.2.Verify Avaya AuraTM Communication Manager Feature Server Configuration ...... 39
5.3.Call Scenarios Verified ......................................................................................... 42
6. Acronyms .............................................................................................................. 43
7. Conclusion ............................................................................................................ 43
8. Additional References ........................................................................................... 43
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1. Introduction
These Application Notes present a sample configuration for a network that uses Avaya Aura™ Session Manager to support registration of Avaya 10x0 (1010, 1020, 1030, 1040, and 1050) SIP Video endpoints and enables connectivity to Avaya Aura™ Communication Manager Feature Server 6.0 using SIP trunks.
As shown in Figure 1, Avaya Aura™ Session Manager is managed by Avaya Aura™ System Manager. Avaya 10x0 Video Endpoints configured as SIP endpoints utilize the Avaya Aura™ Session Manager User Registration feature and require Avaya Aura™ Communication Manager operating as a Feature Server. Communication Manager Feature Server only supports IP Multimedia Subsystem (IMS)-SIP users that are registered to Avaya Aura™ Session Manager. The Communication Manager Feature Server is connected to Session Manager via an IMS­enabled SIP signaling group and associated SIP trunk group.
For the sample configuration, Avaya Aura™ Session Manager runs on an Avaya S8510 Server. Avaya Aura™ Communication Manager 6.0 Feature Server runs on a S8800 server with an Avaya 450 Gateway. The results in these Application Notes should be applicable to other Avaya servers and media gateways that support Avaya Aura™ Communication Manager 6.0.
These Application Notes will focus on the configuration of the Communication Manager Feature Server and Session Manager. Detailed administration of Communication Manager Evolution Server will not be described (see the appropriate documentation listed in Section 8).
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Figure 1 – Sample Configuration
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1.1. Equipment and Software Validated
The following equipment and software were used for the sample configuration.
Equipment Software
Avaya AuraTM Session Manager Release 6.0.0.0.600020 Avaya AuraTM System Manager Release 6.0 Load: 6.0.0.0.11 Avaya AuraTM Communication Manager
• Avaya S8800 Server Feature Server Avaya IP Telephones10x0 Video Endpoints (SIP):
• 1020
• 1030
• 1040
Release R016x.00.0.345.0-2350
FW: AV_PP1_4.7.0 (19)2.0 FW:3.0 AV_EX2_4.7.0 (19) FW:1.5 AV_TM2_4.7.0 (19)
2. Configuring Avaya AuraTM Communication Manager
Feature Server
This section describes the administration of Communication Manager Feature Server using a System Access Terminal (SAT). Alternatively, some of the station administration could be performed using the Communication System Management application on System Manager. These instructions assume the G450 Media Gateway is already configured on the Communication Manager Feature Server. Some administration screens have been abbreviated for clarity.
Verify System Capabilities and Communication Manager Licensing Administer IP node names Administer codec type Administer IP network region Administer SIP signaling group Administer SIP trunk group Administer numbering plan Administer station endpoints Administer off-pbx-telephone station-mapping Save translations
After completing these steps, the “save translation” command should be performed
.
2.1. Verify System Capabilities and Licensing
This section describes the procedures to verify the correct system capabilities 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.
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2.1.1. SIP Trunk Capacity Check
Issue the display system-parameters customer-options command to verify that an adequate number of SIP trunk members are licensed 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: 12000 0 Maximum Concurrently Registered IP Stations: 18000 0 Maximum Administered Remote Office Trunks: 12000 0 Maximum Concurrently Registered Remote Office Stations: 18000 0 Maximum Concurrently Registered IP eCons: 414 0 Max Concur Registered Unauthenticated H.323 Stations: 100 0 Maximum Video Capable Stations: 18000 0 Maximum Video Capable IP Softphones: 18000 0 Maximum Administered SIP Trunks: 24000 128 Maximum Administered Ad-hoc Video Conferencing Ports: 24000 50 Maximum Number of DS1 Boards with Echo Cancellation: 522 0 Maximum TN2501 VAL Boards: 128 0 Maximum Media Gateway VAL Sources: 250 0 Maximum TN2602 Boards with 80 VoIP Channels: 128 0 Maximum TN2602 Boards with 320 VoIP Channels: 128 0 Maximum Number of Expanded Meet-me Conference Ports: 300 0
(NOTE: You must logoff & login to effect the permission changes.)
2.1.2. AAR/ARS Routing Check
Verify that ARS is enabled (on page 3 of system-parameters customer options)
display system-parameters customer-options Page 3 of 11 OPTIONAL FEATURES
A/D Grp/Sys List Dialing Start at 01? n CAS Main? n Answer Supervision by Call Classifier? n 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?
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2.1.3. Enable Private Numbering
Use the “change system-parameters customer-options” command to verify that Private Networking is enabled as shown below:
display system-parameters customer-options Page 5 of 11 OPTIONAL FEATURES
Multinational Locations? y Station and Trunk MSP? y Multiple Level Precedence & Preemption? n Station as Virtual Extension? y Multiple Locations? y System Management Data Transfer? n Personal Station Access (PSA)? y Tenant Partitioning? n PNC Duplication? n Terminal Trans. Init. (TTI)? y Port Network Support? n Time of Day Routing? n Posted Messages? n TN2501 VAL Maximum Capacity? y Uniform Dialing Plan? y Private Networking? y Usage Allocation Enhancements? y Processor and System MSP? y Processor Ethernet? y Wideband Switching? n Wireless? y
2.2. Add Node Name of Avaya AuraTM Session Manager
Using the change node-names ip command, add the node-name and IP for the Session Manager’s software asset, if not previously added.
change node-names ip Page 1 of 2 IP NODE NAMES Name IP Address default 0.0.0.0 procr 135.9.88.72 procr6 ::
silasm4 135.9.88.62
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2.3. Configure Codec Type
Issue the change ip-codec-set n command where “n” is the next available number. Enter the following values:
Enter “G.711MU” and “G.729” as supported types of Audio Codecs Silence Suppression: Retain the default value “n”. Frames Per Pkt: Enter 2”. Packet Size (ms): Enter “20”. Media Encryption: Enter the value based on the system requirement. For the sample
configuration “none” was used.
change ip-codec-set 1 Page 1 of 2 IP Codec Set
Codec Set: 1
Audio Silence Frames Packet Codec Suppression Per Pkt Size(ms) 1: G.711MU n 2 20 2: G.729 n 2 20
3:
Media Encryption 1: none
2.4. Configure IP Network Region
Using the change ip-network-region 1 command, set the Authoritative Domain. For the sample configuration “dr.avaya.com” was used. Verify the Intra-region IP-IP Direct Audio, and Inter-region IP-IP Direct Audio fields are set to yes.
change ip-network-region 1 Page 1 of 19 IP NETWORK REGION Region: 1 Location: 1 Authoritative Domain: dr.avaya.com Name: CMFS-Video MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 1 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? n UDP Port Max: 16585
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2.5. Add SIP Signaling Group
Issue the add signaling-group n command, where “n” is an available signaling group number, for one of the SIP trunks to the Session Manager, and fill in the indicated fields. In the sample configuration, trunk group “1” and signaling group “1” were used to connect to Avaya Aura™ Session Manager. Default values can be used for the remaining fields.
Group Type: “sip” Transport Method: ”tcp” IMS Enabled?: “y” IP Video?: “y” Peer Detection Enabled?: “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: procr from Section 2.2Far-end Node Name: Session Manager node name from Section 2.2 Near-end Listen Port: “5060”Far-end Listen Port: “5060”Far-end Domain: Authoritative Domain from Section 2.4 Enable Layer 3 Test: “y”Direct IP-IP Early Media?: “y”
display signaling-group 1 Page 1 of 1 SIGNALING GROUP
Group Number: 1 Group Type: sip Transport Method: tcp IMS Enabled? y IP Video? y Priority Video? y Peer Detection Enabled? y Peer Server: SM
Near-end Node Name: procr Far-end Node Name: silasm4 Near-end Listen Port: 5060 Far-end Listen Port: 5060
Far-end Network Region: 1
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 Direct IP-IP Early Media? y H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6
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2.6. Add SIP Trunk Group
Add the corresponding trunk group controlled by this signaling group via the add trunk-group n command, where “n” is an available trunk group number and fill in the indicated fields.
Group Type: sipGroup Name: A descriptive name. TAC: An available trunk access code. Service Type: “tie”Signaling Group: The number of the signaling group added in Section 2.5 Number of Members: The number of SIP trunks to be allocated to calls
routed to Session Manager (must be within the limits of the total number of trunks configured in Section 2.1.1).
add trunk-group 1 Page 1 of 21 TRUNK GROUP
Group Number: 1 Group Type: sip CDR Reports: y Group Name: SIP Video TG to silasm4 COR: 1 TN: 1 TAC: #001 Direction: two-way Outgoing Display? y Dial Access? n Night Service: Queue Length: 0 Service Type: tie Auth Code? n
Signaling Group: 1
Number of Members: 64
Once the add command is completed, trunk members will be automatically generated based on the value in the Number of Members field.
On Page 2, set the Preferred Minimum Session Refresh Interval to 1200. Note: to avoid extra SIP messages, all SIP trunks connected to Session Manager should be configured with a minimum value of 1200.
add trunk-group 1 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
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On Page 3, set Numbering Format to be “private”. Use default values for all other fields.
add trunk-group 1 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
2.7. Administering Numbering Plan
SIP Users registered to Session Manager needs to be added to either the private or public numbering table on the Communication Manager Feature Server. For the sample configuration, private numbering was used and all 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 References [3] and [8] in Section 8.
To enable SIP endpoints to dial extensions defined in the Communication Manager Feature Server, use the “change private-numbering x” command, where x is the number used to identify the private number plan. For the sample configuration, extension numbers starting with 5-XXXX are used on the Communication Manager Feature Server.
Ext Len: Enter the extension length allowed by the dial plan Ext Code: Enter leading digit (s) from extension number Trunk Grp: Enter the SIP Trunk Group number for the SIP trunk
between the Feature Server and Session Manager
Private Prefix: Leave blank unless an enterprise canonical numbering
scheme is defined in Session Manager. If so, enter the appropriate prefix.
change private-numbering 1 Page 1 of 2 NUMBERING - PRIVATE FORMAT
Ext Ext Trk Private Total Len Code Grp(s) Prefix Len
5 5 1 5 Total Administered: 1 Maximum Entries: 540
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2.8. Configure Stations
The method is the same for administering all of the Avaya 1000 series video endpoints with the exception of the 1040 and 1050’s. The only difference is that the 1040 can be administered to have up to 3 call appearances and the 1050 can have up to 7 call appearances for conferencing via their internal MCU’s. The 1010, 1020, and 1030 have to be administered with only one call­appearance since they are a single-line endpoint with no conferencing or transferring capabilities.
For each SIP user to be defined in Session Manager, add a corresponding station on the Communication Manager Feature Server. Note: instead of manually defining each station using the Communication Manager SAT interface, the preferred option is to automatically generate the SIP station when adding a new SIP user. See Section 3.3.6 for more information on adding SIP users.
The phone number defined for the station will be the number the SIP user enters to register to Session Manager. Use the “add station x” command where x is a valid extension number defined in the system. In this example extension 55002 is an Avaya 1020 video endpoint. On page 1 of the change station form:
Phone Type: Set to 9630SIP Name: Display name for user  Security Code: Number used when user logs into station. Note: this code
should match the “Shared Communication Profile Password” field defined when adding this user in Session Manager. See Section 3.3.6. IP Video? Enable endpoint for video
add station 55002 Page 1 of 6
STATION
Extension: 55002 Lock Messages? n BCC: 0 Type: 9630SIP Security Code: 123456 TN: 1 Port: S00006 Coverage Path 1: 1 COR: 1 Name: SIL Video Lab - 1020 Coverage Path 2: COS: 1 Hunt-to Station: STATION OPTIONS Time of Day Lock Table: Loss Group: 19 Message Lamp Ext: 55002 Display Language: english Button Modules: 0 Survivable COR: internal Survivable Trunk Dest? y IP SoftPhone? n
IP Video? y
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Note: It is important to assign only one call-appearance for the 1010, 1020, and 1030’s.
add station 55002 Page 4 of 6
STATION SITE DATA Room: Headset? n Jack: Speaker? n Cable: Mounting: d Floor: Cord Length: 0 Building: Set Color:
ABBREVIATED DIALING List1: List2: List3:
BUTTON ASSIGNMENTS 1: call-appr 5: 2: 6: 3: 7: 4: 8:
On page 6, set:
SIP Trunk option: Enter SIP Trunk Group defined in Section 2.6
change station 55002 Page 6 of 6
STATION SIP FEATURE OPTIONS Type of 3PCC Enabled: None SIP Trunk: 1
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2.9. Configure Off-PBX-Telephone Station-Mapping
Use the “change off-pbx-telephone station-mapping” command for each extension associated with SIP users defined in Session Manager. On page 1, enter the SIP Trunk Group defined in Section 2.6 and use default values for other fields.
change off-pbx-telephone station-mapping 55002 Page 1 of 3
STATIONS WITH OFF-PBX TELEPHONE INTEGRATION
Station Application Dial CC Phone Number Trunk Config Dual Extension Prefix Selection Set Mode 55002 OPS - 55002 1 1
-
-
On Page 2, enter the following values:
Mapping Mode: “both” Calls Allowed: “all”
change off-pbx-telephone station-mapping 55002 Page 2 of 3
STATIONS WITH OFF-PBX TELEPHONE INTEGRATION
Station Appl Call Mapping Calls Bridged Location Extension Name Limit Mode Allowed Calls 55002 OPS 1 both all none
-
2.10. Save Translations
Configuration of Communication Manager Feature Server is complete. Use the “save translations” command to save these changes
Note: After a change on Communication Manager Feature Server which alters the dial plan,
synchronization between Communication Manager Feature Server and Session Manager needs to be completed and SIP phones must be re-registered. To request an on demand synchronization, log into the System Manager console and use the Synchronize CM Data feature under the Communication System Management menu.
3. Configure Avaya Aura™ Session Manager
This section provides the procedures for configuring the Session Manager and includes the following items:
Administer SIP domain Define Logical/Physical Locations that can be occupied by SIP Entities
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