Avaya 1000 Service Guide

Avaya Solution Interoperability Test Lab

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 2.2  Far-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: “sip”  Group 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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 For each SIP entity in the sample configuration:

o Define SIP Entity o Define Entity Links, which define the SIP trunk parameters used by Avaya

Aura ™ Session Manager when routing calls to/from SIP Entities

o Define Routing Policies, which control call routing between the SIP Entities o Define Dial Patterns, which govern to which SIP Entity a call is routed

 Define the Communication Manager Feature Server as an Managed Element  Adding SIP Endpoints/SIP URE users

Configuration is accomplished by accessing the browser-based GUI of Avaya Aura™ System Manager, using the URL “http://<ip-address>/SMGR”, where “<ip-address>” is the IP address of Avaya Aura™ System Manager.

Log in with the appropriate credentials and accept the Copyright Notice. Expand the Routing Link on the left side of Navigation Menu. Select a specific item such as Domains.

3.1. Administer SIP Domains

Expand Routing and select Domains.

 Click New  In the General Section, under Name add a descriptive name. Under Notes add a brief

description.

 Click Commit to save. The screen below shows the information for the sample configuration.

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3.2. Define Locations

Expand Routing and select Locations. Locations are used to identify logical and/or physical locations where SIP Entities reside, for purposes of bandwidth management or location-based routing.

 Click New  In the General Section, under Name add a descriptive name.  Under Notes add a brief description.  In the Location Pattern Section, under IP Address Pattern enter pattern used to

logically identify the location Under Notes add a brief description.

 Click Commit to save.

The screen below shows the information for Communication Manager Feature Server in the sample configuration.

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3.3. Add Avaya AuraTM Communication Manager Feature Server

The following section captures relevant screens for defining Avaya AuraTM Communication Manager Feature Server applicable for the sample configuration.

3.3.1. Define a SIP Element for Avaya AuraTM Communication Manager Feature Server

The following screen shows addition of Communication Manager Feature Server. The IP address used is that of the Processor Ethernet (procr) of the Avaya Communication Manager Feature server.

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3.3.2. Define Element Link for Avaya AuraTM Communication Manager Feature Server

The following screen shows the Element Link defined for the Avaya AuraTM Communication Manager Feature Server.

3.3.3. Define Routing Policy for Avaya AuraTM Communication Manager Feature Server

Since the SIP users are registered on Session Manager, a routing policy does not need to be defined for the Communication Manager Feature Server.

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3.3.4. Define Applications for Avaya AuraTM Communication Manager Feature Server

To define the Avaya AuraTM Communication Manager Feature Server Applications,

 Elements -> Session Manager, Application Configuration  Applications

o Click New (Not shown) o Under Name, enter a name for the Application entry o Under SIP Entity drop-down menu, select the appropriate SIP Entity. o Under CM System for SIP Entity, this field can be left as the default of Select

CM System.

o Under Description, enter a description if desired.

o Click Commit to save.

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3.3.5. Define Application Sequences for Avaya AuraTM Communication Manager Feature Server

To define the Avaya AuraTM Communication Manager Feature Server Application Sequences,

 Elements -> Session Manager, Application Configuration  Application

Sequences

o Click New (Not shown) o Under Name, enter a name of the Application Sequence. o Under Description, enter a description if desired. o Under Available Applications, select the Application that was created in

Section 3.3.4. The way to select the Application of choice is to click on the “+” symbol next to the Application desired. This will be added to the Applications in this Sequence list.

o Click Commit to save.

Second, define an Application Sequence for call application sequencing in the Avaya Aura Communication Manager Feature Server as shown below:

TM

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3.3.6. Define Avaya AuraTM Communication Manager Feature as an Administrable Entity

Before adding SIP users, the Avaya AuraTM Communication Manager Feature Server must also be added to System Manager as an administrable entity. This action allows System Manager to access Communication Manager over its administration interface similar to how other administration tools such as Avaya Site Administrator access Communication Manager. Using this administration interface, System Manager will notify the Communication Manager Feature Server when new SIP users are added.

To define the Avaya AuraTM Communication Manager Feature Server as an administrable entity,

 Elements -> Inventory -> Manage Elements

o Click New (Not shown) o Under Name, enter an identifier for the Communication Manager Feature

Server.

o Under Type drop-down menu, select CM. o Under Node, enter the IP address of the administration interface for the Feature

Server as shown below:

Defining the Avaya AuraTM Communication Manager Feature Server as an administrable entity (continued):

 Manage Elements - Attributes

o Under Login and Password, enter the login and password used for

administration access to the Feature Server.

o Select SSH access. o Under Port, enter the port number for the administration interface of 5022 as

shown below:

Defining the Avaya AuraTM Communication Manager Feature Server as an administrable entity (continued):

 Entities – Port  Entities – Access Point

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3.3.7. Add SIP Users

Add SIP users corresponding to the 96XX SIP stations defined in Section 2.8. Alternatively, use the option to automatically generate the SIP stations on Communication Manager Feature Server when adding a new SIP user.

 Expand Users

o Select Manage Users

 Click New

Step 1: Enter values for the following required attributes for a new SIP user in the General and Identity sections of the new user form.

 Last Name: enter last name of user  First Name: enter first name of user  Login Name: enter extension no.@sip domain

defined in Section 3.1. This field is primary handle of user.

 Authentication Type: select Basic  SMGR Login Password: enter password which will be used to log

into System Manager application (password). NOTE: This field is only displayed if adding a new user.

 Confirm Password: repeat value entered above. NOTE: This

field is only displayed if adding a new user.

 Shared Communication Profile Password: enter a numeric value which will

be used to logon to SIP phone. Note: this field must match the Security Code field on the station form defined in Section 2.8.

 Confirm Password: repeat numeric password

The screen below shows the information when adding a new SIP user to the sample configuration.

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Step 2: Scroll down to the Communication Profile section and Select New to define a Communication Profile for the new SIP user. Enter values for the following required

attributes:

 Name: enter name of communication profile  Default: enter checkmark to indicate this profile is default profile

Select New to define a Communication Address for the new SIP user. Enter values for the following required attributes:

 Type: select SIP  Handle: enter extension number  Domain: enter SIP domain defined in Section 3.1

Click Add (not shown) to save the Communication Address for the new SIP user.

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The screen below shows the information when adding a new SIP user to the sample configuration.

Step 3: Assign the Application Sequence defined in Section 3.3.4 to the new SIP user as part of defining the SIP Communication Profile. The Application Sequence can be used for both the originating and terminating sequence. Enter values for the following required attributes of the Station Profile section:  System: select the SIP Entity of the Communication Manager

Feature Server defined in Section 3.3.5 from menu

 Use Existing Stations: enter checkmark if station was already defined. Else, station will

automatically be created.

 Extension: enter extension number  Template: Select the template (system defined or user defined) you want to

associate with the endpoint. Select the template based on the set type you want to add.

 Security Code: enter numeric value which will be used to logon to SIP phone.

Note: this field must match the value entered for the Shared Communication Profile Password field

 Port: select port number from the list for the selected template

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The screen below shows the information when adding a new SIP user to the sample configuration.

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4. Configure Avaya 10x0 Video Endpoint

To administer the 10x0 video endpoints log in to the web interface using the IP address of the video endpoint. You will be redirected to a screen that looks similar to the one below. This is a sample configuration on how to administer a 10x0 video endpoint.

Step 1: Enter the proper login credentials and press Submit. Most of the Preferences can be customized to meet your needs. Mentioned below are the absolute necessary items that need to be administered to get the 10x0 up and running on the network.

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Once logged in select the Preferences tab and then the Network option.

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Select General option and enter values for the following required attributes.

 DHCP: enabled/disabled  IP Address: enter IP Address if DHCP is disabled  Subnet Mask: enter Subnet Mask if DHCP is disabled  Default Gateway: enter Default Gateway if DHCP is disabled  Hostname: enter the appropriate Hostname  DNS Servers: enter the appropriate DNS Servers  NTP Server Hostname: enter NTP Server Hostname

Select the Save Changes button to save the administration just added.

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Select the Preferences option and select the Communications option.

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Select the SIP option.

 SIP: enabled  SIP Username: enter the SIP Username for the device. NOTE: The SIP

Username should be unique and meaningful to the endpoint.

 Authorization Name: enter the SIP Server authorization username. NOTE: The

Authorization Name should be unique and meaningful to the endpoint.

 Authorization Password: enter the SIP Server authorization password  SIP Registration: select the communication path to use when registering with a

SIP Registrar

 SIP Proxy: choose ‘Enabled’ to use the SIP proxy  Proxy Hostname: enter the hostname or IP address of the SIP proxy server.

NOTE: This is the Session Manager software asset card IP address

 Proxy IP Port: enter the IP port number of SIP proxy server  SIP Registrar: choose ‘Enabled’ to use the SIP registrar  Registrar Hostname: enter the hostname or IP address of the SIP registrar server

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 Registrar IP Port: enter the IP port number of the SIP registrar server  UDP Signaling Port: enter the UDP port number of the SIP configuration  TCP Signaling: choose ‘Enable’ to use TCP for placing SIP call  TCP signaling Port: enter the TCP port number of the SIP configuration

Select the Save Changes button to save the administration just added.

Select the Preferences option again and select System.

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Select the Identification option. This option will allow the user to display the name and video/voice numbers on the menu bar.

 System Name: enter a descriptive name for the system  Video Number: enter the video number of the system  Voice Number: enter the voice number of the system

Select the Save Changes button to save the administration just added.

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5. Verification Steps

5.1. Verify Avaya Aura

5.1.1. Verify Avaya Aura

TM

Session Manager Configuration

TM

Session Manager is Operational

Verify the overall system status for Session Manager as shown below:

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Navigate to Elements  Session Manager  System Status  Security Module Status to view more detailed status information on the status of Security Module for Session Manager. Verify the Status column displays “Up” as shown below.

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5.1.2. Verify SIP Link Status

Expand the Session Manager menu on the left and click SIP Entity Monitoring. Verify all SIP Entity Links are operational as shown below:

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5.1.3. Verify Registrations of SIP Endpoints

Verify SIP users have been created in the Session Manager. In the sample configuration, Extension 55002 SIP user was created as shown in the highlighted area below:

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Verify the SIP endpoints have successfully registered with the Session Manager as shown below:

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5.2. Verify Avaya AuraTM Communication Manager Feature Server Configuration

Verify the status of the SIP trunk group by using the “status trunk n” command, where “n” is the trunk group number administered in Section 2.6. Verify that all trunks are in the “inservice/idle” state as shown below:

status trunk 1 TRUNK GROUP STATUS

Member Port Service State Mtce Connected Ports Busy 0001/001 T00001 in-service/idle no 0001/002 T00002 in-service/idle no 0001/003 T00003 in-service/idle no 0001/004 T00004 in-service/idle no 0001/005 T00005 in-service/idle no 0001/006 T00006 in-service/idle no 0001/007 T00007 in-service/idle no 0001/008 T00008 in-service/idle no 0001/009 T00009 in-service/idle no 0001/010 T00010 in-service/idle no

Verify the status of the SIP signaling groups by using the “status signaling-group n” command, where “n” is the signaling group number administered in Section 2.5 Verify the signaling group is “in-service” as indicated in the Group State field shown below:

status signaling-group 1 STATUS SIGNALING GROUP

Group ID: 1 Group Type: sip

Group State: in-service

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Use the Communication Manager SAT command, ‘list trace tac #’, where tac # is the trunk access code defined in Section 2.6 to trace trunk group activity for the SIP trunk between the Session Manager and the Communication Manager Feature Server as shown below:

list trace tac #001 Page 1 LIST TRACE time data

10:53:34 TRACE STARTED 08/21/2010 CM Release String cold-00.0.345.0-2350 10:54:29 SIP<INVITE sip:55001@dr.avaya.com;transport=tcp;user=ph 10:54:29 SIP<one SIP/2.0 10:54:29 dial 10:54:29 term trunk-group 1 cid 0xb9 10:54:29 dial 10:54:29 seize trunk-group 1 member 21 cid 0xb9 10:54:29 Calling Number & Name NO-CPNumber NO-CPName 10:54:29 Proceed trunk-group 1 member 21 cid 0xb911:01:07 Setup 10:54:29 SIP>SIP/2.0 180 Ringing 10:54:29 Alert trunk-group 1 member 21 cid 0xb9 10:54:31 active trunk-group 1 member 21 cid 0xb9 10:54:31 G711MU ss:off ps:20 rgn:1 [135.9.88.216]:60656 rgn:1 [135.9.88.174]:60142 10:54:31 G711MU ss:off ps:20 rgn:1 [135.9.88.174]:60142 rgn:1 [135.9.88.216]:60656 10:54:31 SIP>SIP/2.0 200 OK 10:54:31 Video: H264 [135.9.88.216]:60658 10:54:31 Video: H264 [135.9.88.174]:60144 logChl:110 sessId:2 bw:21760 tx/rx:11520 10:54:31 Video: H264 [135.9.88.174]:60144 10:54:31 Video: H264 [135.9.88.216]:60658 logChl:110 sessId:2 bw:21760 tx/rx:11520 10:54:31 SIP<ACK sip:55001@135.9.88.72;transport=tcp SIP/2.0 10:54:37 SIP<BYE sip:55001@135.9.88.72;transport=tcp SIP/2.0 10:54:37 SIP>SIP/2.0 200 OK 10:54:37 idle station 55002 cid 0xb9

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Use the Communication Manager SAT command, ‘list trace station xxx’, where xxx is the extension number of the 96XX SIP telephone as shown below:

list trace station 6663000 Page 1 LIST TRACE time data

10:58:38 TRACE STARTED 08/21/2010 CM Release String cold-00.0.345.0-2350 10:59:07 active station 55002 cid 0xbc 10:59:07 SIP>INVITE sip:55001@dr.avaya.com SIP/2.0 10:59:07 dial 10:59:07 term trunk-group 1 cid 0xbc 10:59:07 dial 10:59:07 seize trunk-group 1 member 23 cid 0xbc 10:59:07 Setup digits 55001 10:59:07 Calling Number & Name 55002 SIL Demo Vide 10:59:07 SIP<SIP/2.0 100 Trying 10:59:07 Proceed trunk-group 1 member 23 cid 0xbc 10:59:07 SIP<INVITE sip:55001@dr.avaya.com SIP/2.0 10:59:07 SIP>SIP/2.0 100 Trying 10:59:07 SIP>SIP/2.0 180 Ringing 10:59:07 SIP<SIP/2.0 180 Ringing 10:59:07 Alert trunk-group 1 member 23 cid 0xbc 10:59:09 SIP>SIP/2.0 200 OK 10:59:09 SIP<SIP/2.0 200 OK 10:59:09 active trunk-group 1 member 23 cid 0xbc 10:59:09 G711MU ss:off ps:20 rgn:1 [135.9.88.216]:60664 rgn:1 [135.9.88.174]:60150 10:59:09 G711MU ss:off ps:20 rgn:1 [135.9.88.174]:60150 rgn:1 [135.9.88.216]:60664 10:59:09 Video: H264 [135.9.88.216]:60666 10:59:09 Video: H264 [135.9.88.174]:60152 logChl:110 sessId:2 bw:21760 tx/rx:11520 10:59:09 Video: H264 [135.9.88.174]:60152 10:59:09 Video: H264 [135.9.88.216]:60666 logChl:110 sessId:2 bw:21760 tx/rx:11520 10:59:09 SIP>ACK sip:55001@135.9.88.72;transport=tcp SIP/2.0 10:59:10 SIP<ACK sip:55001@135.9.88.72;transport=tcp SIP/2.0 11:00:43 SIP>BYE sip:55002@135.9.88.72;transport=tcp SIP/2.0 11:00:43 SIP<BYE sip:55002@135.9.88.72;transport=tcp SIP/2.0 11:00:43 SIP>SIP/2.0 200 OK

11:00:43 idle trunk-group 1 member 23 cid 0xbc

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5.3. Call Scenarios Verified

Verification scenarios for the configuration described in these Application Notes included the following call scenarios:

Calls initiated from the GUI of the respective endpoint

 Place a point-to-point video call from a 1020/1030/1040 video endpoint registered to SM

(CMFS) to another 1020/1030/1040 video endpoint registered on SM (CMFS). Answer the call and verify two-way video and two-way talk path for all combinations of calls between10x0 video endpoints. Verify Call statistics on the endpoint GUI.

 Place a point-to-point video call from a 1040 video endpoint registered to SM (CMFS) to

another 1020/1030/1040 video endpoint registered on SM (CMFS). Answer the call and verify two-way video and talk path. Place a video conference call from 1040 to a 1020. Answer the call and verify three-way video and audio conference call. Add a fourth video endpoint to the call and verity video and audio. Verify Call statistics on the endpoint GUI.

 Place a point-to-point audio call from a 1020/1030/1040 video endpoint registered to SM

(CMFS) to another 1020/1030/1040 video endpoint registered on SM (CMFS). Answer the call and verify two-way talk path for all combinations of calls between10X0 video endpoints. Verify Call statistics on the endpoint GUI.

 Place a point-to-point audio call from a 1040 video endpoint registered to SM (CMFS) to

another 1020/1030/1040 video endpoint registered on SM (CMFS). Answer the call and verify two-way talk path. Place an audio conference call from 1040 to a 1020/1030/1040. Answer the call and verify talk path on conference call. Add a fourth video endpoint to the call and verify talk path. Verify Call statistics on the endpoint GUI.

Calls initiated from the Web interface of the respective endpoint