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1 Administering Avaya Video Conferencing Solution Advanced Topics Issue 1 Release 6.1 April 2012

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3 Contents Chapter 1: Overview of Avaya Video Conferencing Solution Components Avaya Aura Conferencing Release Avaya Video Conferencing Manager Configurations H.323 configuration SIP configuration H.323 and SIP configuration New in Release Release 6.1 administration requirements Required components and software Pre-administration tasks Required prior knowledge Related documents Chapter 2: Managing bandwidth allocation and administering bandwidth pools on Avaya Video Conferencing Solution Managing bandwidth allocation Class of Service Classifying users for COS Bandwidth pools Bandwidth pools for H.323 endpoints Bandwidth pools for SIP endpoints Bandwidth pools for H.323 and SIP endpoints Administering bandwidth pools Administering locations in System Manager Administering bandwidth pools through network regions in Communication Manager36 Chapter 3: Call Admission Control CAC interactions with Quality of Service CAC bandwidth calculation CAC implementation CAC interactions on Session Manager and Communication Manager Recommendations for CAC interactions on Session Manager and Communication Manager Location determination on Session Manager CAC media priority Engineering the Avaya Video Conferencing Solution site Adjusting bandwidth requirements Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

4 Contents Administering CAC for locations on Session Manager Disabling CAC alarms Disabling Session Manager from analyzing SDP headers Administering SIP entities to perform CAC processing Viewing bandwidth usage on System Manager Viewing real-time status of bandwidth usage on System Manager Viewing perfomance status of each location on System Manager Optimizing video on the network Deploying QoS Selecting the default maximum call rate Ensuring adequate inter-pbx network connectivity rates Ensuring adequate VPN connectivity rates Feature interactions and limitations Chapter 4: Administering conferencing bridges and trunks on Avaya Video Conferencing Solution Administering SIP trunks on Communication Manager Administering a video bridge on Communication Manager Administering a direct routing gatekeeper Administering the Polycom RMX SIP entity link on Session Manager Administering the Polycom RMX H.323 trunk to Communication Manager Administering the Polycom RMX H.323 trunk to Polycom CMA Administering the Polycom RMX video conferencing bridge Administering Polycom RMX for SIP and H.323 dual connectivity Administering Polycom RMX to support H.320 ISDN calls Administering the Polycom VBP-E video border proxy Administering Polycom VBP-E to support external video calls Chapter 5: Administering Communication Manager for video endpoints 83 Administering global settings on Communication Manager Enabling video on Communication Manager Administering audio codecs on Communication Manager Administering CAC on Communication Manager Administering Communication Manager for SIP endpoints Administering global settings for SIP endpoints on Communication Manager 93 Administering call routing for SIP endpoints on Communication Manager.. 94 Administering Avaya SIP video endpoints on System Manager Administering Communication Manager for H.323 endpoints Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

5 Contents Administering IP network regions on Communication Manager Administering Class of Service on Communication Manager Administering stations to support a Tandberg MXP series endpoint Administering Avaya one-x Communicator as an H.323 endpoint on Communication Manager Administering Polycom VSX series endpoints and Polycom HDX series endpoints on Communication Manager Chapter 6: Administering Avaya Aura Conferencing Logging in to Avaya Aura Conferencing Manager Adding elements to Avaya Aura Conferencing Administering links to components on Avaya Aura Conferencing Administering SIP on Avaya Aura Conferencing Administering an outbound proxy route on the Avaya Aura Conferencing bridge 106 Administering a SIP entity for Avaya Aura Conferencing on System Manager. 106 Administering call routing digits on Avaya Aura Conferencing Administering call branding on Avaya Aura Conferencing Administering a pattern on Avaya Aura Conferencing Manager to enable Avaya Aura Conferencing to recognize a URI Administering a pattern on Avaya Aura Conferencing Manager to enable Avaya Aura Conferencing to recognize a Telnum Administering Client Registration Server Administering Quality of Service on Avaya Aura Conferencing Administering Communication Manager for Avaya Aura Conferencing Administering conferencing on Avaya Aura Conferencing Administering ad hoc conferencing on Avaya Aura Conferencing Administering Avaya Desktop Video Device conferencing Administering Flex Call Flow on Avaya Desktop Video Device Chapter 7: Administering endpoints on Avaya Video Conferencing Solution115 Administering SIP endpoints Administering Avaya 1000 series endpoints Administering Avaya one-x Communicator Administering Avaya Desktop Video Device Administering the Polycom HDX endpoint on Session Manager Administering H.323 endpoints Administering Polycom VSX endpoint or Polycom HDX endpoint Administering the Polycom MGC series endpoint Administering the Tandberg MXP endpoint Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

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9 Chapter 1: Overview of Avaya Video Conferencing Solution Avaya Video Conferencing Solution provides the following video conferencing services in the Avaya Aura solution: Rich voice and video Content sharing Resilient and distributed network for the following components: - Media gateways - Analog, digital, IP-based communication devices, and SIP-based communication devices Avaya Aura solution includes the following applications: Avaya Aura System Manager: A central management system that delivers a common console across multiple products and a set of shared management services. Avaya Aura Session Manager: A SIP-routing-and-integration tool and the core component within the Avaya Aura Enterprise Edition solution. Avaya Aura Session Manager integrates all SIP and non SIP devices across the entire enterprise network within a company. Avaya Aura Communication Manager administered as Evolution Server: A gateway that provides Communication Manager features to SIP and non SIP endpoints. Avaya Aura Presence: A service that conveys the ability and willingness of a user to communicate across a set of services, such as telephony and instant messaging. SIP and H.323 video endpoints: The desktop and conference room devices that deliver audio and video. Components Avaya Video Conferencing Solution consists of the following key components: Gatekeeper provides call admission control (CAC) services and bandwidth management services for SIP video calls, H.323 video calls and video telephony features. Video border proxy provides combined services of a gatekeeper and gateway for SIP video calls and H.323 video calls. Gateways provide SIP-to-H.323 video call interworking services. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

10 Overview of Avaya Video Conferencing Solution Conferencing bridges support content sharing between video endpoints. Conferencing bridges are also called Multimedia Conferencing Units (MCU). Conferencing scheduling and endpoint management applications manage and monitor video endpoints. Endpoints are devices on desks of users and devices in conference rooms which provide audio and video. The Avaya 1050 endpoint supports 8 conference participants, which includes the Avaya 1050 endpoint itself, but it displays a maximum of only 4 participants. The Avaya 1050 endpoint displays the last 4 participants who were involved in conducting presentations. For conferences with more than 4 participants, the Avaya 1050 endpoint changes the audio codec from wide-band to G.711. All key components of Avaya Video Conferencing Solution may not be installed in each customer specific Avaya Video Conferencing Solution installation. Table 1: Avaya Video Conferencing Solution components on page 10 lists the products that function as Avaya Video Conferencing Solution components. Table 1: Avaya Video Conferencing Solution components Component Product Gatekeeper Video border proxy Gateway Conferencing bridge Conferencing scheduling and endpoint management application Endpoint Communication Manager Session Manager System Manager Polycom CMA Polycom VBP-E Communication Manager administered as Evolution Server Polycom RMX Avaya Aura Conferencing Avaya 1040 endpoint Avaya 1050 endpoint Polycom RMX Avaya Video Conferencing Manager Polycom CMA Avaya 1000 series Avaya Desktop Video Device Avaya one-x Communicator Polycom HDX Polycom VSX 10 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

11 Components For more information on compatible versions of products in Avaya Video Conferencing Solution, see Avaya Video Conferencing Solution 6.1 Compatibility Matrix at: Avaya Aura Conferencing Release 6.0 Avaya Aura Conferencing Release 6.0 provides CIF video conferencing services with voice-activated switching. Avaya Aura Conferencing Release 6.0 supports a maximum of 100 video endpoints with a maximum of 500 participants in a conference call. The components of Avaya Aura Conferencing Release 6.0 provide the following functions for calls: Booking engines Account management utilities Data sharing function Billing outputs Directory server integration capabilities Audio management Video conferencing Avaya Video Conferencing Manager Avaya Video Conferencing Manager is a comprehensive management software solution for network administrators who manage video and voice communications systems in an IP environment. With Avaya Video Conferencing Manager, network administrators can perform the following tasks: Monitor and manage multivendor video and voice communications devices across the entire enterprise from a web browser. Manage and automate video and voice call scheduling. Generate reports to track inventory and to analyze device utilization and performance. For more information on installation and operation of Avaya Video Conferencing Manager, see Avaya Video Conferencing Manager Deployment Guide on the Avaya Support Web site at Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

12 Overview of Avaya Video Conferencing Solution Configurations H.323 configuration The Avaya Video Conferencing Solution H.323 configuration consists of the following components: Communication Manager functions as a gatekeeper. H.323 devices connect to the Avaya Aura core through this gatekeeper. Polycom CMA functions as a gatekeeper. Polycom RMX functions as an MCU. This MCU registers through Communication Manager or Polycom CMA. If Polycom RMX registers through Communication Manager, Polycom RMX can host ad hoc video conferences and make outgoing calls using E.164 aliases. If Polycom RMX registers through Polycom CMA, Polycom RMX supports encryption to devices registered to Polycom CMA, and Polycom RMX can make outgoing calls using IP addresses. Polycom VBP-E functions as a video border proxy. The video border proxy connects through Communication Manager and Polycom RMX. Polycom endpoints, Tandberg endpoints, and Avaya H.323 endpoints function as endpoints that provide audio and video services. Avaya Video Conferencing Solution H.323 configuration on page 13 illustrates the deployment of Avaya Video Conferencing Solution in an H.323 video configuration. 12 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

13 Configurations Figure 1: Avaya Video Conferencing Solution H.323 configuration SIP configuration The Avaya Video Conferencing Solution SIP configuration consists of the following components: Session Manager functions as SIP registrar, location server, and SIP router for SIP endpoints. SIP devices connect to the Avaya Aura core through Session Manager. System Manager manages Session Manager. Communication Manager administered as Evolution Server functions as the telephony feature engine for all endpoints. Polycom RMX functions as an MCU. Avaya Video Conferencing Manager schedules and manages voice and video communications systems in an IP environment. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

14 Overview of Avaya Video Conferencing Solution Avaya Aura Conferencing hosts audio and video conferences. Polycom and Avaya SIP endpoints function as endpoints that provide audio and video services. Avaya Video Conferencing Solution SIP configuration on page 14 illustrates the deployment of Avaya Video Conferencing Solution in a SIP configuration. Figure 2: Avaya Video Conferencing Solution SIP configuration H.323 and SIP configuration The Avaya Video Conferencing Solution H.323 and SIP configuration consists of the following components: Session Manager functions as SIP registrar, location server, and SIP router for SIP endpoints. SIP devices connect to the Avaya Aura core through Session Manager. System Manager manages Session Manager. 14 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

15 Configurations Communication Manager administered as Evolution Server functions as a gatekeeper for H.323 endpoints. H.323 devices connect to the Avaya Aura core through Session Manager. Polycom RMX functions as an MCU. Polycom RMX connects through a SIP entity link to Session Manager, and through an H.323 trunk to Communication Manager administered as Evolution Server. If Polycom RMX registers through Communication Manager administered as Evolution Server, it can host ad hoc video conferences, and make outgoing calls using E.164 aliases. Polycom CMA functions as a gatekeeper and scheduler for Polycom H.323 endpoints. Polycom VBP-E functions as a video border proxy for H.323 endpoints. Avaya Video Conferencing Manager schedules and manages voice and video communications systems in an IP environment. Avaya Aura Conferencing hosts audio and video conferences. Polycom endpoints, Tandberg endpoints, Avaya SIP endpoints, and Avaya H.323 endpoints function as endpoints that provide audio and video services. In the Avaya Video Conferencing Solution H.323 and SIP configuration, customers can specify various combinations of the Avaya Video Conferencing Solution components. Multivendor video devices can operate with Avaya video devices. Avaya Video Conferencing Solution H.323 and SIP configuration on page 16 illustrates the deployment of Avaya Video Conferencing Solution in an H.323 and SIP configuration. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

16 Overview of Avaya Video Conferencing Solution Figure 3: Avaya Video Conferencing Solution H.323 and SIP configuration New in Release 6.1 Avaya Video Conferencing Solution Release 6.1 enhancements provide the following capabilities to Avaya Aura for video communications: Communication Manager scalability to administer video capable IP endpoints, concurrent H.323 video calls, and 2000 concurrent SIP video calls Enhanced SIP-to-H.323 video gateway capability in Communication Manager administered as Evolution Server Support for Avaya 1000 series SIP video conferencing endpoints Support for Avaya one-x Communicator as a SIP video endpoint 16 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

17 Release 6.1 administration requirements Support for Polycom H.323 video endpoints on Polycom CMA configurations and Communication Manager administered as Evolution Server Support for SIP registration of Polycom endpoints to Session Manager Support for the People+Content feature over SIP for Avaya 1000 series endpoints, the Polycom HDX endpoint, and the Polycom RMX MCU. Release 6.1 administration requirements Required components and software Before administering the Avaya Video Conferencing Solution components, ensure that you have installed the following components and software: Avaya Aura Communication Manager Release Avaya Aura System Manager Avaya Aura Session Manager Avaya 1010 endpoint and Avaya1020 endpoint Avaya 1030 endpoint, Avaya 1040 endpoint, and Avaya 1050 endpoint Polycom RMX with MPM card Before you install video features on endpoints, you must administer Session Manager. For instructions on how to administer Session Manager, see Administering Avaya Aura Session Manager, , on the Avaya support Web site at You do not need to upgrade to Communication Manager Release 6.2 and Session Manager Release 6.2. Avaya Virtual Link Avaya Virtual Link is an optional component. Using Avaya Virtual Link, users can share content on the Avaya 1010 endpoint and the Avaya 1020 endpoint with other Avaya 1000 series endpoints. SIP or H.323 endpoints Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

18 Overview of Avaya Video Conferencing Solution Pre-administration tasks Ensure you have completed the following tasks: Connect all Avaya Aura components and endpoints to the network. Perform a network readiness test or network assessment to ensure the customer network is capable of supporting bandwidth demands of VoIP. Avaya recommends implementing Quality of Service (QoS) across the network. Install Avaya licensing keys for the Polycom RMX MCU and the Polycom HDX endpoint. Required prior knowledge Ensure that you are familiar with the following: System Manager domains, creating SIP entities, creating SIP entity links, administering locations, administering routing, and administering dial patterns. Administering Communication Manager, administering domains on signaling groups, administering IP network regions, creating AAR/ARS routing, and endpoint interfaces. Related documents For more information on Avaya Video Conferencing Solution, Communication Manager, Session Manager, and System Manager, see the following product documentation on the Avaya support Web site at Administering Network Connectivity for Avaya Aura Communication Manager, Administering Avaya Aura Session Manager, Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

19 Chapter 2: Managing bandwidth allocation and administering bandwidth pools on Avaya Video Conferencing Solution This chapter provides information on how to manage bandwidth allocation and how to administer bandwidth pools. Managing bandwidth allocation To design and deploy Avaya Video Conferencing Solution on the customer network, you must perform the following tasks: To ensure the customer network supports high bandwidth demands of video, perform a network assessment. Determine whether the customer network has implemented Class of Service (COS). If the customer network has not implemented COS, recommend implementation of COS. For more information on COS, see Class of Service on page 20. Determine the method to allocate bandwidth for audio and video on the customer network. Audio endpoints and standard and high-definition video endpoints use different amounts of bandwidth. To ensure adequate bandwidth allocation for video, you can group bandwidth in the following sequence: 1. Geographic locations 2. Audio and video endpoints 3. SIP and H.323 endpoints 4. Normal users and priority users for H.323 endpoints For H.323 audio and video endpoints, in Communication Manager, configure network regions for bandwidth pools. For SIP audio and video endpoints, in System Manager, configure locations to group endpoints. Table 2: Bandwidth allocation in Avaya Aura on page 20 lists the components you can use to allocate bandwidth to endpoints in Avaya Aura. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

20 Managing bandwidth allocation and administering bandwidth pools on Avaya Video Conferencing Solution Table 2: Bandwidth allocation in Avaya Aura Endpoints Bandwidth allocation through SIP endpoints Session Manager and Communication Manager H.323 endpoints Communication Manager Table 3: Bandwidth requirements for Avaya Video Conferencing Solution endpoints on page 20 lists the bandwidth requirements of Avaya Video Conferencing Solution audio and video endpoints. Table 3: Bandwidth requirements for Avaya Video Conferencing Solution endpoints Endpoint Bandwidth Pixels Frames per second Avaya 1010 and Avaya 1020 Avaya 1030, Avaya 1040, and Avaya Mbps 720p 30fps 768Kbps 720p 60fps 1.1Mbps 720p 60fps 1.7Mbps 1080p 30fps Ensure that you add the following bandwidth usage overhead when you calculate bandwidth requirements for the customer network: When you configure Avaya 1000 series endpoints for content sharing, add a bandwidth usage overhead of 128Kbps. For wideband audio, add a bandwidth usage overhead of 80Kbps. Class of Service COS provides the ability to configure differentiated services to users, voice, and data. COS guarantees quality to data flow. To enable COS, assign a COS tag when you configure the video endpoint, or when you configure video on Communication Manager. If a COS tag is not assigned to a video endpoint, the network uses its best effort for a video transmission. 20 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

21 Bandwidth pools Classifying users for COS Users can be classified as normal users or priority users. To classify priority users, perform the following steps on Communication Manager: 1. Set up a priority class of service (COS) group. 2. Assign priority IP video to the priority COS group. 3. Assign a user to the priority COS group. 4. Assign a higher maximum call bit rate for priority IP multimedia. For priority video users of H.323 endpoints, you can set up separate bandwidth pools within a network region. Procedures on how to implement COS is out of the scope of this document. Bandwidth pools To ensure adequate bandwidth to support video, you can partition the total bandwidth by administering the following: For H.323 endpoints, administer network regions through Communication Manager. For information on how to assign bandwidth for H.323 endpoints through network regions, see Administering bandwidth pools through network regions in Communication Manager on page 36. For SIP endpoints, administer locations through System Manager. For information on how to assign bandwidth for SIP endpoints through locations, see Administering locations in System Manager on page 34. The following sections provide examples on different methods to partition bandwidth. Bandwidth pools for H.323 endpoints For customers who use only H.323 endpoints, you can partition bandwidth in the following sequence: 1. Geographic location 2. Audio calls and video calls 3. Normal users of video endpoints and priority users of video endpoints Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

22 Managing bandwidth allocation and administering bandwidth pools on Avaya Video Conferencing Solution You can categorize users into the following three bandwidth pools: Audio bandwidth: The audio bandwidth pool contains bandwidth for audio calls and the audio stream of multimedia calls. Normal video bandwidth: The normal video bandwidth pool contains bandwidth for the video stream of a call initiated by a normal video bandwidth pool user. You can configure the video bandwidth pool to be shared by users. When the video bandwidth pool is shared, audio calls can access bandwidth from the video bandwidth pool. If adequate bandwidth is not available, video endpoints of normal video bandwidth pool users may not receive video. Priority video bandwidth: The priority video bandwidth pool contains bandwidth that is dedicated to priority video bandwidth users. Audio bandwidth pool users and normal video bandwidth pool users cannot access bandwidth from the priority video bandwidth pool. If all the allocated bandwidth of the priority video bandwidth pool is in use, priority video bandwidth users can access bandwidth from the normal video bandwidth pool. Video endpoints of priority video bandwidth pool users receive a higher allocation of bandwidth than video endpoints of normal video bandwidth endpoints. The higher allocation of bandwidth to priority video bandwidth pool users ensures reliable and better quality video streaming during calls. Priority video bandwidth pool users may not receive video in calls from normal video bandwidth pool users in the following situations: Bandwidth is unavailable. Normal bandwidth pool is unavailable. If a normal video bandwidth pool user initiates the call, the priority video bandwidth pool user may not receive video due to lack of availability of bandwidth in the normal video bandwidth pool. Normal video bandwidth pool cannot share the bandwidth of priority video bandwidth pool. Figure 4: Bandwidth allocation between H.323 endpoints on page 23 illustrates bandwidth allocation among two geographical areas, H.323 endpoints, and types of users. Total bandwidth of 100Mbps is available. Network region 1 (NR1) uses multimedia endpoints in conference rooms and multimedia endpoints on desks in executive suites. NR1 is allocated 50Mbps bandwidth. Network region 2 (NR2) uses multimedia endpoints in conference rooms and multimedia endpoints on desks in executive suites. NR2 also uses multimedia endpoints and audio endpoints on desks in the facility. NR2 is allocated 50Mbps. In NR1 and NR2, the following bandwidth allocation is an example of how you can allocate the 50Mbps bandwidth among the three bandwidth pools: Audio bandwidth pool: 40% Normal video bandwidth pool: 20% Priority video bandwidth pool: 40% 22 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

23 Bandwidth pools For information on how to classify users to the normal video bandwidth pool and the priority video bandwidth pool, see Classifying users for COS on page 21 Figure 4: Bandwidth allocation between H.323 endpoints Site 1 Site 2 Audio Pool Normal Video Pool Priority Video Pool NR 1 NR Mbps 50 Mbps Virtual NR NR 2 Audio Pool Normal Video Pool Priority Video Pool WAN Cloud cycmnrh3 LAO Examples of H.323 bandwidth pools The following examples describe how you can specify bandwidth settings on the customer network. For information on how to administer bandwidth pools through network regions, see Administering bandwidth pools through network regions in Communication Manager on page 36. Example 1 Audio calls only: - Table 4: Bandwidth allocation only for audio calls on page 23 describes how you can configure bandwidth pools to allocate bandwidth only for audio calls. Table 4: Bandwidth allocation only for audio calls Total bandwidth Audio bandwidth pool Priority video bandwidth pool Normal video bandwidth pool 50Mbps 50Mbps 0Mbps 0Mbps No Share normal video bandwidth pool? Example 2 Video calls support for all users: - Table 5: Bandwidth allocation to support video calls for all users on page 24 describes how you can configure bandwidth pools to allocate bandwidth to support video calls for all users. Users have access to all available bandwidth. Do not classify users. Audio calls and multimedia calls share one bandwidth pool. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

24 Managing bandwidth allocation and administering bandwidth pools on Avaya Video Conferencing Solution Audio calls access bandwidth from the normal video bandwidth pool. Table 5: Bandwidth allocation to support video calls for all users Total bandwidth Audio bandwidth pool Priority video bandwidth pool Normal video bandwidth pool Share normal video bandwidth pool? 50Mbps 0Mbps 0Mbps 50Mbps Yes Example 3: Video calls support for normal video bandwidth pool users and priority video bandwidth pool users: - Table 6: Bandwidth allocation to support video calls for normal bandwidth pool users and priority bandwidth pool users on page 24 describes how you can configure bandwidth pools to allocate bandwidth to support video calls for normal video bandwidth pool users and priority video bandwidth pool users. Reserve 50% bandwidth for priority video bandwidth pool users. Priority video bandwidth pool users can access bandwidth from the normal video bandwidth pool. Audio calls access bandwidth from the normal video bandwidth pool. Table 6: Bandwidth allocation to support video calls for normal bandwidth pool users and priority bandwidth pool users Total bandwidth Audio bandwidth pool Priority video bandwidth pool Normal video bandwidth pool 50Mbps 0Mbps 25Mbps 25Mbps No Share normal video bandwidth pool? Example 4: Reserving bandwidth for video calls: - Table 7: Bandwidth allocation to reserve maximum bandwidth for video bandwidth pool users on page 25 describes how you can allocate bandwidth in order to reserve bandwidth for video calls. Reserve maximum bandwidth for video bandwidth pool users. Allocate mininum bandwidth for audio bandwidth pool users. Minimum bandwidth allocation limits audio calls to a few audio calls and a maximum of 2 priority calls. The number of total audio calls depends on the bit rate. When the audio bandwidth is exhausted, audio calls connect through an alternate route. Small number of priority video pool users. If the priority video bandwidth pool is exhausted, priority video calls access bandwidth from the normal video bandwidth pool before the priority video calls access bandwidth from the audio bandwidth pool. 24 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

25 Bandwidth pools Table 7: Bandwidth allocation to reserve maximum bandwidth for video bandwidth pool users Total bandwidth Audio bandwidth pool Priority video bandwidth pool Normal video bandwidth pool 50Mbps 15Mbps 10Mbps 25Mbps No Share normal video bandwidth pool? Example 5: Shared bandwidth for normal video pool users: - Table 8: Bandwidth allocation to share bandwidth between normal video pool users on page 25 describes how you can allocate bandwidth among normal video pool users. Enable normal video bandwidth pool to be shared among users. No priority video bandwidth pool users. All bandwidth can be allocated for audio calls. Table 8: Bandwidth allocation to share bandwidth between normal video pool users Total bandwidth Audio bandwidth pool Priority video bandwidth pool Normal video bandwidth pool 50Mbps 15Mbps 0Mbps 35Mbps Yes Share normal video bandwidth pool? Example 6: Shared bandwidth with priority video bandwidth pool users: - Table 9: Bandwidth allocation to share bandwidth with priority video bandwidth pool users on page 25 shows how you can allocate bandwidth to share bandwidth with priority video pool users. Guarantees minimum bandwidth for audio calls and video calls. Share normal video bandwidth pool. 10Mbps bandwidth reserved for priority video bandwidth pool users. Audio bandwidth pool users cannot access priority video bandwidth pool. A maximum of 40Mbps bandwidth available for audio calls. Table 9: Bandwidth allocation to share bandwidth with priority video bandwidth pool users Total bandwidth Audio bandwidth pool Priority video bandwidth pool Normal video bandwidth pool 50Mbps 15Mbps 10Mbps 25Mbps Yes Share normal video bandwidth pool? Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

26 Managing bandwidth allocation and administering bandwidth pools on Avaya Video Conferencing Solution Bandwidth pools for SIP endpoints For customers who use only SIP endoints, you can partition bandwidth in the following sequence: 1. Geographic location 2. Audio and video For SIP endpoints, you can categorize users into the following 2 bandwidth pools: Audio:The audio bandwidth pool contains bandwidth for audio calls and the audio stream of multimedia calls. Video:The video bandwidth pool contains bandwidth for the video stream of a call initiated by a video endpoint user. You can configure the video bandwidth pool to be shared by users. When the video bandwidth pool is shared, audio calls access bandwidth from the video bandwidth pool. Normal video bandwidth pool users and priority video bandwidth pool users share the video bandwidth pool. Calculating bandwidth allocation Calculate allocation of bandwidth for each bandwidth pool with the average bandwidth usage. To determine average bandwidth used for each call, use the following formula: Y=AC+BD The following list explains the variables in the formula to calculate average bandwidth: Y: Average bandwidth for each call A: Percentage of audio calls B: Percentage of video calls C: Average bandwidth used by calls D: Average bandwidth used by video calls. Example (0.80x83)+(0.20x768)=220Kbps The following values explain the calculation of the average bandwidth of 220Kbps in the example: A: 80% B: 20% C: 83Kbps 26 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

27 Bandwidth pools The average bandwidth of 83Kbps used by calls is based on calls that use the G.711-mu codec. D: 768Kbps The average bandwidth of 768Kbps used by video calls is based on the average screen resolution. Figure 5: Bandwidth allocation among SIP endpoints in Avaya Aura on page 27 illustrates bandwidth allocation between two geographic areas and SIP endpoint types. Location A uses audio endpoints and video endpoints on desks and audio endpoints and video endpoints in conference rooms. Location A is allocated bandwidth of 100Mbps. The audio bandwidth pool is allocated bandwidth of 25Mbps, and the video bandwidth pool is allocated bandwidth of 75Mbps. Location B uses audio endpoints and video endpoints on desks and audio endpoints and video endpoints in conference rooms. Location B is allocated bandwidth of 25Mbps. The audio bandwidth pool is allocated bandwidth of 5Mbps, and the video bandwidth pool is allocated bandwidth of 20Mbps. Avaya Aura supports bandwidth sharing. The audio bandwidth pool can access bandwidth from the video bandwidth pool. Figure 5: Bandwidth allocation among SIP endpoints in Avaya Aura Site 1 Site 2 Audio Pool Video Pool Loc A 100 Mbps 25 Mbps WAN Loc B Audio Pool Video Pool cycmlsip LAO Examples of bandwidth pools for SIP endpoints The following examples describe how you can allocate bandwidth for SIP endpoints in various scenarios on the customer network. For information on how to set up bandwidth pools through locations, see Administering locations in System Manager on page 34. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

28 Managing bandwidth allocation and administering bandwidth pools on Avaya Video Conferencing Solution Example 1 Audio calls only: - Table 10: Bandwidth allocation to support only audio calls on page 28 describes how you can allocate bandwidth to support audio calls at one location. Do not allocate bandwidth for video calls. Configure the network to initiate only audio calls. Table 10: Bandwidth allocation to support only audio calls Total bandwidth Audio bandwidth pool Video bandwidth pool 25Mbps 25Mbps 0Mbps No Share bandwidth pools? Example 2: SIP video calls only: - Table 11: Bandwidth allocation to support only video calls on page 28 describes how you can allocate bandwidth to support only video calls at one location. Do not allocate bandwidth for audio calls. Configure the network to initiate only video calls. Table 11: Bandwidth allocation to support only video calls Total bandwidth Audio bandwidth pool Video bandwidth pool 100Mbps 0Mbps 100Mbps No Share bandwidth pools? Example 3: SIP audio calls and video calls: - Table 12: Bandwidth allocation to support audio calls and video calls on page 28 describes how you can allocate bandwidth to support audio calls and video calls in one location. Allocate bandwidth to support audio calls and video calls. Allocate maximum bandwidth to support video calls. Table 12: Bandwidth allocation to support audio calls and video calls Total bandwidth Audio bandwidth pool Video bandwidth pool 100Mbps 20Mbps 80Mbps Yes Share bandwidth pools? 28 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

29 Bandwidth pools Bandwidth pools for H.323 and SIP endpoints For customers who use H.323 and SIP endpoints, you can partition bandwidth in the following sequence: 1. Geographic location 2. H.323 endpoints and SIP endpoints 3. Normal users of H.323 endpoints and priority users of H.323 endpoints Calculating bandwidth allocation Calculate allocation of bandwidth for each bandwidth pool with the average bandwidth usage. To determine average bandwidth used for each call, use the following formula: Y=AC+BD The following list explains the variables in the formula to calculate average bandwidth: Y: Average bandwidth for each call A: Percentage of audio calls B: Percentage of video calls C: Average bandwidth used by calls D: Average bandwidth used by video calls. Example (0.80x83)+(0.20x768)=220Kbps The following values explain the calculation of the average bandwidth of 220Kbps in the example: A: 80% B: 20% C: 83Kbps The average bandwidth of 83Kbps used by calls is based on calls that use the G.711-mu codec. D: 768Kbps The average bandwidth of 768Kbps used by video calls is based on the average screen resolution. Network regions and locations must have a 1:1 mapping. This mapping has the following limitations: Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

30 Managing bandwidth allocation and administering bandwidth pools on Avaya Video Conferencing Solution Communication Manager supports a maximum of 250 network regions, but Session Manager supports thousands of locations. Session Manager maps IP addresses to locations. Communication Manager maps IP addresses to network regions. These mappings must be manually synchronized. As part of the Communication Manager administration, the SIP trunk to Communication Manager must be configured within a dummy network region for which no bandwidth limits are set. You must divide the bandwidth limits of the physical links between the Session Manager bandwidth and the Communication Manager bandwidth. The appropriate division depends on the expected call traffic details. The process to set up bandwidth pools is identical to the process of setting up bandwidth pools for H.323 endpoints and SIP endpoints. For information on administering bandwidth pools: Through network regions, see Administering bandwidth pools through network regions in Communication Manager on page 36. Through locations, see Administering locations in System Manager on page 34. Figure 6: Bandwidth allocation to support H.323 endponts and SIP endpoints on page 31 describes how you can allocate bandwidth to support H.323 endpoints and SIP endpoints among two geographic areas, two endpoint types, and types of users. Total bandwidth of 225Mbps is available. Bandwidth of 125Mbps is allocated to SIP endpoints. Bandwidth of 100Mbps is allocated to H.323 endpoints. Location A and NR1 largely use SIP video endpoints on desks along with a few H.323 video endpoints on desks, in conference rooms, and other areas in the facility. Location A is allocated bandwidth of 100Mbps for SIP endpoints. NR1 is allocated bandwidth of 50Mbps for H.323 endpoints. Location B and NR2 largely use H.323 endpoints along with a few SIP video endpoints on desks, in conference rooms, and in other area of the facility. Location B is allocated bandwidth of 25Mbps for SIP endpoints. NR2 is allocated bandwidth of 50Mbps for H.323 endpoints. In NR1 and NR2, the following bandwidth allocation is an example of how you can allocate bandwidth of 50Mbps between the three bandwidth pools: Audio bandwidth pool: 20% Normal video bandwidth pool: 50% 30 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

31 Bandwidth pools Priority video bandwidth pool: 30% For information on how to classify users between the three pools, see Classifying users for COS on page 21. The terminating endpoint allocates bandwidth for calls initiated between locations and network regions. For example, if you initiate a call from an H.323 endpoint in Location A to a SIP endpoint in Location B, the bandwidth pool of Session Manager allocates the bandwidth for the call. If you initiate a call from a SIP endpoint in Location A to an H.323 endpoint in Location B, the bandwidth pool of Communication Manager allocates the bandwidth for the call. Figure 6: Bandwidth allocation to support H.323 endponts and SIP endpoints Site 1 Site 2 Audio Pool Video Pool Audio Pool Normal Video Pool Priority Video Pool Loc A NR 1 Network Pipe from Site 1 Total BW: Mbps 50 Mbps Loc CM Virtual Location NR 100 Virtual NR WAN Cloud Network Pipe from Site 2 Total BW: Mbps 50 Mbps Loc B NR 2 Audio Pool Video Pool Audio Pool Normal Video Pool Priority Video Pool cycmlocc LAO Examples of H.323 and SIP bandwidth pools The following examples describe how you can specify bandwidth settings on the customer network. Example 1 Audio calls only: - Table 13: Bandwidth allocation to support only audio calls for H.323 endpoints and SIP endpoints on page 32 describes how you can allocate bandwidth to support only audio calls for a single location. The total bandwidth available in the customer network can support only audio calls at one location. SIP audio endpoints cannot share bandwidth with H.323 audio endpoints. For intralocation calls, bandwidth is allocated from the total bandwidth allocated to the location. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

32 Managing bandwidth allocation and administering bandwidth pools on Avaya Video Conferencing Solution Table 13: Bandwidth allocation to support only audio calls for H.323 endpoints and SIP Total bandwidth SIP audio bandwidth pool SIP video bandwidth pool H.323 audio bandwidth pool H.323 audio bandwidth pool 75Mbps 25Mbps 0Mbps 50Mbps 0Mbps endpoints Example 2 Video calls only: - Table 14: Bandwidth allocation to support only video calls for H.323 endpoints and SIP endpoints on page 32 describes how you can allocate bandwidth to support only video calls at one location. The total bandwidth available in the customer network can support only video calls at one location. SIP video endpoints are allocated bandwidth of 100Mbps. H.323 video endpoints are allocated bandwidth of 50Mbps. H.323 video endpoints are divided into normal video bandwidth pool users and priority video bandwidth pool users. SIP video endpoints cannot share bandwidth with H.323 video endpoints. For intralocation calls, bandwidth is allocated from the total bandwidth allocated to the location. Table 14: Bandwidth allocation to support only video calls for H.323 endpoints and SIP endpoints Total bandwidth SIP audio bandwidth pool SIP video bandwidth pool H.323 audio bandwidth pool H.323 normal video bandwidth pool 150Mbps 0Mbps 100Mbps 0Mbps 30Mbps 20Mbps H.323 priority video bandwidth pool Example 3: Audio calls and video calls for H.323 endpoints and SIP endpoints: - Table 15: Bandwidth allocation for audio calls and video calls for H.323 endpoints and SIP endpoints on page 33 describes how you can allocate bandwidth to support audio calls and video calls for H.323 endpoints and SIP endpoints. The total bandwidth available in the customer network can support audio calls and video calls at one location. SIP video endpoints are allocated bandwidth of 100Mbps. H.323 video endpoints are allocated bandwidth of 50Mbps. H.323 video endpoints are divided into normal video bandwidth pool users and priority video bandwidth pools users. SIP video endpoints cannot share bandwidth with H.323 video endpoints. 32 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

33 Bandwidth pools For intralocation calls, bandwidth is allocated from the total bandwidth allocated to the location. Table 15: Bandwidth allocation for audio calls and video calls for H.323 endpoints and SIP endpoints Total bandwidth SIP audio bandwidth pool SIP video bandwidth pool H.323 audio bandwidth pool H.323 normal video bandwidth pool H.323 priority video bandwidth pool 150Mbps 25Mbps 75Mbps 20Mbps 20Mbps 10Mbps Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

34 Managing bandwidth allocation and administering bandwidth pools on Avaya Video Conferencing Solution Administering bandwidth pools The following procedures provide information on how to administer bandwidth pools through two methods: Administering locations in System Manager Administering bandwidth pools through network regions in Communication Manager Administering locations in System Manager To administer locations in System Manager, perform the following tasks: 1. Log in to the System Manager Web console. 2. Click Elements > Routing > Locations. 3. To view the Location Details page, click New. 4. In the Name field, enter the location name. 5. Enter notes about the location, if required. 6. In the Overall Managed Bandwidth section, set the following parameters for the location: Managed Bandwidth Units Total Bandwidths 7. In the Per-Call Bandwidth Parameters section, set all the parameters. 8. To add a location pattern, under Location Pattern, click Add. 9. Enter an IP address pattern to match. 10. Enter notes about the location pattern, if required. 11. Continue clicking Add until all the required Location Pattern matching patterns are configured. 12. To save the settings, click Commit. Table 16: Location Details page field descriptions on page 34 provides descriptions of the fields on the Location Details page. Table 16: Location Details page field descriptions Name General section Description Name Notes The name of the location Notes about the location 34 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

35 Administering bandwidth pools Name Description Overall Managed Bandwidth section Managed Bandwidth Units The bandwidth unit for Overall Managed Bandwidth values. Total Bandwidth Multimedia Bandwidth Audio Calls Can Take Multimedia Bandwidth Per-Call Bandwidth Parameters section Maximum Multimedia Bandwidth (Intra-Location) The total bandwidth available for use by calls between the current location and other locations. Any attempt to exceed the total bandwidth limit results in calls being routed through an alternate route or denied. If you do not specify a value, the bandwidth limit is processes as infinite. The bandwidth available for use by multimedia calls between the current location and other locations. Multimedia bandwidth is a subset of the Total Bandwidth value. Any attempt to exceed the multimedia bandwidth limit results in calls being routed through an alternate route or denied. If you do not specify a value, the use of the limit defined for Total Bandwidth depends on the value of Audio Calls Can Take Multimedia Bandwidth. If you administer this parameter, Total Bandwidth can be used for any call type. If you do not administer this parameter, Total Bandwidth can be used only for audio calls. Specifies the use of multimedia call bandwidth for audio calls. If you do not select this check box, the bandwidth reserved for multimedia calls may also be used for audio calls. If you select this check box, this bandwidth may only be used for multimedia calls. The maximum bandwidth allowed for a single multimedia call within this location. Calls requesting more bandwidth than this value are modified to use less bandwidth. The range is 0Kbps to 15360Kbps. The default value is 1000Kbps. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

36 Managing bandwidth allocation and administering bandwidth pools on Avaya Video Conferencing Solution Name Maximum Multimedia Bandwidth (Inter-Location) Minimum Multimedia Bandwidth Default Audio Bandwidth Description The maximum bandwidth allowed for the multimedia stream of a multimedia call between the current location and another location. Calls requesting more bandwidth than this value are modified to use less bandwidth. The range is 0Kbps to 15360Kbps. The default value is 1000Kbps. The minimum bandwidth that Session Manager allocates to each multimedia stream when Session Manager reduces the bandwidth allocation to a call to enforce bandwidth restriction. If a bandwidth restriction requires Session Manager to reduce a media stream below this level, the stream is removed from the call. Removing the stream may result in the entire call being blocked. Media requests for bandwidth beneath this minimum bandwidth allocation limit will not be blocked. This is solely a restriction on the Session Manager ability to modify requests. The range is 64Kbps to 15360Kbps. The default value is 64Kbps. The audio bandwidth assumed to be used by a call originating in the current location that does not specify its bandwidth needs using SDP. Such calls are assumed to be audio calls only. The range is 0Kbps to 15360Kbps. The default value is 80Kbps. Administering bandwidth pools through network regions in Communication Manager To administer network regions for normal bandwidth pools and priority bandwidth pools in Communication Manager, perform the following steps: 1. On the SAT screen, type change ip-network-region <IP network region>. 2. Navigate to the Inter Network Region Connection Management screen. 3. Set the following parameters: codec set 36 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

37 Administering bandwidth pools direct WAN Units Total Video Norm Video Prio Shr 4. Repeat steps 1 to 3 for each IP network region where you want to set up normal bandwidth pools and priority bandwidth pools. Table 17: Network Regions field descriptions on page 37 provides descriptions of the fields on the Network Regions screen. Table 17: Network Regions field descriptions Field Value range Usage codec-set 1 to 7 pstn blank The codec set used between two regions. If the route passes through two nonadjacent pairs of regions, this field must not be blank. If the two regions are not connected, this field must be blank. direct-wan This field indicates whether a WAN link connects the two regions. If a codec set is administered, Communication Manager enables the default value. WAN-BW-limits (Units) WAN-BW-limits (Total) Calls Dynamic kbps mbps Blank for no limit 1 to to 65 Blank The unit of measure corresponding to the value of bandwidth limitation. Bandwidth must be limited by the number of connections, bandwidth in kbps, or mbps. Blank: No limits. This is the default value. The bandwidth limits for direct WAN links. Values for this field can be entered in the number of connections. The units for each value range are: 1 to 9999: kbps 1 to 65: mbps Blank: No limits. This is the default value. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

38 Managing bandwidth allocation and administering bandwidth pools on Avaya Video Conferencing Solution Field Value range Usage Video (Norm) 1 to to 65 Blank Video (Prio) 1 to to 65 Blank The amount of bandwidth to allocate for the normal video bandwidth pool of each IP network region. The units for each value range are: 1 to 9999: kbps 1 to 65: mbps Blank: No limits. This is the default. The amount of bandwidth to allocate for the priority video bandwidth pool of each IP network region. The units for each value range are: 1 to 9999: kbps 1 to 65: mbps Blank: No limits. This is the default. Video (Shr) Specifies whether audio calls can share bandwidth with the normal video bandwidth pool for each link among IP network regions. 38 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

39 Chapter 3: Call Admission Control Call Admission Control (CAC) is a service Avaya Aura provides to prevent IP telephony from exceeding the usage of allocated bandwidth in an IP network. To reduce network congestion, Avaya Aura limits the amount of bandwidth allocated to each call. If IP telephony exceeds the usage of the allocated bandwidth, the network may experience call degradation due to discarded packets and disconnected calls. If CAC determines that a particular link is reaching the maximum usage limit of its allocated bandwidth, CAC prevents establishment of new calls. During network congestion, CAC redirects new calls through an alternate route. For example, CAC may redirect the call through a public trunk. When CAC redirects a calls, users at both ends of the call are not aware that CAC has set up the call through an alternate route. Customers incur a cost when CAC redirects a call through a public trunk. However, it is preferable for CAC to redirect a call through a public trunk because customers prefer to incur a cost on the call rather than have CAC prevent establishment of new calls. When CAC determines network congestion and redirects a call, it ensures customer satisfaction. CAC ensures customer experience is identical, irrespective of whether the telephony network is functioning or facing network congestion. CAC interactions with Quality of Service To understand CAC interactions with Quality of Service (QoS), you must have basic knowledge of the QoS architecture. Implementation of QoS is outside the scope of this document. Tagging packets with DSCP tags ensures that, during network congestion, routers drop packets in the following determined order: 1. Data packets 2. Video packets 3. Audio packets of a video call 4. Audio packets of an audio call Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

40 Call Admission Control CAC only controls audio and video packets. During a network congestion, QoS ensures that packets are dropped in the determined order. Table 18: Avaya recommended DSCP tags for packets on page 40 lists the DSCP tags Avaya recommends for IP packets. Table 18: Avaya recommended DSCP tags for packets Packet DSCP tag Audio EF (DSCP 46) (Expedited Fowarding: RFC-2598) Audio packets in video calls AF41 (DSCP 34) (Assured Forwarding: Class 4, Low Drop Precedence, RFC 2597) Video packets in video calls AF42 (DSCP 36) (Assured Forwarding: Class 4, Medium Drop Precedence, RFC 2597) Data AF11 (DSCP 10) (Assured Forwarding: Class 1, Low Drop Precedence) CAC bandwidth calculation Administration of CAC bandwidth limits is calculated in Kbps. To determine whether CAC can allow a new call, CAC measures the active bandwidth usage. Bandwidth usage and the specified bandwidth allocation limit is bidirectional. For example, if an audio call using the G.729 codec sends voice at the rate of 18Kbps, CAC calculates the bandwidth usage of the call as 18Kbps. If the available bandwidth is less than the call bandwidth usage of 18Kbps, CAC denies establishment of the call. CAC does not track different levels of bandwidth usage in each direction of the call. 40 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

41 CAC implementation CAC implementation Session Manager manages CAC. As the central proxy in an Avaya Aura installation, Session Manager views the establishment of calls in the form of SIP signals. Through SIP signals, Session Manager determines following details of the call: CAC reports Bandwidth requirement through SDP Locations involved in the call. Location administration contains details on bandwidth allocation limits. Bandwidth allocation limits apply to the link from the originating location to the Avaya Aura core. Session Manager generates following reports: CAC process Real-time bandwidth usage data and bandwidth usage history for each location Events generated for each call that is impacted by bandwidth allocation limits Alarms generated when a high occupancy state persists in locations For every INVITE dialog that Session Manager receives, CAC processes the SIP message in the following manner: 1. CAC parses SDP details in the SIP message to determine the following: Locations involved in the call Bandwidth usage of the call 2. If CAC determines that a location is exceeding the bandwidth allocation limit, it performs one of the following actions on the SIP request: CAC denies the request to establish a call. It sends a SIP 488 response message that contains a 370 Warning header, which indicates the reason for the denial. CAC determines an alternate route for the call. In the alternate route, CAC again processes the locations for bandwidth allocation limits. CAC establishes the call with reduced bandwidth. 3. CAC processes in-dialog SIP requests in an identical manner to an INVITE dialog. If adequate bandwidth is unavailable for in-dialog SIP requests, CAC cannot determine an alternate route for in-dialog SIP requests. 4. CAC determines bandwidth usage changes of a call in real-time. CAC determines the bandwidth usage changes of a call in the following situations: Addition of a video stream to an audio call. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

42 Call Admission Control Removal of the video stream from a call, which modifies the call to an audio call. Session Manager cannot calculate bandwidth usage and enforce bandwidth allocation limits in the following scenarios: When Session Manager is not involved in the route of a call. When Session Manager cannot determine the terminating location of a call. For example, after Session Manager processes the call signaling of a particular call, the route of the call involves other components of Avaya Video Conferencing Solution. For example, Session Manager cannot determine the bandwidth usage of H.323 endpoints in the following scenarios: Calls between two H.323 endpoints. H.323 endpoints do not generate SIP messages. Calls from a SIP endpoint to an H.323 endpoint. Session Manager determines the route of the call from the SIP endpoint to Communication Manager of the H.323 endpoint. In this case, Session Manager cannot determine the location of the H.323 endpoint. CAC records details of the call traffic from the H.323 endpoint only if Communication Manager initiates the CAC process on the call. In this case, CAC determines the locations of the H.323 endpoints. For calls from H.323 endpoints to SIP endpoints, Communication Manager cannot determine the bandwidth usage and the location of SIP endpoints. CAC can track the bandwidth usage and enforce bandwidth allocation limits only if CAC can determine bandwidth usage of the entire route of a call. When CAC cannot determine the total bandwidth usage of a call, Session Manager and Communication Manager record subsets of the bandwidth usage of a location. To ensure accurate CAC processing, Session Manager and Communication Manager communicate this bandwidth usage data to CAC. CAC interactions on Session Manager and Communication Manager The CAC process must initiate before the SIP INVITE dialog reaches the destination endpoint because CAC may determine that there is not enough bandwidth for the call in the original route, and CAC may deny the establishment of the call. If CAC denies the establishment of the call, it can determine an alternate route for the call. CAC can determine this alternate route for the call because CAC determined the bandwidth requirement of the call before the SIP INVITE dialog reached the destination endpoint. Call traffic routes through more than one Avaya Aura component. To ensure accurate calculation of bandwidth usage and enforcement of bandwidth allocation limits, the component located closest to the destination endpoint must calculate the bandwidth usage because the components located closest to the destination endpoint can determine the destination location. 42 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

43 CAC interactions on Session Manager and Communication Manager Session Manager performs the CAC process before the SIP INVITE message exits the Session Manager cluster. To ensure Communication Manager performs the CAC process after the SIP INVITE message exits the Session Manager cluster, on System Manager, you must administer Communication Manager as a SIP entity that supports CAC. Figure 7: CAC calculations on Communication Manager located nearest the H.323 endpoint on page 43 illustrates the following setup: SIP endpoint is registered to Session Manager 1. Session Manager 1 runs Avaya Aura Sequenced Applications. H.323 endpoint is registered to Communication Manager. Communication Manager is accessible through a link to Session Manager 2. If the SIP endpoint user calls the H.323 endpoint user, Session Manager 1 and Session Manager 2 cannot determine the location of the H.323 endpoint when they receive the SIP INVITE message. Communication Manager performs the CAC process, and determines the location of the H.323 endpoint before Communication Manager terminates the SIP INVITE message. If the H.323 endpoint user calls the SIP endpoint user, Communication Manager and Session Manager 2 cannot determine the location of the SIP endpoint when they receive the SIP INVITE message. Session Manager 1 performs the CAC process, and determines the location of the SIP endpoint before Session Manager 1 terminates the SIP INVITE message. Figure 7: CAC calculations on Communication Manager located nearest the H.323 endpoint Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

44 Call Admission Control Figure 8: CAC calculations on Session Manager 2 with CAC capability located nearest the PBX endpoint on page 44 illustrates the following scenario: SIP endpoint is registered to Session Manager 1. Session Manager 1 runs Avaya Aura Sequenced Applications. The PBX user endpoint is registered to the third-party PBX. PBX is administered in System Manager as an entity that does not support CAC. If the SIP endpoint user calls the PBX endpoint user, Session Manager 2 performs the CAC process. PBX cannot perform the CAC process. Session Manager 2 determines the location of the PBX endpoint from the administered location of the PBX in System Manager. Figure 8: CAC calculations on Session Manager 2 with CAC capability located nearest the PBX endpoint Recommendations for CAC interactions on Session Manager and Communication Manager In scenarios where Session Manager cannot determine bandwidth usage because Session Manager cannot determine the destination location, Avaya recommends the following methods for mitigation: 1. Administer Communication Manager to not support CAC. In this case, Session Manager processes all non SIP calls to terminate at the administered location in Communication Manager that does not support CAC. Session Manager performs all CAC processes. 44 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

45 Location determination on Session Manager 2. Administer partial limit enforcement. For example, a link has bidirectional bandwidth capacity of 10Mbps. The endpoints at the location of the link are SIP endpoints and non SIP endpoints in the ratio of 3:1. The frequency of calls is approximately equal for all endpoints. The endpoints make audio calls using the G.729 codec. In this scenario, administer the following configuration: Enable CAC on Session Manager. Enable CAC on Communication Manager to process only non SIP endpoints. On Session Manager, allocate a bandwidth of 9Mbps. On Communication Manager, allocate a bandwidth of 3Mbps. The network allows a total traffic flow of 12Mbps. CAC denies establishment of calls in the following scenarios: - If the bandwidth usage of the link reaches 9mbps, CAC denies establishment of calls to SIP endpoints. CAC denies establishment of calls only if all H.323 endpoints are idle. In this scenario, CAC denies establishment of calls because, when bandwidth usage of the link reaches 9Mbps, CAC on Session Manager determines that the bandwidth allocation for SIP endpoints is exhausted. - If the bandwidth usage of the link reaches 3Mbps, CAC denies establishment of calls to H.323 endpoints. CAC denies establishment of calls only when all SIP endpoints are idle. In this scenario, CAC denies establishment of calls because, when bandwidth usage of the link reaches 3Mbps, CAC on Communication Manager determines that the bandwidth allocation for H.323 endpoints is exhausted. Location determination on Session Manager Determination of location of the endpoints involved in a call is crucial for the CAC process. CAC calculates bandwidth usage of a call only for interlocation calls. Session Manager determines the following three locations for an endpoint that is involved in a call: Signaling: The location where the SIP signal originates. CAC determines the signaling location by performing an IP pattern match on the IP address from the Via header in the SIP request. If the IP address does not match an IP pattern, CAC determines the location from the provisioned location of the SIP endpoint. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

46 Call Admission Control Home: The location where the endpoint is located. CAC determines the home location from the endpoint details administered in the endpoint profile. - If CAC determines the home location, CAC utilizes the media location to process the call. - If CAC cannot determine media location, CAC utilizes the signaling location to process the call. - If CAC cannot determine the signaling location, CAC utilizes the home location to process the call. - If CAC cannot determine the home location, it processes the endpoint as a non SIP endpoint. CAC does not implement any bandwidth allocation limits on the non SIP endpoint. Media: The location where the media originates. CAC determines the media location by performing an IP pattern match on the IP address of the media origin that is specified in the SDP tag. Details on IP patterns are available in System Manager location administration. If you do not provision IP patterns when you administer locations in System Manager, CAC cannot determine the location of a SIP endpoint. CAC processes SIP endpoints as if the endpoints are calling from the home location even when the SIP endpoints are calling from another location. CAC media priority CAC prioritizes media based on the type of media from data available in SDP tags. CAC categorizes media in the following two types: Audio Multimedia: This type of media includes video, content sharing, and audio associated with video. You can administer the following bandwidth allocation limits when you administer locations in System Manager: Total bandwidth allocation of the link Total bandwidth allocation of the multimedia stream in the link: This sublimit prevents a few high bandwidth video endpoints from exhausting the bandwidth allocation of the location. When multimedia stream bandwidth allocation sublimit is reached, subsequent calls can gain access only to audio. Administration of the limit for the total bandwidth allocation of a link and the sublimit for multimedia stream bandwidth allocation in the link is optional. 46 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

47 Engineering the Avaya Video Conferencing Solution site Engineering the Avaya Video Conferencing Solution site You can engineer an Avaya Video Conferencing Solution site based on the following two methods: Endpoint quantity: To engineer a site based on the number of endpoints, you must calculate the bandwidth requirement of the site from number of employees, anticipated calling patterns, and the quality of experience required. Bandwidth limits: To engineer a site based on the amount of bandwidth available, you must determine the number of employees the site can support with the available bandwidth, the expected calling patterns, and the quality of experience required. Adjusting bandwidth requirements To ensure availability of adequate bandwidth, you must periodically monitor and analyze the call pattern of a site. Table 19: Bandwidth adjustment on page 47 provides information on how you can adjust bandwidth required by a site. Table 19: Bandwidth adjustment Desired result Target component Action Side effect Decrease bandwidth usage Maximum multimedia bandwidth Reduce bandwidth limit for each call Reduced end user experience Audio codec Select a compressed codec Reduced end user experience Number of endpoints Deploy less endpoints Busy Hour Call Completions Decrease the amount of simultaneous calls in the calculations Increased potential of calls being blocked Increase bandwidth usage Maximum multimedia bandwidth Increase bandwidth limit of each call Enhanced end user experience Audio codec Select a better quality codec Enhanced end user experience Busy Hour Call Completions Increase the amount of simultaneous calls in the calculations Decreased potential of calls being blocked Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

48 Call Admission Control For information on how to monitor CAC usage, see Viewing bandwidth usage on System Manager on page 52. Administering CAC for locations on Session Manager Prerequisites Ensure bandwidth limits for locations are allocated on Session Manager. Procedure To administer CAC for locations on Session Manager, perform the following steps: 1. Log in to the System Manager Web console. 2. Click Elements > Routing > Locations. The Locations page on Session Manager contains fields that govern bandwidth constraints on the link from the endpoint location to the Avaya Aura core, bandwidth limits for each call, CAC alarm parameters, and data on IP addresses mapped to the location. The Locations page on Session Manager contains fields only for IP address mapping, total bandwidth limit, and calculation of bandwidth for each call. The Locations page on Session Manager Release 6.1 does not contain fields for CAC alarm parameters. 3. Set the following parameters: Total Bandwidth: The total bandwidth available between the current location and other locations. - If a call tries to exceed this limit, call quality may degrade, or calls may be blocked. - If the value of this field is not defined, CAC processes it as availability of unlimited bandwidth. 48 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

49 Administering CAC for locations on Session Manager Multimedia Bandwidth: The total bandwidth available for multimedia calls between the current location and other locations. This bandwidth is a subset of Total Bandwidth. - If a multimedia call tries to exceed this limit, call quality may degrade or calls may be blocked. - If the value of this field is not defined, multimedia calls are subject to the bandwidth limit provisioned in Total Bandwidth. Audio Calls Can Take Multimedia Bandwidth: If you select this check box, the bandwidth allocated to multimedia calls may also be used for audio calls. If you do not select this check box, to prevent overlap, Session Manager partitions the total bandwidth between audio and multimedia calls. A multimedia call is defined as a call that contains nonaudio media, such as a video call. You can configure the multimedia bandwidth sublimit to restrict multimedia calls. Session Manager processes a call either as an audio call or a multimedia call. If administered, the audio portion of a video call is considered multimedia, and subjected to the multimedia sublimit. If a call establishes as a multimedia call, but the users downgrade the call to an audio call, Session Manager continues to treat the call as a multimedia call. Maximum Multimedia Bandwidth: The maximum bandwidth allocation limit of a multimedia call in the current location. If a call requires more bandwidth than the maximum allocated multimedia bandwidth, Session Manager modifies the call to use less bandwidth. Minimum Multimedia Bandwidth: The minimum bandwidth that Session Manager uses to enforce bandwidth restriction when it modifies a multimedia call to use less bandwidth. The minimum allocated multimedia bandwidth ensures call quality. If lack of availability of adequate bandwidth requires Session Manager to reduce the bandwidth usage of a call below the minimum allocated multimedia bandwidth, Session Manager blocks the call or downgrades the call to an audio call. Default Audio Bandwidth: Session Manager allocates the default audio bandwidth to calls that do not contain details on bandwidth requirement through SDP, or if the SDP of the call is unreadable. Calls for which Session Manager cannot determine the bandwidth requirement are processed as audio calls. Session Manager Release 6.0 does not analyze SDP. It processes all calls as audio calls, and allocates the default audio bandwidth. Overall Alarm Threshold: The alarm threshold percentage over which Session Manager raises an alarm. Either select the Disabled check box or configure a value to this field. If you configure a value of 100%, Session Manager raises an alarm when calls sustain the maximum administered CAC values over the latency. Multimedia Alarm Threshold: The alarm threshold percentage over which Session Manager raises an alarm. Either select the Disabled check box or configure a value to Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

50 Call Admission Control this field. If you configure a value of 100%, Session Manager raises an alarm when calls sustain the maximum administered CAC value over the latency. Latency before Overall Alarm Trigger: The delay, in minutes, before Session Manager raises an alarm. The value of this field is between 0 minutes to 30 minutes. Latency before Multimedia Alarm Trigger: The delay, in minutes, before Session Manager raises an alarm. The value of this field is between 0 minutes to 30 minutes. The following points provide more information on the fields you need to administer CAC for locations in Session Manager: CAC alarm: CAC raises an alarm when it determines excess call traffic at a specific location. In Session Manager releases prior to Session Manager Release 6.2, you cannot customize alarm settings. In Session Manager Release 6.2, you can administer trigger settings for alarms. Session Manager takes a snapshot of the usage patterns every 60 seconds. If you configure the value of the latency to 1, Session Manager raises an alarm when the snapshots display over 80% bandwidth usage for 6 consecutive snapshots. If you configure the value of the latency to 0, Session Manager raises an alarm when 1 snapshot displays over 80% bandwidth usage. Disabling CAC alarms: You can globally disable all CAC alarms. If you globally disable all alarms, CAC alarms are disabled in Session Manager. For information on how to disable CAC alarms, see Disabling CAC alarms on page 50. Disabling SDP analysis: You can disable Session Manager from analyzing SDP headers to determine bandwidth values. For information on how to disable Session Manager from analyzing SDP headers, see Disabling Session Manager from analyzing SDP headers on page 51. IP Address Patterns: Avaya recommends you use IP Address to map IP addresses to location. Mapping IP addresses to locations has the following advantages: It aids location-based call routing and CAC processing. CAC determines an endpoint location by analyzing IP addresses specified in the SDP header to IP addresses mapped to locations. Disabling CAC alarms To disable CAC alarms, perform the following steps: 1. Log in to the System Manager Web console. 2. Go to Elements > Session Manager > Session Manager Administration. 50 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

51 Administering SIP entities to perform CAC processing 3. Select Global Setting. 4. Disable Call Admission Control Threshold Alarms. Disabling Session Manager from analyzing SDP headers To disable Session Manager from analyzing SDP headers, perform the following steps: 1. Log in to the System Manager Web console. 2. Go to Elements > Session Manager > Session Manager Administration. 3. Select Global Setting. 4. Remove Shared Bandwidth Manager for all administered SIP entities. 5. Set the following parameters: Default Audio Bandwidth Overall Managed Bandwidth Ensure the values you configure for Default Audio Bandwidth and Overall Managed Bandwidth adhere to the following formula: Desired call count limit for each location x Default Audio Bandwidth = Overall Managed Bandwidth If you disable Session Manager from analyzing SDP headers, Session Manager does not populate CDR records isvideo and maxbandwidth. Administering SIP entities to perform CAC processing To administer SIP entities to perform CAC processing, perform the following steps: 1. Log in to the System Manager Web console. 2. Set the following parameters: Supports CAC: Select this parameter to configure the SIP entity to perform CAC processing. Session Manager does not perform CAC processing on calls to the SIP entity. The SIP entity does not perform CAC processing on calls to Session Manager. Shared Bandwidth Manager: Select this parameter to configure the SIP entity to send bandwidth PUBLISH messages. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

52 Call Admission Control Avaya recommends you configure the SIP entity to send the bandwidth PUBLISH message to a minimum of 2 Session Managers for redundancy. To enable Shared Bandwidth Manager, you must select Supports CAC. The SIP entity sends bandwidth PUBLISH messages only for calls it receives from Session Manager. The SIP entity sends bandwidth PUBLISH messages for calls that do not route through Session Manager. Primary Session Manager Bandwidth Association: If you select Shared Bandwidth Manager, from the drop-down menu, you must select a Session Manager SIP entity as the primary Session Manager for bandwidth communication. Backup Session Manager Bandwidth Association: If you select Shared Bandwidth Manager, from the drop-down menu, you must select a Session Manager SIP entity as the backup Session Manager for bandwidth communication. Viewing bandwidth usage on System Manager Viewing real-time status of bandwidth usage on System Manager To view real-time status of bandwidth usage, perform the following steps: 1. Log in to the System Manager Web console. 2. Go to Elements > Session Manager > System Manager > Managed Bandwidth Usage. The Managed Bandwidth Usage page lists the following data for each location: Current bandwidth usage Bandwidth allocation limit Bandwidth usage percentage Number of calls The real-time bandwidth usage only records data for interlocation calls.the real-time bandwidth usage data does not include data for intralocation multimedia calls. 52 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

53 Optimizing video on the network Viewing perfomance status of each location on System Manager To view performance status of each location on System Manager, perform the following steps: 1. Log in to the System Manager Web console. 2. Go to Elements > Session Manager > Performance > Location Performance. 3. Select the location for which you want to view the performance data. 4. Specify the time frame for which you want to view the performance data. The Location Performance page displays the performance data in the following formats: CSV XML Graphs: The Location Performance page displays the following three graphs: - Location bandwidth in use: This graph records data only for interlocation calls. - Location bandwidth available: This graph records data only for interlocation calls. - Historical counts of call attempts for the location: This graph categorizes each call to denied, downgraded, or completed. This graph records data for all calls. Optimizing video on the network This section provides a list of requirements to optimize video on the network. Deploying QoS To deploy QoS, perform the following steps: 1. Identify real-time bandwidth demand for each link. 2. Identify data bandwidth demand for each link. 3. Assess available bandwidth for each link. If required, upgrade. 4. Determine enterprise QoS classification approach. 5. Determine enterprise QoS technology for the LAN network. 6. Determine enterprise QoS technology for WAN, and select WAN vendors. 7. Deploy QoS in all components of the network to support real-time traffic. 8. Test WAN support for enterprise QoS approach. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

54 Call Admission Control 9. Verify network support of real-time traffic with synthetic testing. 10. Implement a bandwidth management methodology. 11. Implement real-time network monitoring capability. 12. Create, negotiate, and sign an enterprise SLA. The following Avaya Video Conferencing Solution endpoints support QoS for video calls: Avaya one-x Communicator Avaya 1000 series endpoints Avaya Desktop Video Device Polycom endpoints For more information on preparing the customer network for video conferencing, see Chapter 11.0, "Checklist," in Supporting Real-time Traffic: Preparing Your IP Network for Video Conferencing, which is a white paper from Polycom.To access this white paper, search the Internet for Supporting Real-time Traffic: Preparing Your IP Network for Video Conferencing. Selecting the default maximum call rate Avaya recommends you ensure the following when you select the default maximum call rate: Deployment rate of 384Kbps. Allocation of 20% of the bandwidth for IP protocol overheads. 4Mbps bandwidth support for high-definition endpoint calls. Enter change ip-codec-set, and enable the Allow Direct IP Multimedia field. In Communication Manager 6.1, there are two options: normal users and priority users. Ensuring adequate inter-pbx network connectivity rates Avaya recommends you ensure the following inter-pbx network connectivity rates: Connection establishment in less than 150ms end-to-end with one-way delay. Packet loss of less than 1%. If you do not ensure the inter-pbx network connectivity rates, customers will experience slow call establishment rates. Multimedia calls have five times more call establishment signaling messages than audio calls. 54 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

55 Optimizing video on the network Ensuring adequate VPN connectivity rates Avaya recommends you ensure the following VPN connectivity rates for Avaya one-x Communicator: Connection establishment in less than 150ms with one-way delay to Communication Manager. Packet loss of less than 1%. Jitter rate of less than 20ms. Avaya one-x Communicator does not support automatic bit rate downgrade to compensate for packet loss or perform ping test for video assessment. To achieve the best possible video experience in existing network conditions, you must manually reduce the call rate and again try to establish the call. If the network receives excessive video update requests, Communication Manager may disable video in calls. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

56 Call Admission Control Feature interactions and limitations Table 20: Feature interactions and limitations on page 56 provides a list of possible interactions and limitations among features. Table 20: Feature interactions and limitations Limitations and interactions Limitations of Call Recording, Whisper Page, or Service Observing on video calls Limitations of video conference calls with one audio endpoint Mitigation If the customer uses features on video calls, ensure that you enable the Ad hoc Video Conferencing feature: Call Recording Whisper Page Service Observing If you do not enable the Ad hoc Video Conferencing feature, to use features such as Call Recording, Whisper Page and Service Observing, you can establish only audio calls. Each feature is accounted as 1 audio participant in a call. To ensure video on video conference calls between endpoints when one of the endpoints is not a video endpoint, enable the Ad hoc Video Conferencing feature. If you do not enable the Ad hoc Video Conferencing feature in video conference calls when one of the endpoints is not a video endpoint, you can establish only audio calls. This limitation does not apply if you try to establish a video call to auxiliary devices over trunk lines. For example, dedicated video conferencing devices. 56 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

57 Feature interactions and limitations Table 20: Feature interactions and limitations Limitations and interactions Limitations of video conference calls when insufficient ad hoc video conferencing ports are available Bandwidth is unavailable Mitigation If insufficient ad hoc video conferencing ports are available, the following limitations apply on video conference calls: If ad hoc video conferencing ports are unavailable, and a call involves 2 video endpoints, participants receive the video stream. If a call involves more than 2 video endpoints, participants do not receive the video stream. You can involve the maximum supported number of audio endpoints. For more information on the limitations of video conference calls, note the following points: Session Manager processes an audio endpoint that establishes a call over a video-enabled trunk as a video endpoint. Session Manager processes a video endpoint that establishes a call over a nonvideo-enabled trunk as an audio endpoint. A nonvideo device registered on a video endpoint does not count as a video endpoint. A video endpoint with video disabled or muted counts as a video endpoint. In a video conference call, if a participant receiving a video stream adds a participant with a video endpoint, the video stream is disabled. To enable the video stream, reduce the number of participants with video endpoints to a maximum of 2 participants. When the total available bandwidth is being used by endpoints on existing calls, Avaya 1000 series endpoints display the following message: No Bandwidth Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

58 Call Admission Control 58 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

59 Chapter 4: Administering conferencing bridges and trunks on Avaya Video Conferencing Solution This chapter provides information on administering bridges and trunks to connect the components of Avaya Video Conferencing Solution. Administering SIP trunks on Communication Manager To administer SIP trunks on Communication Manager, perform the following steps: 1. On the SAT screen, type change signaling group <group number>, and set the following parameters: Group Type: SIP The system displays a screen for SIP groups. Transport Method: TLS Near-end Node Name: The name of the IP interface at the near-end of the SIP trunk signaling group. The values are PROCR or C-LAN. Far-end Node Name: The name of the node that you administered as the SIP proxy server in Step 5 of Administering call routing for SIP endpoints on Communication Manager. Near-End Listen Port: The recommended TLS port value of Far-end Listen Port: The recommended TLS port value of Far-end Network Region: If you want to fulfill the following two conditions, enter the region of the SIP proxy: The SIP proxy server that you administered in Step 5 of Administering call routing for SIP endpoints on Communication Manager to use the codec set. The parameters that you specified for an IP network region to be different from the parameters of the LAN IP interface. Far-end Domain: Enter the IP address that represents your SIP Domain. For example, to route SIP calls within your enterprise, enter the domain assigned to your proxy edge server. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

60 Administering conferencing bridges and trunks on Avaya Video Conferencing Solution For external SIP calls, enter the domain name of your SIP service provider. DTMF over IP: rtp-payload SIP Session Establishment Timer: Avaya recommends 3 minutes. Enable Layer 3 Test: N If you retain the default value N, Communication Manager performs ping tests. If you enter the value Y, Communication Manager performs the OPTIONS test. This field is optional. IP Early Media: Change the value of this field to Y. 2. On the SAT screen, type change system paramters features, and set the following parameters: DID/Tie/ISDN/SIP Intercept Treatment: attd Trunk-to-Trunk Transfer: restricted If you set the value of Trunk-to-Trunk Transfer to restricted, it averts toll fraud. 3. On the SAT screen, type change trunk group, and set the following parameters: Group Type: SIP The Change Trunk Group screen displays fields that correspond to SIP groups. If you enter SIP, Communication Manager changes the fields of other corresponding screens. Call detail recording:y/n To enable CDR reports, enter Y. SIP calls may generate a very large number of CDR reports. Signaling Group: The number of the SIP signaling groups that you administered. Number of Members: The number of SIP trunks that belong to the current group. The value range of this parameter is between 0 to 255. Based on the value you assign, Communication Manager automatically populates the value of Group Member Assignments on the Trunk Group page and corresponding pages. You cannot individually administer group members. Members of an administered group share the same characteristics. The total number of SIP trunks you specify for the groups must not exceed the number you administered in Maximum Administered SIP Trunks on the System Parameters Customer-Options screen. Group Type: SIP 60 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

61 Administering a video bridge on Communication Manager Administer other fields on this screen according to the customer network requirements. Numbering Format: Based on customer requirement, enter one of the following formats: Public Private unk-pvt Unknown Administer other parameters on this screen according to the customer network requirements. Repeat step 3 for each SIP trunk group you need to administer. Administering a video bridge on Communication Manager To configure a video bridge, type add video-bridge <bridge number>, and set the following parameters: Name: The bridge name Max Ports: The number of ports that are available for use Trunk Groups: The administered incoming or outgoing ISDN H.323 or SIP trunk groups added to the media gateway controller. Far End Resource Info: N ID Range: Corresponds to the actual conference ID numbers of meeting rooms configured on the media gateway controller. Administering a direct routing gatekeeper This section describes how to configure direct routing gatekeepers, such as SE200, PathNavigator, and Polycom CMA. Prerequisites Before administering a direct routing gatekeeper, note the following recommendations: Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

62 Administering conferencing bridges and trunks on Avaya Video Conferencing Solution Calls from Communication Manager to the direct routing gatekeeper must route through an outbound route pattern, ISDN tie trunk group, and H.323 LRQ signaling group. Calls from the direct routing gatekeeper to Communication Manager must route through a second incoming ISDN tie trunk group, and either one incoming H.323 signaling group with an unspecified far-end or many incoming-per-ept H.323 signaling groups. Dial plan analysis, UDP, and AAR analysis must route the direct routing gatekeeper extension range to the outbound trunk group. Communication Manager manages the direct routing gatekeeper inter-gatekeeper LRQ administration for extensions. Ensure you have the following information: The IP codec sets you want to use The IP network regions you want to use The IP address of the IP board of the gatekeeper Adding an entry for the direct routing gatekeeper in the IP Node Names screen To add an entry for the direct routing gatekeeper in the IP Node Names screen, perform the following steps: 1. On the SAT screen, type change node-names ip, and set the following parameters: Name: The name of the system. IP Address: The IP address of the IP board of the system. 2. Navigate to page 2 of the screen, and set the following parameters: Allow Direct-IP Multimedia: Y Maximum Call Rate for Direct-IP Multimedia: This parameter represents the combined audio and video transmission rate or receive rate for normal video calls. You can use this setting to limit the amount of bandwidth used for normal video calls. For example, if you select 384Kbps, the audio stream and video stream of normal video calls will not exceed a maximum bandwidth usage of 384Kbps. Maximum Call Rate for Priority Direct-IP Multimedia: This parameter represents the combined audio and video transmission rate or receive rate for priority video calls. You can use this setting to limit the amount of bandwidth used for priority video calls. For example, if you select 384Kbps, the audio and video stream of priority video calls will not exceed a maximum bandwidth usage of 384Kbps. 3. To administer the gatekeeper in its own network region, on the SAT screen, type change ip-network-region <region number>, and set the following parameters: Intra-region IP-IP Direct Audio: Y 62 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

63 Administering a direct routing gatekeeper If the customer configuration uses only Polycom HDX endpoints registered to the Polycom CMA gatekeeper, set Intra-region IP-IP Direct Audio to Y. If the customer configuration includes multiple endpoints, such as Polycom VSC endpoints and Polycom V series endpoints, that are connected to multiple gatekeepers, such as the Polycom SE200 gatekeeper and the Path Navigator gatekeeper, set Intra-region IP-IP Direct Audio to N. Inter-region IP-IP Direct Audio: Y If the customer configuration includes only Polycom HDX endpoints registered to the Polycom CMA gatekeeper, set Inter-region IP-IP Direct Audio to Y. If the customer configuration includes multiple endpoints, such as Polycom VSC endpoints and Polycom V series endpoints, that are connected to multiple gatekeepers, such as the Polycom SE200 gatekeeper and the Path Navigator gatekeeper, set Inter-region IP-IP Direct Audio to N. Security Procedures 1: any-auth Video Norm: The amount of bandwidth that you want to allocate to the normal video bandwidth pool for each IP network region. Video Prio: The amount of bandwidth that you want to allocate to the priority video bandwidth pool for each IP network region. Video Shr: Specify whether the normal bandwidth video pool can be shared for each link between IP network regions. Adding signaling groups to the direct routing gatekeeper To add signaling groups to the direct routing gatekeeper, perform the following steps: 1. To add an outbound signaling group to the direct routing gatekeeper, on the SAT screen, type add signaling-group <signaling group>, and set the following parameters: Group Type: H.323 IP Video: Y Priority Video: Y If you set Priority Video to Y, all incoming calls receive priority video transmissions. Near-end Listen Port: 1719 LRQ Required: N RRQ Required: N Far-end Node Name: The name of the system Far-end Listen Port: 1719 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

64 Administering conferencing bridges and trunks on Avaya Video Conferencing Solution Far-end Network Region: IP network region Direct IP-IP Audio Connections: Y IP Audio Hairpinning: N 2. To add an inbound signaling group to the direct routing gatekeeper, on the SAT screen, type add signaling-group <chosen signaling group>, and set the following parameters: Group Type: H.323 IP Video: Y Priority Video: Y If you set Priority Video to Y, all incoming calls receive priority video transmissions. Near-end Listen Port: 1720 LRQ Required: N RRQ Required: N Far-end Network Region: IP network region Direct IP-IP Audio Connections: Y IP Audio Hairpinning: N Adding trunk groups to the direct routing gatekeeper To add trunk groups to the direct routing gatekeeper, perform the following steps: 1. To add an outbound trunk group to the direct routing gatekeeper, on the SAT screen, type add trunk-group <chosen trunk group>, and set the following parameters: Group Type: ISDN Carrier Medium: IP On page 5 of the Trunk Group screen, add members to the trunk group. 2. To add an inbound trunk group to the direct routing gatekeeper, type add trunk-group <chosen trunk group>, and set the following parameters: Group Type: ISDN Carrier Medium: IP On page 5 of the Trunk Group screen, add members to the trunk group. 64 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

65 Administering a direct routing gatekeeper Modifying signaling groups to the direct routing gatekeeper To modify signaling groups to the direct routing gatekeeper, perform the following steps: 1. To modify the outbound signaling group to the direct routing gatekeeper, on the SAT screen, type change signaling-group <chosen signaling group>, and set the following parameter: Trunk Group for Channel Selection: The trunk group 2. To modify the inbound signaling group to the direct routing gatekeeper, type change signaling-group <chosen signaling group>, and set the following parameter: Trunk Group for Channel Selection: The trunk group Creating a route pattern for the direct routing gatekeeper To create a route pattern that points to the trunk group, on the SAT screen, type change route-pattern <route patten>, and set the following parameter: Grp No: The number of the trunk group Adding an H.323 video signaling group To add an H.323 video signaling group for each endpoint that the direct routing gatekeeper manages, perform the following steps: 1. On the SAT screen, enter add signaling-group <chosen signaling group>, and set the following parameters: Group Type: H.323 LRQ Required: Y RRQ Required: N 2. For endpoints that do not support third-party pause, include the signaling group in an IP network region, and set the following parameter: direct-ip inter-region: Disabled Modifying the inbound trunk group to the direct routing gatekeeper Modify the inbound trunk group to include one trunk member for each endpoint that the direct routing gatekeeper manages. Table 21: Signaling group 51 on page 65 provides an example of a signaling trunk group Table 21: Signaling group 51 SIGNALING Group Number 51 Group Type h.323 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

66 Administering conferencing bridges and trunks on Avaya Video Conferencing Solution Remote Office n Max number of NCA TSC 0 SBS n Max number of CA TSC: 0 IP Video y Priority Video y Trunk Group for NCA TSC Trunk Group for Channel Selection 51 TSC Supplementary Service Protocol a Network Call Transfer n T303 Timer(sec) 10 H.245 DTMF Signal Tone Duration(msec) - Near-end Node Name procr Far-end Node Name CMA4000 Near-end Listen Port 1719 Far-end Listen Port 1719 Far-end Network Region 2 LRQ Required y Calls Share IP Signaling Connection n RRQ Required n Media Encryption n Bypass If IP Threshold Exceeded n H.235 Annex H Required n DTMF over IP out-of-band Direct IP-IP Audio Connections y Link Loss Delay Timer(sec) 90 IP Audio Hairpinning n Enable Layer 3 Test n Interworking Message PROGress 66 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

67 Administering the Polycom RMX SIP entity link on Session Manager H.323 Station Outgoing Direct Media n DCP/Analog Bearer Capability 3.1kHz Setp - Administering the Polycom RMX SIP entity link on Session Manager To administer the Polycom RMX SIP entity link on Session Manager, perform the following steps: 1. Log in to the System Manager Web console. 2. From the options on the left pane of the page, click Routing. The Routing option expands to display a list of options. 3. To create a SIP entity, click SIP Entities. The SIP Entities Details page opens. 1. Under General, set the following parameters: Name: RMX FQDN or IP Address: <IP address of the Polycom RMX signaling host> Type: Gateway SIP Time B/F (in seconds): 4 2. Under Entity Links, add an entity link between Session Manager and Polycom RMX. 4. To create an entity link, click Entity Link. The System Manager Web console opens the Entity Links page. To add an entity link between Session Manager and Polycom RMX, set the following parameters: Name: RMX SIP Entity 1: SM Protocol: TCP Port: 5060 SIP Entity 2: RMX Port: 5060 Click Trusted. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

68 Administering conferencing bridges and trunks on Avaya Video Conferencing Solution 5. To create a routing policy, click Routing Policy. The Routing Policy Details page opens. 1. To create a new routing policy, click New. 2. Under General, set the following parameter: Name: Policy to RMX 3. Under SIP Entity as Destination, click Select. 4. Select the entity link to Polycom RMX. 5. Click Commit. 6. To create dial patterns, click Dial Patterns. The System Manager Web console opens the Dial Patterns Details page. 1. To create new dial patterns, click New. 2. Under General, set the following parameters: Pattern: Extension to be dialed to connect to Polycom RMX Min and Max: Dial string length for the Polycom RMX extension 3. Under Originating Locations and Routing Policies, click Add. Select the originating location. Select the routing policy Policy to Polycom RMX. Click Select. 4. To commit the dial patterns, click Commit. Administering the Polycom RMX H.323 trunk to Communication Manager This section describes how to administer the Polycom RMX H.323 trunk to Communication Manager. Prerequisites Before you administer the Polycom RMX trunk, you must know the following information: The IP codec sets that you want to use. The IP network regions that you want to administer. The IP address of the Polycom RMX signaling host. 68 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

69 Administering the Polycom RMX H.323 trunk to Communication Manager The IP address of C-LAN. For information on how to configure the Polycom RMX MCU, see Polycom RMX Installation and Configuration Guide. Adding an entry for the Polycom RMX trunk in the IP Node Names screen on Communication Manager To add an entry for the Polycom RMX trunk in the IP Node Names screen, on the SAT screen, type change node-names ip, and set the following parameters: Name: The name for the Polycom RMX bridge IP Address: The IP address of the signaling host of the Polycom RMX bridge. Adding a two-way trunk group for the Polycom RMX trunk on Communication Manager To add a two-way trunk group for the Polycom RMX trunk, on the SAT screen, type add trunk-group <chosen trunk group>, and set the following parameters. Group Type: ISDN Set Direction: Two-way Carrier Medium: H.323 Service Type: Tie Adding a signaling group for the Polycom RMX trunk To add a signaling group for the Polycom RMX trunk, type add signaling-group <chosen signaling group>, and set the following parameters: Group Type: H.323 IP Video: Y Priority Video: Y If you set Priority Video to Y, all incoming calls receive priority video transmissions. Trunk Group for Channel Selection: The two-way trunk Near-end Node Name: PROCR or C-LAN Near-end Listen Port: 1720 LRQ Required: N Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

70 Administering conferencing bridges and trunks on Avaya Video Conferencing Solution RRQ Required: Y ARQ Required: Y Enable Layer 3 Test: N Far-end Node Name: The name of the Polycom RMX bridge Far-end Listen Port: 1720 Far-end Network Region: 1720 Calls Share IP Signaling Connection: N Direct IP-IP Audio Connections: Y IP Audio Hairpinning: N Administering the Polycom RMX H.323 trunk to Polycom CMA To administer the Polycom RMX H.323 trunk to Polycom CMA, on Polycom RMX Manager, perform the following steps 1. Log in to Polycom RMX Manager. 2. Change the IP network service. 3. Click Gatekeeper from the options on the left, and set the following parameters: IP Network Type: H.323 & SIP Primary Gatekeeper IP address or Name: <IP address or name> MCU Prefix in Gatekeeper: <customer preferred prefix> MCU Prefix in Gatekeeper is required only if you are registering to Polycom CMA. Refresh Registration every: 30 seconds 4. Click the SIP Servers screen from the options on the left, and set the following parameters: IP Network Type: H.323 & SIP Refresh Registration every: 3600 seconds Transport Type: TCP SIP Servers > Server IP: <SIP proxy server IP address> SIP Server > Server Domain: <customer domain> Outbound Proxy Servers > Server IP: <customer SIP proxy server IP address> 70 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

71 Administering the Polycom RMX video conferencing bridge Outbound Proxy Servers > Port: 5060 Administering the Polycom RMX video conferencing bridge To administer the Polycom RMX video conferencing bridge, perform the following steps: 1. Install Polycom RMX and connect it to your network. If required, upgrade the Polycom RMX software. 2. Log in to Polycom RMX Manager. 3. From the Setup menu, select the System Parameters tab. Under MCMS_PARAMETERS_USER, set the following parameters: MCU_DISPLAY_NAME: Polycom RMX ENABLE_AUTO_EXTENSION: YES NUMERIC_CONF_ID_LEN: 5 CP_REGARD_TO_INCOMING_SETUP_RATE: NO NUMERIC_CONF_ID_MAX_LEN: 8 NUMERIC_CONF_ID_MIN_LEN: 4 TERMINATE_CONF_AFTER_CHAIR_DROP: NO H323_FREE_VIDEO_RESOURCES: NO ENABLE_SIP_PEOPLE_PLUS_CONTENT: YES SUPPORT_HIGH_PROFILE: NO 4. Under CS_MODULE_PARAMETERS, set the following parameter: H245_TUNNELING = YES 5. On the Default IP Server Properties dialog, create and configure an H.323 IP service with Communication Manager settings, such as the C-LAN IP address. Ensure that the MCU prefix and aliases match the dial plan that you configured on Communication Manager. Using a status signaling group, confirm that Polycom RMX has registered. 6. On the Default Video Properties dialog, configure the conference profile with the line rate, codec settings, and the high definition video switching option. Multiple meeting rooms can use this profile. 7. Create a meeting room to use a test direct dial conference ID. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

72 Administering conferencing bridges and trunks on Avaya Video Conferencing Solution Administering Polycom RMX for SIP and H.323 dual connectivity Prerequisites Before you administer Polycom RMX for SIP and H.323 dual connectivity, ensure you have administered the following trunks: 1. An H.323 trunk between Polycom RMX and Communication Manager. For more information on how to administer an H.323 trunk between Polycom RMX and Communication Manager, see Administering the Polycom RMX H.323 trunk to Communication Manager on page A SIP entity link between Polycom RMX and Session Manager. For more information on how to administer a SIP entity link between Polycom RMX and Session Manager, see Administering the Polycom RMX SIP entity link on Session Manager on page 67. Verifying the configuration of Communication Manager for H.323 and SIP dual connectivity To verify the configuration of Communication Manager for SIP and H.323 dual connectivity, on the SAT screen, type display system-parameters customer-options, and verify that the following optional features are enabled: System Parameters screen page Parameters 2 Maximum Video Capable Stations Maximum Video Capable IP Softphones Maximum Administered SIP Trunks 3 ARS/AAR Partitioning 5 Time of Day Routing If the optional features are not enabled, to acquire a license to enable the optional features, contact your Avaya sales representative. 72 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

73 Administering Polycom RMX for SIP and H.323 dual connectivity Verifying the configuration of Polycom RMX for H.323 and SIP dual connectivity To verify the configuration of Polycom RMX for H.323 and SIP dual connectivity, log in to Polycom RMX Manager, and perform the following steps: 1. From the menu, click Administration > System Information. Polycom RMX Manager displays the System Information dialog box. 2. In the list against Polycom Partners, verify that an entry for Avaya exists. If an entry for Avaya does not exist in the list against Polycom Partners, to acquire a license to configure Avaya in the list, contact your Polycom sales representative. 3. From the left pane on the page, under RMX Management, click Rarely Used > IP Network Services. Polycom RMX Manager displays the list of IP network services options. 4. From the list of IP network service options, right-click the default H.323 IP network services option, and select Properties. Polycom RMX Manager displays the IP Network Services Properties dialog box. 5. In the IP Network Services Properties dialog box, verify the following parameters: From the options on the left, click IP, and verify the following parameter: IP Network Type: H.323 & SIP From the options on the left, click Gatekeeper, and verify the following parameters: Primary Gatekeeper IP Address or Name: <IP address of PROCR or C-LAN of Communication Manager> MCU Prefix in Gatekeeper: <Dialed string of Polycom RMX> Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

74 Administering conferencing bridges and trunks on Avaya Video Conferencing Solution From the options on the left, click SIP Servers, and verify the following parameters: SIP Server: Specify SIP Server Type: Generic Transport Type: TCP or UDP SIP Servers Service IP Address or Name: <Session Manager SIP Entity IP Address> Server Domain: <Session Manager SIP Domain> Port: 5060 Outbound Proxy Servers Server IP Address or Name: <Session Manager SIP Entity IP Address> Port: 5060 Verifying the configuration of Session Manager for H.323 and SIP dual connectivity To verify the configuration of Session Manager, log in to System Manager, click Elements > Routing > SIP Entities, and verify the following parameters: Name: RMX FQDN or IP Address: <Signaling IP address of Polycom RMX> SIP Link Monitoring: Link Monitoring Disabled Administering route patterns for SIP and H.323 dual connectivity on Communication Manager To administer route patterns for SIP and H.323 dual connectivity on Communication Manager, perform the following steps: 1. To create a route pattern for H.323-based calls to Polycom RMX, on the SAT screen, type change route-pattern <number of the H.323 trunk group>, and set the following parameters: Pattern name: <Name of the route pattern for the H.323 trunk group> Grp No: <Number of the H.323 trunk group> FRL: 0 DCS/QSIC Intw: n IXC: user BCC Value W: y y y y y n TSC: n 74 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

75 Administering Polycom RMX for SIP and H.323 dual connectivity CA-TSC Request: none ITC: Rest LAR: none 2. To create a route pattern for SIP-based calls to Polycom RMX, on the SAT screen, type change route-pattern <number of the SIP trunk group>, and set the following parameters: Pattern name: <Name of the route pattern for the SIP trunk group> Grp No: <Number of the SIP trunk group> FRL: 0 DCS/QSIC Intw: n IXC: user BCC Value W: y y y y y n TSC: n CA-TSC Request: none ITC: Rest LAR: none 3. To assign a route pattern to Polycom RMX, on the SAT screen, type change aar analysis <number of the aar analysis table>, and assign a Partitioned Route Index to the Polycom RMX Dialed String. For example, P1. Administering route partitions on Communication Manager To administer route partitions on Communication Manager, perform the following steps: 1. On the SAT screen, type change partition-route-table <number of the partition route table>. 2. In Route Index 1, assign the number of the route pattern for the SIP trunk to Partition Group Number 1 (PGN1). 3. In Route Index 1, assign the number of the route pattern for the H.323 trunk to Partitioned Group Number 2 (PGN2). Figure 9: Administering route partitions on page 76 displays how to administer route partitions on Communication Manager. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

76 Administering conferencing bridges and trunks on Avaya Video Conferencing Solution Figure 9: Administering route partitions Administering endpoints to a specific partitioned group in Communication Manager To administer endpoints to a specific partitioned group in Communication Manager, perform the following steps: 1. To administer SIP-based calls to route through the SIP trunk, on the SAT screen, type change station <number of the SIP endpoint>, and enter a value for the COR field. 2. To administer H.323-based calls to route through the H.323 trunk, on the SAT screen, type change station <number of the H.323 endpoint>, and enter a value for the COR field. The value of the COR field must represent the Class of Restriction that includes the Partition Group Index that is assigned with the route to Polycom RMX. Either the value of the COR field for H.323 endpoints or the value of the COR field for SIP endpoints must be unique. For example, if you want the value of the COR field for H.323 endpoints to be unique, ensure that you enter this particular value of the COR field only for H.323-based endpoints. 76 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

77 Administering Polycom RMX to support H.320 ISDN calls Administering Polycom RMX to support H.320 ISDN calls Prerequisites Before you administer Polycom RMX, ensure the following: Procedure 1. Polycom RMX has the latest version of the software. 2. Polycom RMX is connected to the network. To administer Polycom RMX to support H.320 ISDN calls, perform the following steps: 1. In Polycom RMX Manager, click ISDN/PSTN Service Properties > Span Definition, and set the following parameters: Framing Side: User side User side is the default value. If you administer PBX on the network side, you must administer Polycom RMX on the user side. If you administer PBX on the user side, you must administer Polycom RMX on the network side. You can also administer PBX and Polycom RMX symmetrically. Line Coding Switch Type Parameters T1 default value E1 default value 2. Click Next. Framing SFSF FEBE Line Coding B8ZS HDB3 Switch Type AT&T 4ESS EURO ISDN The Phones dialog box appears on the screen. 3. Click Add. 4. To assign DID, click ISDN/PSTN Service Properties > Phones, and set the following parameters: First Number: The first number in the phone number range. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

78 Administering conferencing bridges and trunks on Avaya Video Conferencing Solution Last Number: The last number in the phone number range. MCU CLI: In dial-in connections, the MCU CLI indicates the MCU number dialed by the participant. In a dial-out connection, the MCU CLI indicates the MCU number as seen by the participant. 5. Click Save. 6. Click Conference Profile, and set the following parameters: Display Name: The name of the chosen conference profile Conference Profile: Select the Profile that the Entry Queue must use. ID: A unique number identifying the conferencing entity for dial-in calls. The default string length is 4 digits. Gateway Dial Out Protocols: Enable all options. ISDN/PSTN Access: Select the check box. ISDN/PSTN Service: ISDN/PSTN Service Dial in Number (1) : Polycom RMX automatically assigns a number from the selected ISDN/PSTN Network Service. To manually define a dial-in number, enter the required number from the dial-in number range defined for the selected Network Service. Dial in Number (2): Enter the required number from the dial-in number range defined for the selected Network Service. 7. Click Meeting Rooms, select the appropriate option, and set the following parameters: Display Name: The name and icon of the Meeting Room in the Polycom RMX Web client. Routing Name: The name used by all conferencing devices to connect to conferences that must be registered with the gatekeeper and SIP servers. ID: The Meeting Room ID. Polycom RMX communicates this ID to H.323 conference participants. The conference participants use this ID to dial in. Duration: The duration of the Meeting Room in hours using the format HH:MM. The default value is 01:00. Conference Password: The password participants enter to access the meeting room. Chairperson Password: The password the users enter to identify themselves as the Chairperson. Profile: The name of the profile assigned to the meeting room. 8. Click OK. 78 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

79 Administering the Polycom VBP-E video border proxy For an incoming call, an external user dials the Polycom RMX ISDN public number, and at the IVR prompt enters the E.164 extension of the SIP or H.323 extension. For an outgoing call, an internal user dials the ARS routing code followed by the ISDN public number. Administering the Polycom VBP-E video border proxy To configure the Polycom VBP-E video border proxy on Communication Manager, perform the following steps: 1. On the SAT screen, type change signaling group <group number>, and set the following parameters: IP Video: Y Near end listen port: 1720 Far end listen port: 1720 LRQ required: N RRQ required: N 2. On the SAT screen, type change trunk group <group number>, and set the following parameters: Group type: ISDN Direction: Two Way 3. On the SAT screen, type change ip network region <region number>, and set the following parameters: Intra region IP-IP Direct audio: N Inter region IP-IP Direct audio: N For information on how to configure Polycom VBP-E, see Administering Polycom VBP-E to support external video calls on page 80 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

80 Administering conferencing bridges and trunks on Avaya Video Conferencing Solution Administering Polycom VBP-E to support external video calls To administer Polycom VBP-E to support external video calls, perform the following steps: 1. On the Polycom VBP-E administration page, click Configuration Menu > Network, and set the following parameters: LAN Interface Settings IP Address: Change the default IP address to the Polycom VBP-E IP address. Subnet Mask WAN Interface Settings Static IP Address: Select the radio button for Static IP address. IP Address Subnet Mask: Enter the IP address of the default gateway. Network Settings Default Gateway: Enter the IP address of the WAN router. Primary DNS Server Secondary DNS Server 2. Click Configuration Menu > VoIP ALG > H.323, and set the following parameters: Gatekeeper Mode Select the radio button for Embedded gatekeeper mode. LAN/Subscriber-side Gatekeeper mode settings LAN/ Subscriber-side Gatekeeper address: The IP address of the gatekeeper 3. Click Configuration Menu > VoIP ALG > H.323 > Neighboring. The configuration menu displays the Prefix Routing table. 4. To enable prefix routing, set the displayed parameters, and review the following parameters: Action: Indicates whether to add or edit the prefix routing rule. Prefix: Specifies the prefix pattern to match against the dialing string. Index: Determines the order in which the Prefix table and Gatekeeper Neighboring table scan rules. To insert a rule between rules with consecutive indexes, select the index with the higher indicator. 80 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

81 Administering Polycom VBP-E to support external video calls Strip: Indicates whether to remove the matching prefix from the dialing string. Add: Specifies a string to prefix to the dialing string. Neighbor: Determines whether Polycom VBP-E sends an LRQ when the prefix matches. If you enable this parameter, the prefix becomes a neighboring statement. If you disable this parameter, Polycom VBP-E forwards the incoming Q.931 SETUP message to the given address without a preceding LRQ. When Polycom VBP-E forwards the incoming Q.931 SETUP message without a preceding LRQ, this is sometimes referred to as prefix-routing. Neighbor provides interoperability with other gatekeepers that may not accept the Q.931 SETUP message without a preceding LRQ. Polycom VBP-E alias manipulation Polycom VBP-E does not support dialing SIP calls with the IP address. You must map the E.164 number to the IP address of the endpoint. H.323 endpoints use alias manipulation to dial calls with the IP address. Most H.323 endpoints cannot dial calls with the ANNEX O method using the format methods. H.323 endpoint keypads may not contain symbols like at the rate and period (.) To dial a call with an IP address, the H.323 endpoints with a pound sign (#), and a period (.) with an asterisk (*). For example, to dial a call with the IP address from an H.323 endpoint, the user must dial 1234#1*1*1*1. Polycom VBP-E translates the dialed string as an ANNEX O method. To administer the Polycom VBP-E Alias Manipulation feature, perform the following steps: 1. Click Configuration Menu > VoIP ALG > H.323 > Alias Manipulation. 2. Set the following parameters: Action: Indicates whether to add or edit prefix routing rule. Pattern: Specifies the pattern to match. Index: Determines the order in which the Prefix table and Gatekeeper Neighboring table scan rules. To insert a rule between rules with consecutive indexes, select the index with the higher indicator. Replace: Specifies the string that replaces the matched pattern. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

82 Administering conferencing bridges and trunks on Avaya Video Conferencing Solution 82 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

83 Chapter 5: Administering Communication Manager for video endpoints This chapter provides information on how to administer Communication Manager for video endpoints. Administering global settings on Communication Manager To administer global settings on Communication Manager, perform the following steps: 1. On the SAT screen, type change system-parameters customer-options, and set the following parameters: Maximum Video Capable H.323 Stations: The number of single point video H.323 endpoints. Maximum Video Capable IP Softphones: The number of video softphones. For example, Avaya one-x Communicator. Maximum Administered Ad-hoc Video Conferencing Ports: The number of Polycom RMX systems, and the maximum number of ports you can administer on the Polycom RMX systems. Enhanced Conferencing: Y IP Trunks: Y IP Stations: Y ISDN-BRI Trunks: Y ISDN-PRI: Y Multimedia Call Handling (Basic): N Multimedia Call Handling (Enhanced): N 2. On the SAT screen, type change ip-network-region, and set the following parameters: Intra-region IP-IP Direct Audio: Y Inter-region IP-IP Direct Audio: Y For network regions containing third-party gatekeepers, these values must be set to N for direct audio. All third-party endpoints may not support shuffling. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

84 Administering Communication Manager for video endpoints 3. On the IP Network Region screen, ensure that you have correctly configured all regions used by video endpoints. To authenticate, H.323 SECURITY PROFILES must contain any-auth for Polycom VSX series endpoints and Polycom HDX series endpoints. any-auth encompasses both challenge and pin-eke. Avaya IP Softphone and Avaya one-x Communicator use challenge. Polycom VSX series endpoints and Polycom HDX series endpoints use pin-eke. 4. Configure the IP Codec Set screen. Review all codec sets used by the customer. Review codec sets used by media, processors, trunk lines, and endpoints. Review inter-region codec sets. 5. On the IP Codec Set screen, ensure that the intra-region audio administration is correct. Review all codec sets used by the customer. Review codec sets used by media, processors, trunk lines, and endpoints. 6. On the IP Codec Set screen, ensure that you have correctly administered the inter-region audio. Review all codec sets used by the customer. Review codec sets used by media, processors, trunk lines, and endpoints. If adequate bandwidth is available, administer the codec set to be reused for intra-region calls. If limited bandwidth is available, administer a low bandwidth codec with the G.711 series codec set. If bandwidth is less, remove the G.711 series codec set, and disable video. 7. On the IP Codec Set screen, ensure that you have correctly administered audio for Polycom VSX endpoints. Review all codec sets used by the customer. Review codec sets used by media, processors, trunk lines, and endpoints. If codec sets for endpoints include the G.729 series codec set, you must define a new network region for Polycom VSX endpoints, and administer the G.729A series codec set for Polycom VSX endpoints Avaya video endpoints do not support media encryption for video. 8. On the IP Codec Set screen, ensure that you have correctly administered video. Review page 2 of the codec sets that video endpoints and video trunks use. Ensure Allow Direct-IP Multimedia is set to Yes. 84 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

85 Administering global settings on Communication Manager Ensure Maximum Call Rate for Direct-IP Multimedia is set for codec sets used for inter-region calls with limited bandwidth availability. 9. On the IP Network Region screen, ensure that you have administered Inter Region Video Bandwidth Management to be used across WAN links, and you have correctly administered total bandwidth limits for video. If you do not correctly administer the Inter Region Video Bandwidth Management feature, users experiences low quality video streams. Avaya recommends you implement cumulative bandwidth management for CAC processing. Communication Manager bandwidth management does not account for variable video headers. Ensure that you provide for bandwidth overheads of 20%. The total bandwidth utilization by video applications must not exceed 35% of the total available bandwidth. 10. On the SAT screen, type change station <station number>, and set the following parameters: IP Video: Y IP Video Softphone: Y You do not need to set IP Video Softphone for video endpoints, such as Avaya Desktop Video Device, Avaya 1000 series endpoints, and Polycom devices. Direct IP-IP Audio Connections: Y 11. On the SAT screen, type change signaling group <group number>, and ensure that you correctly administered video support for the group. Review the signaling group to Session Manager, the signaling group to Polycom MGC endpoints, the signaling group to the Polycom RMX MCU, and the signaling group to SE200 endpoints. Set the following parameters: IP Video: Y Direct IP-IP Audio Connections: Y Calls Share IP Signaling Connection: No 12. On the Signaling Group screen, ensure the network-region value corresponds to an IP codec set that supports video. 13. On the Signaling Group screen, ensure you have correctly provisioned DSCP tags. Ensure that DSCP parameters are provisioned for the network region to which endpoints are registered. For Communication Manager Release 4.0 and later releases, the following endpoints automatically obtain the configured DSCP parameters in the network region: Polycom VSX Release and later releases Polycom MGC Release and later releases Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

86 Administering Communication Manager for video endpoints You must manually provision DSCP tags for all endpoints. Ensure that DSCP tags are identical to DSCP tags provisioned in Communication Manager. 14. Ensure that the network region of Avaya IP Softphone is identical to the network region of the IP set. If the network region of the Avaya IP Softphone is not identical to the network region of the IP set, perform the following step: On the SAT screen, type change ip-network-map and set the Avaya IP Softphone to be mapped to the identical network region of the IP set. For shared control when using video, Avaya Video Conferencing Solution supports only via the server. 15. For each class of user with ad hoc video conferencing privileges, ensure that Ad-hoc Video Conferencing is set to Y. On the SAT screen, type change station <station number>, and set COS for appropriate users. Enabling video on Communication Manager To enable video on Communication Manager, perform the following steps: This procedure contains steps for the fields that you must change. For the rest of the fields, retain the default value. 1. On the SAT screen, type change system-parameters customer-options, and set the following parameter: Multimedia IP SIP Trunking: Y 2. To enable video for the SIP signaling group, on the Communication Manager SAT screen, type change signaling-group <group name>, and set the following parameters: IP Video: Y Direct IP-IP Video Connections: Y Initial IP-IP Direct Media: Y You do not need to modify Priority video options. For more details on Priority video options, see Avaya Video Conferencing Solutions Networking Guide on the Avaya Support Web site at 3. To identify the codec set, on the Communication Manager SAT screen, type display-ip-network-region, and navigate to the Inter Network Region Connection Management page. Ensure the dst rgn number is identical to the Far-end Network Region. Note the codec set number 86 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

87 Administering audio codecs on Communication Manager 4. To add video to ip-codec-set, on the SAT screen, type change ip-codec-set <codec set number used by the SIP signaling group>. Add or change the required audio codecs, and set the following parameters: Verify that the list of audio codecs includes a minimum of one audio codec that the video endpoints support. For example, G.722 or G.711. Allow Direct-IP Multimedia: Y Maximum Call Rate for Direct-IP Multimedia: 1024Kbps Maximum call rate of 1024Kbps allows 720p high definition video streaming. For higher resolution video streaming, you need to allocate more bandwidth. For more details on allocating bandwidth for multimedia calls, see Avaya Video Conferencing Solutions Networking Guide on the Avaya Support Web site at Maximum Call Rate for Priority Direct-IP Multimedia: 2048Kbps Maximum Call Rate for Priority Direct-IP Multimedia applies only to priority video users. For more details on allocating bandwidth for multimedia calls, see Avaya Video Conferencing Solutions Networking Guide on the Avaya Support Web site at Administering audio codecs on Communication Manager Administration of audio codecs in Communication Manager is independent of the type of CAC administered, the type of Session Manager administered, or the type of Communication Manager administered in Avaya Video Conferencing Solution. Administration of audio codec on Communication Manager is required for Communication Manager sequenced applications calls. Communication Manager selects the audio codec by finding a matching audio codec between the ip-codec-set list and the audio codec capabilities of the endpoints. Communication Manager processes the ip-codec-set list as the master list. Communication Manager searches for matching audio codecs in the same sequence that the audio codecs are listed in the master list. To partition bandwidth by geographic location, Communication Manager uses network regions. To administer CAC and associate an ip-codec-set, network regions are paired together. Each physical site is assigned a network region. Each network region is assigned separate ip-codec-sets for intra-region calls and for calls to remote network regions. To use endpoint network region mapping, audio codec selection requires a full mesh topology. Each physical site Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

88 Administering Communication Manager for video endpoints must be assigned a network region, and each network region must be paired with another network region. Bandwidth CAC: CAC works with a hub and spoke model to represent WAN connections for each physical site. The WAN represents the hub, and each physical site represents a spoke. A network region functions as the WAN. Administration of CAC and ip-codec-set is performed for each network region pair. Audio codec selection: Each network region pair administered in a full mesh topology is assigned an ip-codec-set. The ip-codec-set assigned to the network region pair determines the audio codec set list and the maximum multimedia bandwidth usage for calls. Mapping endpoints to a network region To map H.323 or SIP endpoints to a network region, on the ip-network-form, you must perform the following steps: 1. Add the IP address of the endpoint. 2. Add the target subnet. 3. Assign a network region to the subnet. If you do not map H.323 endpoints, the endpoints map to the network region assigned to the gatekeeper where the endpoints register. If you do not map SIP endpoints, the endpoints map to the network region determined by the near-end and far-end network region pair of the SIP signaling group. SIP endpoints are processed as if the SIP endpoints are mapped to Network Region 1. Calls between SIP endpoints are treated as calls within Network Region 1. Assigning audio codecs Separate ip-codec-set can be assigned for calls within a network region and calls between different network regions. For calls on audio endpoints within a network region, assign better quality audio codecs. Calls within a network region have access to sufficient bandwidth so these calls can be assigned audio codecs that require more bandwidth. For calls on audio endpoints between different network regions, assign compressed audio codecs. Calls between different network regions are limited by bandwidth availability so, to minimize bandwidth usage, these calls must be assigned compressed audio codecs that do not require more bandwidth. For sites with audio and video endpoints, assign the same audio codec list. For multimedia calls within a network region, if you assign a wideband audio codec, you can assign the same wideband audio codec to audio endpoints. Ensure that the audio endpoints support wideband codecs. 88 Administering Avaya Video Conferencing Solution 6.1 Issue 1 April 2012

89 Administering audio codecs on Communication Manager Administering audio codecs Communication Manager administered as Evolution Server supports H.323 and SIP endpoints. CAC administered on either Communication Manager or Session Manager can perform CAC processing. All endpoints must be mapped to a network region. Each Session Manager location must be mapped to a Communication Manager network region. The subnets must be added to Communication Manager and Session Manager. Communication Manager processes Session Manager as a component of the WAN network region that is administered through the SIP signaling group. The near-end network region is configured with the IP address of PROCR. To administer audio codecs to Communication Manager administered as Evolution Server, perform the following steps: 1. Assign a media gateway to Network Region Assign endpoints that are located with PROCR and media gateway to Network Region Assign the far-end network region as Network Region 100. Network Region 100 is an example of a virtual network region that represents WAN in the topology. 4. If you set CAC administered on Communication Manager to perform CAC processing, on the SIP signaling group, disable Priority Ip Video. - If you disable Priority Ip Video, the endpoint Class of Service determines priority users. - If you enable Priority Ip Video, unexpected results occur between usage of bandwidth of the priority bandwidth pool and bandwidth of the normal bandwidth pool. To enable/ disable Priority Ip Video through COS, perform the following steps: 1. On the SAT screen, type change cos-group <COS group number>. 2. Navigate to page 2 of the screen, and enable or disable Priority Ip Video, as shown in Figure 10: Enabling or disabling Priority Ip video through COS on page 89. Figure 10: Enabling or disabling Priority Ip video through COS 3. On the SAT screen, type change station <station number>. Administering Avaya Video Conferencing Solution 6.1 Issue 1 April

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