© 2007 Avaya Inc. All rights reserved. SIP Video with Communication Manager Chris Kendall.

© 2007 Avaya Inc. All rights reserved.

SIP VideoSIP Videowith Communication Managerwith Communication ManagerSIP VideoSIP Videowith Communication Managerwith Communication Manager

Chris Kendall

© 2007 Avaya Inc. All rights reserved.Avaya Proprietary

The addition of SIP to Avaya’s Video Telephony Solution provides the framework for the future of enterprise video conferencing

– Leveraging existing IETF open standards– Ensuring higher levels of interoperability with third parties– Maintaining a sophisticated feature set that is consistent and

compatible with today’s H.323 video solutionHighlights include:

H.323-SIP inter-working, with a common subset of telephony featuresCall admission control and video enablement policiesBandwidth management

– Priority video callers– Cumulative pools for audio and video– Call rate negotiation

Video Conferencing (scheduled and ad-hoc)

Overview DescriptionOverview DescriptionSIP Video


Solution viewSolution view


Solution ComponentsSolution ComponentsCM is the

– Feature server - extends many CM features to SIP phones thru OPTIM architecture

– Gateway – interworks SIP to all other protocols (H.323, ISDN, DCP, analog, etc..)

– Back-to-back user-agent (B2BUA)

SES contains the following components

– Proxy

– Registrar

– Event Server

– Personal Profile Manager (PPM)

– Location service (database)


Solution Components (continued)Solution Components (continued)Supported SIP Adjuncts

– Modular Messaging

– Voice Portal

– Meeting Exchange (version 5 with Video)

Supported SIP Endpoints– Avaya SIP Softphone (AST)

– 46xx SIP

– 96xx SPARK phone (AST)

– Toshiba SIP phone (AST)

– One-X mobile edition (CHAMP)

– Cisco and other third-party SIP phones

– SIP Video phones NEW!

NEW!


Meeting ExchangeMeeting ExchangeMeeting Exchange is the existing large scale SIP conferencing platform for CM

– Version 5 supports video conferencing of up to 16 simultaneous users.

– Meet-me, scheduled and ad-hoc conferences may be used

– Configured as a video-enabled trunk on CM

– Video-bridge configuration must be set for ad-hoc conferencing


SIP Video PhonesSIP Video PhonesMost Polycom & Tandberg room systems are dual-protocol H.323 and SIP

– Tandberg T150 (v4.1) has been tested extensively with CM in SIP

– For presentations and collaboration, H.323 is still recommended

There are many third party SIP phones out in the market, especially Softphones that can be connected to SES

– Some will work as OPTIM endpoints, i.e. “CM-routed” and others will not

Avaya’s next generation SIP Softphone (with video) is TBA


Jargon BusterJargon Buster

Lots of jargon and acronyms to digest…

– SIP: Session Initiation Protocol

– SDP: Session Description Protocol

– Dialog: A conversation/communication

– Session: Media (audio/video/data) between two entities

– AST: Advanced SIP Telephony

– SUSHI: Toshiba SIP Phone (supporting AST)

– B2BUA: Back-to-Back User Agent

– OPTIM/OPS: Off PBX/Premises Station

– OATS: Origination And Termination Service


High-level Call-flowHigh-level Call-flow

Edge ProxySIP phone-A

NJ CM

SIP phone-B

DR Home Proxy NJ Home Proxy

INV

ITE sip:b

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INV

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:b

INVI

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ip:b

INVITE sip:b

INV

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INV

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sip

:bIN

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DR CM


Why OPTIM?Why OPTIM?

Using OPTIM architecture, CM becomes a central point of control for video policy

– BWM, CAC, video and conferencing capabilities

As a B2BUA, CM will modify the SDP to enforce these policies

– Allow or deny video

– Restrict call-rates

– Allow/deny use of shared resources

As a SIP-H.323 gateway, CM bridges the protocol gap between different users and systems

– Don’t need to dial special addresses

– Don’t need to think about protocols or devices

– Telephony features “just work”

Video is still as easy as making a phone call


SIP – H.323 InteroperabilitySIP – H.323 InteroperabilityCM places particular requirements on endpoints for H.323 interoperability:

– RFC3890, for enterprise bandwidth management

– SIP endpoints to present their full capabilities in the SDP

– Supporting asymmetric SDP payload types is also required

Some SIP endpoints can be configured to support the functionality required by CM

For example, In Counterpath Eyebeam (1.5) you can dial ***7469 to reveal a hidden config UI and set the following options:

– Rtp:media:send_bandwidth_modifier = 1

– Media:sdp:specify_all_codecs_in_offer_answer = 1

– Media:sdp:force_describe_well_known_codecs = 1


Solution LimitationsSolution Limitations

This release has some limitations with CM as the inter-working SIP-H.323 gateway using video endpoints.

– SIP video endpoints are not supported dialing into the Polycom MGC (either H.323 or SIP)

– SIP video endpoints cannot view H.239 streams (e.g. Polycom “People plus Content”)

– Polycom SIP firmware is not currently supported when configured as an OPTIM endpoint

– Multipoint SIP room systems are not supported with CM

Future releases aim to resolve these limitations


SIP – Priority Video UsersSIP – Priority Video Users

Priority video is not signaled over SIP trunks

– So it works a bit differently

If Priority video is enabled and the SIP signaling group is terminating the call:

– The priority status of the originating caller is preserved

If Priority video is enabled and there is an incoming call on the SIP signaling group:

– The call gets promoted to priority

SIP sub-domains may therefore be used to separate priority users from regular users

– [email protected]

– [email protected]


SES AdministrationSES Administration

SES Administration is effectively unchanged from existing SIP audio deployments.

Primary consideration is whether a user is mapped to a media server extension (CM station) or not

Video calls are unmanaged when they are “pure SIP” and do not route via CM.

– Calling a H.323 user will still route the call through CM if the media-server address maps are configured

– This also holds true for calls involving OPTIM SIP users


CM Admin – LicensingCM Admin – Licensing

New Feature to be enabled in license file.

– Multimedia IP SIP Trunking? (FEAT_MMIP_SIP)

“Multimedia IP SIP Trunking?” can then be enabled in customer options


CM Admin – SIP Video TrunkCM Admin – SIP Video Trunk

Two new fields for video on the SIP signaling group.

– ‘IP Video’ and ‘Priority Video’

SIP trunk-group administration is unchanged


CM Admin – OPTIM stationsCM Admin – OPTIM stationsNew set type ‘4620SIP’ – should be used for all non-AST SIP endpoints

‘IP Video’ should be enabled, while ‘IP Softphone’ should be disabled


Common ProblemsCommon Problems

“The call fails to connect when calling from a SIP endpoint”– If response is “403 forbidden (Denial 11)”, check COR/COS, ip-network-region domain

– If response is “482 Loop detected”, check dial plan (UDP/AAR/EXT)

– If response is “488 Not acceptable”, check audio codec is supported

“I get no video when calling from a SIP endpoint”– Check licensing and capacities

– Check signaling group, network-region and codec-set admin

– Check station admin - ensure IP-direct audio enabled for station form page 2

– Check bandwidth is available

– Check endpoint enabled for video, and is offering bandwidth

– Due to protocol complexity, SIP users will not get video dialing into the H.323 MGC via CM

“My SIP endpoint wont register”– If response is “401 Unauthorized”, check password

– If response is “404 User not found”, check username and domain are correct,

– check that the SES configuration has been saved/updated

More items in the AVTS configuration checklist…


References and ResourcesReferences and Resources

SIP support in Communication Manager 4.0:

– http://support.avaya.com/elmodocs2/sip/245206_7.pdf

Installing and administering SES:

– http://support.avaya.com/elmodocs2/sip/03_600768_4.pdf

Date post:	31-Mar-2015
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© 2007 Avaya Inc. All rights reserved. SIP Video with Communication Manager Chris Kendall.

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