Administrator Guide: Video Support

Asterisk 1.2 and Communication Server 2

Introduction.

  • The Video Telephony support feature enables 2 SIP video endpoints to communicate each other using H263 protocol
  • This application note covers the following infrastructure:
    • 2 eyebeam soft phones with 2 web cams
    • 1 Escaux UCS
  • The users experience the following scenario:
    • eyebeam A calls eyebeam B extension, B answers the call.
    • A and B has an audio conversation.
    • A clicks on 'send video'. B can see A
    • B clicks on 'send video'. A can see B

Limitations

  • When using H.264 and Direct Media together, video stream might not be working after a call transfer (M9087)
  • After a call transfer done from an audio call, the video is not enabled (M8951)

Video Telephony support implementation

  • Install at least version 2.8 of the Asterisk-1.2 module with the following parameters. (Required)
    • SIP canreinvite : yes
    • SIP videosupport : yes
  • Click on 'Apply Change' and restart telephony service.
  • Configure 2 soft phones resources in Escaux UCS:
    • Example: SDX10001 and SDX10002 (resource type Softphone 1.00)
  • Configure 2 internal extensions for SDX10001 and SDX10002
    • 100 for SDX10001
    • 101 for SDX10002
  • On 2 PCs equipped with web cam, download eyebeam Enhanced, see the Administrator Guide Eyebeam 1.5
    • Configure H263: Right-Click, Options, Advanced, Video Codec. Disable H.263+ (1998). Codec Enabled: H.263
  • SDX10001 can now calls 101 and click on 'start video' (left panel). The video flow will be directly from one end point to another end point.

Video Telephony support implementation in a cluster

  • Install at least version 2.8 of the Asterisk-1.2 module with the following parameters. (Required)
    • SIP canreinvite : yes
    • SIP videosupport : yes
  • Enable H.263 codec in your Mesh SIP Trunk.
  • Click on 'Apply Change' and restart telephony service.
  • Configure 2 soft phones resources in Escaux UCS:
    • Example: SDX10001 and SDX10002 (resource type Softphone 1.00)
  • Configure 2 internal extensions for SDX10001 and SDX10002
    • 100 for SDX10001
    • 101 for SDX10002
  • On 2 PCs equipped with web cam, download eyebeam Enhanced, see the Administrator Guide Eyebeam 1.5
    • Configure H263: Right-Click, Options, Advanced, Video Codec. Disable H.263+ (1998). Codec Enabled: H.263
  • SDX10001 can now calls 101 and click on 'start video' (left panel). The video flow will be directly from one end point to another end point.

Video Support with Polycom VVX1500 Phones.

Asterisk 1.2

  • Install at least version 2.35.0 of the Asterisk-1.2 module with the following parameters. (Required)
    • SIP videosupport : yes
  • Install at least version 4.4.0 of Polycom Phone Module Support.
  • Use at least version 4.18 of PolycomVVX1500 resource with the following parameters. (Recommended)
    • SIP reinvite :
      • If you use h261 : can be set to yes or no
      • If you use h263 : must be set to no
    • Force specific codec : yes
    • First codec : a voice codec (ulaw, alaw, ...)
    • Second codec: a video codec (h261, h263)

If you want to use PUM with the VVX1500 you will also have to install at least version 1.12.0 of the Polycom Virtual Phone with the same parameters as the PolycomVVX1500 resource

Known limitations

  • When using h263, if you make transfer and that one of the phone doesn't support video, following transfer will loose proper video capability even if every phones in communication supports video.

Communication Server

  • Install at least version 2.3.3 of the Communication Server module with the following parameters. (Required)
    • SIP videosupport : yes
  • Install at least version 4.4.0 of Polycom Phone Module Support.
  • Use at least version 4.18 of PolycomVVX1500 resource with the following parameters. (Recommended)
    • SIP reinvite : yes
    • Force specific codec : yes
    • First codec : a voice codec (ulaw, alaw, ...)
    • Second codec: a video codec (h261,h263)

If you want to use PUM with the VVX1500 you will also have to install at least version 1.12.0 of the Polycom Virtual Phone with the same parameters as the PolycomVVX1500 resource

Compatibility with Helios 2N

  • Install at least version 2.3.3 of the Communication Server module with the following parameters. (Required)
    • SIP videosupport : yes
  • Use the resource Generic SIP Phone (SDX1) at version 1.07+ with the following parameters. (Recommended)
    • SIP reinvite : no
    • Force specific codec : yes
    • First codec : a voice codec (ulaw, alaw, ...)
    • Second codec: h264

You will also need to log on the Administration interface of the Helios 2N at the following address : http://ip_address_of_helios_device In the Advanced Settings, in the menu Video Codecs, you must provision those options :

  • Preferred video codecs
    • Choice 1 : H.264
  • Video codecs settings
    • Video resolution : CIF (352x288)
    • Frame rate : 10 fps
    • Video bitrate : 512 kbps
    • Video packet size : 1400
  • Advanced RTP settings
    • H.264 payload type (1) : 99

Notes

If you want to use the Helios 2N with a Polycom VVX500 or a VVX1500, the Polycom resource must be at least 4.22 and one of their video codec must be in h264.

If you want to use the Helios 2N with net.Buzz, the netBuzz resource (SDX4) must be at least 1.1.0 and one of their video codec must be in h264.

Compatibility with net.Buzz

  • Install at least version 2.3.3 of the Communication Server module with the following parameters. (Required)
    • SIP videosupport : yes

  • Install at least version 1.3.0 of the net.Buzz module.

  • Use the net.Buzz Resource (SDX4) at version 1.2.0 or higher with the following parameters. (Recommended)
    • SIP reinvite : Yes
    • Force specific codec : Yes
    • First codec : a voice codec (ulaw, alaw, ...)
    • Second codec: h264

  • On the net.Buzz clients, make sure that H.264 is listed in the "Enabled Codecs" for video.

Communication Server 3

Introduction

  • The following infrastructure is covered:
    • 1 Escaux UCS
    • Single video codec environment: H.264
    • Single and multi audio codecs environment: alaw, ulaw, g729, ...
    • SIP Direct Media set to re-invite in communication server and default settings for phones and Mesh Sip Trunks.

  • The following scenarios have been validated:
    • Basic video calls (between two video phones)
    • Basic mixed audio video calls (between audio and video phones)
    • Blind transfer between video phones
    • Blind transfer between mixed video an audio phones
    • attended transfer between video phones
    • attended transfer between mixed video an audio phones

Requirements

  • Modules:
    • Communication Server v3.3.0 or higher
    • Polycom Phone Support v5.0.0 or higher
    • net.Buzz v1.4.0 or higher

  • Resources:
    • Polycom VVX500 v5.1.0 or higher
    • Polycom VVX600 v5.1.0 or higher
    • Polycom VVX1500 v5.1.0 or higher
    • net.Buzz v1.2.0 or higher
    • MeshSipTrunk v1.14.0 or higher

Limitations

  • When using H.264 and SIP Direct Media re-invite, video stream might not be working after a call transfer. With no SIP Direct Media, this limitation does not apply. (M9087)
  • After a call transfer done from an audio call, the video is not enabled (M8951)

Video Telephony support implementation

  • Install the Communication Server module with the following parameters. (Required)
    • SIP Direct Media : re-invite
    • SIP videosupport : yes
  • Click on 'Apply Change' and restart telephony service.
  • Configure video phones resources with the following parameters:
    • SIP reinvite : default
    • Force specific codec : yes
    • First codec : a voice codec (ulaw, alaw, ...)
    • Second codec: (optional) a voice codec (ulaw, alaw, ...)
    • Third codec: h264
  • Configure internal extensions with these resources as primary phone.

Video Telephony support implementation in a cluster

  • Install the Communication Server module with the following parameters. (Required)
    • SIP Direct Media : re-invite
    • SIP videosupport : yes
  • Configure Mesh SIP trunk resources with the following parameters:
    • SIP reinvite : default
    • First codec : a voice codec (ulaw, alaw, ...)
    • Second codec: (optional) a voice codec (ulaw, alaw, ...)
    • Third codec: h264
  • Click on 'Apply Change' and restart telephony service.
  • Configure video phones resources with the following parameters:
    • SIP reinvite : default
    • Force specific codec : yes
    • First codec : a voice codec (ulaw, alaw, ...)
    • Second codec: (optional) a voice codec (ulaw, alaw, ...)
    • Third codec: h264
  • Configure internal extensions with these resources as primary phone.
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