VoIP-Forum

This week, the Asterisk Video Task Force has had it’s first workshop in Paris. (Nov 20-22, 2006)

First of all, we really have to send a big THANK YOU to Inria, the French research institute, that
hosted the event. They have been great hosts and provided us with both food, power and
network connectivity!
A special thank you to Philippe Sultan at Inria for organizing all of this.

To summarize:

1.4 testing

We started with testing the 1.4 beta. Some notes

• Direct RTP media setup did not work as expected. During the setup,
  the payload types was not copied between the call legs, resulting
  in bad calls with failing media. When answering, the capabilities
  of the callee is not used when sending 200 Ok to the caller. We opened a branch called “earlyrtpfix” to try to fix this. Almost there.
• IAX video is broken. Currently, the capabilities fails and video disappears
  somewhere in the call setup. Even with that fixed, there is a problem with
  not sending the frame sequence number across the bridge, which may
  result in bad resequencing on the other side. Out of order frames will be
  renumbered as they where in perfect order… IAX2 also seems to have
  problems with video frames larger than the MTU and video frames that
  are transported in multiple RTP payloads.
• We could set up calls between several phones. Ojo videophones,
  Aupix, Grandstream, Wooksung, Counterpath Eyebeam, Ekiga
  and Kapanga was tested as well as some unbranded softphones
  in development.
• We do have several problems in the RTP subsystem, that we
  need to look into.

Future things discussed

• Direct transcoding from alaw to ulaw. Module developed during the session by Morten.
• AMR codec support
• Integrations with 3G, support for 3G videocalls in libpri/chan_zap
• Exchange of more complete codec capabilities during call setup
  (branch videocaps)
• T.140 realtime text support
• Development of a contrib/videotoolbox for conversions, ivr video prompts etc
• Integration with remote transcoders
• Integration with OpenMCU videoconferencing
• Neil’s videoconference switch for app_conference
• H.323 video support
• Videolan integration for streaming video - VideoOnHold
• Next meeting in a sunny place
  After discussion, sunny is optional, no rain or snow is mandatory

This meeting was really productive and has moved Asterisk forward. Meeting for a few days really
helps, especially when you have a strong focus - like “video”.

/Olle Johansson

During my recent tests with video phones (thanks Grandstream, Aupix and Foniris!) I have found out that we have a list of things to do. I have also found out that there are a lot of developers out there that have done it already - meetme with selectable video streams, chan_local with video and other
patches that we need to incorporate into Asterisk. Smaller changes now for 1.4, bigger changes for 1.6.

The first batch of changes was included in the development tree today. More will come. As of today, Asterisk will handle calls between audio and video phones a bit better than before. We have also included work by Fredrik Olsson and John Martin to improve our support of RTCP, very important for video phones.

In order to open up a forum for those of you that want to work with Video, SIP and Asterisk, I have set up a mailing list for the AVTF - Asterisk Video Task Force!
I have no knowledge of video codecs, standards and other strange stuff, so I rely on the community there.
I can manage a branch for this work and come with input from a SIP standpoint.

It is ok to discuss IAX2 and Video as well - does it work with the jitterbuffer, does trunking work,
any room for improvements?

See you on the Asterisk-video mailing list! /Olle