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