[Asterisk-video] RE: AVTF - What's going on?

[Neil Stratford]

Hi everyone,

It's great to see so much activity around video.

Here are some projects that we've been working on:

- Videoswitch: (thanks to support from AuPix)
A multi-party video conference module based on app_conference. Each 
participant gets to see a single video stream at a time, which can be 
switched using either DTMF input from the user or Manager interface 
commands. You can use it for things like video conferencing, distance 
learning and video broadcast. There is usually a test version running at 
400 at sip.vipadia.com if you would like to play with it. We'll be making 
the code available soon.

- Fast Updates / RTCP / etc etc:
Various patches for fast updates and tweaks to RTCP for different 
Asterisk versions.

- GStreamer & Asterisk:
Modules for GStreamer (www.gstreamer.net) which allow it to generate 
Asterisk format h263 files. With a GStreamer command line you can 
convert video files from other formats into h263 and wav files for 
playback using Asterisk. I believe that the modules are now maintained 
in the GStreamer CVS.

Other stuff:

I've volunteered to talk about Asterisk and Video at the London 
Astricon. I'm intending to talk about what you can do today (what works) 
and what is missing (or broken), both at a high level and the technical 
details.

Neil

> -----Original Message-----
> From: Olle E Johansson [mailto:registry at webway.se] 
> Sent: 16 May 2006 10:13
> To: asterisk-video at edvina.net
> Subject: AVTF - What's going on?
> 
> This is stuff going on that I'm aware of:
> 
> * I am working on fixes for SIP/SDP in the sdpcleanup branch.  
> Grandstream and Foniris generously
>    donated video phones for me to test with. I am implementing some  
> checks if we really have
>    video on the incoming call before we offer it on the outbound  
> call. Seems stupid to offer
>    video without a stream. If we can fix this stuff and test it  
> quickly, I can get it into 1.4.
> 
> * John Martin contributed some patches for FMTP support that is  
> needed. Will open a branch
>    for this hopefully this week. This is too late for 1.4, but will  
> have to be part of 1.6. It will require
>    some architectural changes, and we're beyond freeze for that.
> 
> * John Martin is also doing work on H.323 and video
> 
> What else is going on codewise or should be added to the wish list?
> 
> /Olle
> 
> 
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-video
> 


-- 
Neil Stratford - http://www.vipadia.com/ - sip:call at vipadia.com
Vipadia Limited :: VoIP Research and Development