Dean Collins wrote: > who's done it? and how much money are they talking about? I've been > looking to pay for something like that for a while.
-----Original Message----- From: Neil Stratford [mailto:[EMAIL PROTECTED] Sent: 24 November 2005 09:30 To: John Martin; [EMAIL PROTECTED] Subject: Re: Fwd: [Asterisk-Dev] Chan_sip: video capabilities, call bandwidth and RTCP Hi John/Matt, I am replying to your emails to the Asterisk-Dev mailing list concerning Asterisk and video support. I agree with many of the points that you both raised and would really like to see the support for video in Asterisk improved - it does work, but there are limitations today and I would like to see Asterisk leading the way. (My background is academic/commercial research in the area of multimedia & QoS.) I would like to make you aware of some projects that I have been working on, and to ask if you would be interested in helping to fund any future development to see these projects to completion or to start new projects in this area. >> While I don't think they are ratified, most video UAs support the draft >> RFCs: >> >> - levin-mmusic-xml-media-control-02 - INFO fast updates and Asterisk *should* already support this RFC - I implemented it earlier this year and it is in CVS, and in 1.2. Unfortunately I have just noticed that a minor typo was introduced into the XML when it was integrated into CVS, so it doesn't currently work - which is why you probably didn't realize it existed. It is a single character typo and I'll be feeding the patch into the bug tracker. (from later in your email) - Other fast update mechanisms (the H.261 RTP FU for instance) I also implemented this, but it didn't make it into CVS. If you think it is still important I can revive that code. Multiparty Video Conferencing: I have 95% of a working solution for multi-party video conferencing in Asterisk (based on app_conference). You can test the current version by calling [EMAIL PROTECTED] It currently allows for up to 10 callers per conference, and switches the displayed video when you send DTMF - press 1 for caller 1 etc. (This is running on a test server - if it is not up or has an error, please let me know.) I am currently looking for additional funding to complete this work and enable me to recover some of the development costs so that we can release it as open source. Asterisk h263 file format generation: I have written a couple of modules for GStreamer (www.gstreamer.net) which allow it to generate Asterisk format h263 files. With a GStreamer command line you can now convert video files from other formats into h263 and wav files for playback using Asterisk. I believe that the modules are now in the latest CVS version of GStreamer, but if you would like patches to 0.9 let me know. RTSP Streaming integration: This is a new project which I may have funding to complete already, but if you are interested we may be able to accelerate development. H324m/SIP gateway: This is something that many people are interested in, but there has been little progress. I would really like to drive this project forward. If you are interested in any of these projects, or are looking for any other development work (or collaboration), please do not hesitate to contact me. We are based just outside of Cambridge in the UK. Thanks Neil Stratford -- Neil Stratford | Vipadia Limited | +44 1223 858 111 | sip:[EMAIL PROTECTED] | www.vipadia.com -- Cheers, Matt Riddell _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
