[Asterisk-Users] MeetMe with the V (video) option

[Matt Riddell]

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:neils at vipadia.com]
Sent: 24 November 2005 09:30
To: John Martin; matt.riddell at sineapps.com
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 400 at sip.vipadia.com. 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:call at vipadia.com
| www.vipadia.com

-- 
Cheers,

Matt Riddell
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