[Asterisk-video] Branch to test: sdpcleanup

[John Martin]

Hi All,
  As far as I can see sdpcleanup is working as expected (with one minor
fix required for sdp_add() that Olle has). RTCP also seems to work as
expected.

Regards, John

John Martin
John.Martin at AuPix.com
http://www.AuPix.com


-----Original Message-----
From: asterisk-video-bounces at lists.digium.com
[mailto:asterisk-video-bounces at lists.digium.com] On Behalf Of Olle E
Johansson
Sent: 06 June 2006 12:29
To: Discussion media support in Asterisk of video
Subject: [Asterisk-video] Branch to test: sdpcleanup

Friends,

Most of the work in the sdpcleanup branch is now done. I have worked  
together with John Martin to
fix a number of small things, described in earlier mails. We have not  
fixed it all, but focused on small
changes that will improve things for 1.4. It's too late for large  
architectural changes now, they have to
wait for 1.6.

Please help us test the sdpcleanup branch. Download it using svn

svn checkout http://svn.digium.com/svn/asterisk/team/oej/sdpcleanup  
sdpcleanup

Compile and install as usual.

Make calls between audio phones, between video phones, between audio- 
only phones
and video phones and report any bugs.

Changes in this branch are supposed to be:

- If we get a re-invite that we reject, don't change the rtp  
properties of the call
- If we get an audio-only call, don't offer video to the video phone
- Try to not offer a video stream on the outbound call that is not  
offered on the
   incoming call

There are still some problems with answering a video offer with only  
an audio
media stream, forgetting to deny the video stream in the SDP but it  
seems to work
with most clients. I've seem the same buggy behaviour in video phones  
too,
so we're at least compatible :-)

Thanks for taking your time to test this. If I get no error reports  
during the
coming two days, I'll go ahead and integrate this into svn trunk for  
1.4.

Regards,
/Olle

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