Posted: Tue Jun 06, 2006 2:28 am Post 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
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.
Posted: Tue Jun 06, 2006 4:44 am Post subject: [Asterisk-video] Branch to test: sdpcleanup
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.
-----Original Message-----
From: asterisk-video-bounces@lists.digium.com
[mailto:asterisk-video-bounces@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
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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Posted: Tue Jun 06, 2006 6:53 am Post subject: [Asterisk-video] Branch to test: sdpcleanup
6 jun 2006 kl. 15.44 skrev John Martin:
Quote:
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.
Hopefully fixed the minor issue, so please run "svn update" and continue
testing :-)
Posted: Tue Jun 13, 2006 10:59 pm Post subject: [Asterisk-video] Branch to test: sdpcleanup
Hi all,
Quote:
Please help us test the sdpcleanup branch. Download it using svn
I've been doing some testing and have some feedback:
- You can get different behavior if debug is on or not - the ordering of
the if/else statement starting on L4635 of chan_sip.c seems to be the
culprit.
- The test for (numberofmediastreams > 2) on L4587 is perhaps a bit
harsh - some clients (such as the Microsoft RTC stack) offer extra
media, which we should ignore rather than treat as an error.
More later when I figure out why some of my calls are failing...
BTW - I have some reports of differences in behavior of the current 1.2
branch between 1.2.4 and 1.2.8 with video codec negotiation. Has anyone
else seen this?
Neil
--
Neil Stratford - http://www.vipadia.com/ - sip:call@vipadia.com
Vipadia Limited :: VoIP Research and Development
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