|Summary:||ASTERISK-17227: No RTP port update when SIP RE-INVITE is received|
|Date Opened:||2011-01-11 16:10:15.000-0600||Date Closed:||2011-06-07 14:00:53|
|Description:||Exactly the same issue as: https://issues.asterisk.org/view.php?id=15149 but those issue is closed, so I open new issue.|
When RE-INVITE comes to asterisk with new media port, asterisk sends 200 OK, and send RTP packets further to the old port instead of new port which received with RE-INVITE.
My asterisk version is: 188.8.131.52
Could you help me with this?
|Comments:||By: Leif Madsen (lmadsen) 2011-01-12 08:53:04.000-0600|
You need to provide debugging information. At a minimum:
* SIP trace from Asterisk
* SIP history
* Topology description
* Dialplan and scenario to reproduce issue
* End points involved (types of phones, PBXs, etc)
By: kondik (kondik) 2011-01-19 03:53:12.000-0600
The problem was solved.
RE-INVITE with SDP which comes to Asterisk had unsupported version for Asterisk, therefore asterisk ignore changes in SDP. However Asterisk send 200 OK for that RE-INVITE.
I think that in this situation asterisk should send another response than 200 OK...
Below the part of debug information:
[Jan 12 14:26:29] DEBUG: chan_sip.c:21891 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE
[Jan 12 14:26:29] DEBUG: chan_sip.c:8334 process_sdp: Processing session-level SDP v=0... UNSUPPORTED.
[Jan 12 14:26:29] DEBUG: chan_sip.c:8821 process_sdp_o: Call email@example.com responded to our reinvite without changing SDP version; ignoring SDP.
When I set ignoresdpversion=on in configuration then Asterisk changed RTP port properly.
By: David Woolley (davidw) 2011-01-19 06:23:54.000-0600
It is a perfectly legitimate thing for a UAC to do! 200 is the correct response. The problems lies with the client, if it is making changes to the SDP without updating the SDP version.