Summary:	ASTERISK-11284: RTP gets passed on without early media session
Reporter:	Sebastian Damm (sdamm)	Labels:
Date Opened:	2008-01-23 09:52:10.000-0600	Date Closed:	2008-04-01 12:48:02
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Core/RTP
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) 11823.diff
Description:	When Asterisk sends out an INVITE and receives a provisional response without SDP, it still passes on RTP packets arriving on this leg to the other leg of the call getting established. As a consequence, Asterisk does not generate ringing to the Zap side on the other leg, or it sends out a 183 response to the other leg. Discussion about this problem on the list can be found here: http://lists.digium.com/pipermail/asterisk-dev/2008-January/031660.html A SIP trace is not needed as it does not show anything unusual. Expected behavior is, that Asterisk should drop those RTP packets arriving without an early media session established.
Comments:	By: Olle Johansson (oej) 2008-01-27 05:04:02.000-0600 This bug is in the wrong category - it's not a SIP issue, more of RTP in combination with the core pbx. By: Joshua C. Colp (jcolp) 2008-03-26 13:26:33 Please try the attached patch. By: Sebastian Damm (sdamm) 2008-03-28 04:36:32 Just tried it with a release 1.4.18.1 and heard a ringing even though the other end sent RTP packets after the 180. So the patch does work. Will the patch make it into 1.4 releases? Or only into 1.6? By: Digium Subversion (svnbot) 2008-04-01 12:39:16 Repository: asterisk Revision: 112204 U   branches/1.4/channels/chan_sip.c ------------------------------------------------------------------------ r112204 | file | 2008-04-01 12:39:14 -0500 (Tue, 01 Apr 2008) | 4 lines Do not pass audio until the remote side has indicated they are providing early media, or if the channel has been answered. (closes issue ASTERISK-11284) Reported by: SDamm ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=112204 By: Digium Subversion (svnbot) 2008-04-01 12:44:20 Repository: asterisk Revision: 112205 _U  trunk/ U   trunk/channels/chan_sip.c ------------------------------------------------------------------------ r112205 | file | 2008-04-01 12:44:17 -0500 (Tue, 01 Apr 2008) | 12 lines Merged revisions 112204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112204 | file | 2008-04-01 14:43:46 -0300 (Tue, 01 Apr 2008) | 4 lines Do not pass audio until the remote side has indicated they are providing early media, or if the channel has been answered. (closes issue ASTERISK-11284) Reported by: SDamm ........ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=112205 By: Digium Subversion (svnbot) 2008-04-01 12:48:02 Repository: asterisk Revision: 112206 _U  branches/1.6.0/ U   branches/1.6.0/channels/chan_sip.c ------------------------------------------------------------------------ r112206 | file | 2008-04-01 12:48:01 -0500 (Tue, 01 Apr 2008) | 20 lines Merged revisions 112205 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112205 | file | 2008-04-01 14:48:52 -0300 (Tue, 01 Apr 2008) | 12 lines Merged revisions 112204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112204 | file | 2008-04-01 14:43:46 -0300 (Tue, 01 Apr 2008) | 4 lines Do not pass audio until the remote side has indicated they are providing early media, or if the channel has been answered. (closes issue ASTERISK-11284) Reported by: SDamm ........ ................ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=112206