| Summary: | ASTERISK-11199: in certain scenarios, asterisk can send rtp in an unsupported payload type to an endpoint | ||||
| Reporter: | Torrey Searle (tsearle) | Labels: | |||
| Date Opened: | 2008-01-10 08:37:04.000-0600 | Date Closed: | 2008-01-11 13:51:02.000-0600 | ||
| Priority: | Major | Regression? | No | ||
| Status: | Closed/Complete | Components: | Channels/chan_sip/CodecHandling | ||
| Versions: | Frequency of Occurrence | ||||
| Related Issues: |
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| Environment: | Attachments: | ( 0) new_codec_patch_udiff.patch ( 1) new_codec_patch.patch | |||
| Description: | SIP caller A supports alaw/ulaw Asterisk is configured to support alaw,ulaw,g729 and places all 3 in the offer to B SIP called B supports g729,alaw,ulaw putting all 3 in the 200 OK If B sends audio in G.729 it gets forwarded to A without transcoding. | ||||
| Comments: | By: Torrey Searle (tsearle) 2008-01-10 09:35:55.000-0600 My analysis: Asterisk starts the Packet2Packet bridge so long as at there is at least 1 codec in common between the two endpoints. However, if the audio received is not in this codec, the bridge isn't shut down. Provided patch detects when this happens and shuts down the p2p bridge as needed. By: Torrey Searle (tsearle) 2008-01-11 10:18:27.000-0600 Resubmitted patch with diff -u (instead of -c) as per FAQ By: Joshua C. Colp (jcolp) 2008-01-11 10:26:07.000-0600 I'm curious, what is changing the underlying formats of the channels? By: Torrey Searle (tsearle) 2008-01-11 10:38:04.000-0600 The first section of the code passes the first incompatible rtp packet to the core. There it's detected/treated as a codec change. This changes the raw formats. Sorry, the patch was hastily forward ported from 1.4.4 to 1.4.17. The problem (as described in the ticket) is still present, however the fix isn't working the same as expected. I will look into why this patch isn't working and advise Monday. By: Digium Subversion (svnbot) 2008-01-11 13:49:09.000-0600 Repository: asterisk Revision: 98325 U branches/1.4/main/rtp.c ------------------------------------------------------------------------ r98325 | file | 2008-01-11 13:49:08 -0600 (Fri, 11 Jan 2008) | 6 lines If the incoming RTP stream changes codec force the bridge to break if the other side does not support it. (closes issue ASTERISK-11199) Reported by: tsearle Patches: new_codec_patch_udiff.patch uploaded by tsearle (license 373) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=98325 By: Digium Subversion (svnbot) 2008-01-11 13:51:02.000-0600 Repository: asterisk Revision: 98334 _U trunk/ U trunk/main/rtp.c ------------------------------------------------------------------------ r98334 | file | 2008-01-11 13:51:02 -0600 (Fri, 11 Jan 2008) | 14 lines Merged revisions 98325 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6 lines If the incoming RTP stream changes codec force the bridge to break if the other side does not support it. (closes issue ASTERISK-11199) Reported by: tsearle Patches: new_codec_patch_udiff.patch uploaded by tsearle (license 373) ........ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=98334 | ||||