[asterisk-bugs] [Asterisk 0011729]: in certain scenarios, asterisk can send rtp in an unsupported payload type to an endpoint
noreply at bugs.digium.com
noreply at bugs.digium.com
Fri Jan 11 13:49:10 CST 2008
The following issue has been ASSIGNED.
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http://bugs.digium.com/view.php?id=11729
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Reported By: tsearle
Assigned To: file
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Project: Asterisk
Issue ID: 11729
Category: Channels/chan_sip/CodecHandling
Reproducibility: always
Severity: major
Priority: normal
Status: assigned
Asterisk Version: 1.4.17
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 01-10-2008 08:37 CST
Last Modified: 01-11-2008 13:49 CST
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Summary: in certain scenarios, asterisk can send rtp in an
unsupported payload type to an endpoint
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.
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svnbot - 01-11-08 13:49
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Repository: asterisk
Revision: 98325
U branches/1.4/main/rtp.c
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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 http://bugs.digium.com/view.php?id=11729)
Reported by: tsearle
Patches:
new_codec_patch_udiff.patch uploaded by tsearle (license 373)
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http://svn.digium.com/view/asterisk?view=rev&revision=98325
Issue History
Date Modified Username Field Change
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01-11-08 13:49 svnbot Checkin
01-11-08 13:49 svnbot Note Added: 0076734
01-11-08 13:49 svnbot Status feedback => assigned
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