[asterisk-bugs] [Asterisk 0011823]: RTP gets passed on without early media session
noreply at bugs.digium.com
noreply at bugs.digium.com
Tue Apr 1 12:39:16 CDT 2008
The following issue has been ASSIGNED.
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http://bugs.digium.com/view.php?id=11823
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Reported By: SDamm
Assigned To: file
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Project: Asterisk
Issue ID: 11823
Category: Core/RTP
Reproducibility: always
Severity: minor
Priority: normal
Status: assigned
Asterisk Version: 1.4.13
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: 01-23-2008 09:52 CST
Last Modified: 04-01-2008 12:39 CDT
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Summary: RTP gets passed on without early media session
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.
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svnbot - 04-01-08 12:39
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Repository: asterisk
Revision: 112204
U branches/1.4/channels/chan_sip.c
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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 http://bugs.digium.com/view.php?id=11823)
Reported by: SDamm
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http://svn.digium.com/view/asterisk?view=rev&revision=112204
Issue History
Date Modified Username Field Change
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04-01-08 12:39 svnbot Note Added: 0084864
04-01-08 12:39 svnbot Status ready for testing =>
assigned
04-01-08 12:39 svnbot Assigned To => file
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