[asterisk-bugs] [Asterisk 0012485]: [patch] Answer preferred codec only in SIP response
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Nov 25 02:29:43 CST 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=12485
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Reported By: bamby
Assigned To:
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Project: Asterisk
Issue ID: 12485
Category: Channels/chan_sip/CodecHandling
Reproducibility: always
Severity: feature
Priority: normal
Status: feedback
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 2008-04-21 02:58 CDT
Last Modified: 2008-11-25 02:29 CST
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Summary: [patch] Answer preferred codec only in SIP response
Description:
There is a following situation. The third party SIP gateway sends to
Asterisk INVITEs with codecs A and B in that order. Asterisk answers the
same A and B codecs. But gateway has internal policy to prefer the codec B
if possible and so it immediately sends reINVITE with codec B only. This is
undesirable behavior for the system. However Asterisk cannot be limited to
codec A only as there can be INVITEs with codec B only from the gateway and
they should be handled as well. The task is to avoid codec B when it is
possible at asterisk.
It would be nice that Asterisk is able to answer the most preferred codec
only in SIP response. This leaves no choice to the SIP gateway and thus the
reINVITEs are suppressed.
I've developed a patch that adds a global option to sip.conf that causes
the Asterisk to include the most preferred codec only in response.
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(0095471) oej (manager) - 2008-11-25 02:29
http://bugs.digium.com/view.php?id=12485#c95471
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Well, at least it has to be configurable on a device level too, not only on
the [general] level. I wonder if we could do something in the dialplan here
too, like the SIP_CODEC variable works for outbound invites.
Issue History
Date Modified Username Field Change
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2008-11-25 02:29 oej Note Added: 0095471
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