[asterisk-bugs] [Asterisk 0015149]: No audio on SIP RE-INVITE connecting with AllWorx PBX
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Jul 8 09:09:56 CDT 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=15149
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Reported By: monettes
Assigned To:
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Project: Asterisk
Issue ID: 15149
Category: Channels/chan_sip/Interoperability
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: 1.6.0.9
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-05-19 00:51 CDT
Last Modified: 2009-07-08 09:09 CDT
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Summary: No audio on SIP RE-INVITE connecting with AllWorx
PBX
Description:
We have a user with AllWorx registering a SIP DID with our Asterisk server.
When the submit 2x SIP RE-INVITEs, Asterisk doesn't use the new RTP Port of
the last SIP INVITE and creates a no-audio call.
The SIP DEBUG logs shows the right ports and report Asterisk decoding the
proper RTP port, but when you analyse the RTP packets, we see Asterisk
sending to the RTP port of the first SIP RE-INVITE, not the last one.
I attache the debug sip logs and the captured packets of a sample call.
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(0107481) adamgdt (reporter) - 2009-07-08 09:09
https://issues.asterisk.org/view.php?id=15149#c107481
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I've noticed that in your trace that the protocol version in the SDP is
never increased. Some devices use this information to select the most
recent updated SDP. In the RFC it is recommended to use a time-stamp,
however it should can be any value as long as it is increased. I've seen
some issues where devices (SIPURA sip-adapter) will ignore the SDP because
the protocol version was not incremented. I am not sure if the Asterisk
behavior is similar.
Issue History
Date Modified Username Field Change
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2009-07-08 09:09 adamgdt Note Added: 0107481
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