[asterisk-bugs] [Asterisk 0018601]: No RTP port update when SIP RE-INVITE is received
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Jan 19 03:53:13 CST 2011
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=18601
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Reported By: kondik
Assigned To:
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Project: Asterisk
Issue ID: 18601
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: 1.6.2.15
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2011-01-11 16:10 CST
Last Modified: 2011-01-19 03:53 CST
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Summary: No RTP port update when SIP RE-INVITE is received
Description:
Exactly the same issue as: https://issues.asterisk.org/view.php?id=15149
but those issue is closed, so I open new issue.
When RE-INVITE comes to asterisk with new media port, asterisk sends 200
OK, and send RTP packets further to the old port instead of new port which
received with RE-INVITE.
My asterisk version is: 1.6.2.15
Could you help me with this?
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Relationships ID Summary
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has duplicate 0015149 No audio on SIP RE-INVITE connecting wi...
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(0130660) kondik (reporter) - 2011-01-19 03:53
https://issues.asterisk.org/view.php?id=18601#c130660
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The problem was solved.
RE-INVITE with SDP which comes to Asterisk had unsupported version for
Asterisk, therefore asterisk ignore changes in SDP. However Asterisk send
200 OK for that RE-INVITE.
I think that in this situation asterisk should send another response than
200 OK...
When I set in configuration ignoresdpversion=on then Asterisk changed RTP
port properly.
Issue History
Date Modified Username Field Change
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2011-01-19 03:53 kondik Note Added: 0130660
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