[asterisk-bugs] [Asterisk 0010481]: SIP with canreinvite=yes through multiple Asterisk instances fails
noreply at bugs.digium.com
noreply at bugs.digium.com
Sun Aug 19 19:50:54 CDT 2007
The following issue has been set as DUPLICATE OF issue 0010449.
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http://bugs.digium.com/view.php?id=10481
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Reported By: mavetju
Assigned To:
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Project: Asterisk
Issue ID: 10481
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.4.10.1
SVN Branch (only for SVN checkouts, not tarball releases): 1.4
SVN Revision (number only!): 79553
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 08-17-2007 08:50 CDT
Last Modified: 08-17-2007 12:19 CDT
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Summary: SIP with canreinvite=yes through multiple Asterisk
instances fails
Description:
The story at http://www.mavetju.org/~edwin/asterisk-sip-reinvite.html
describes a problem I experienced with calls coming from one of our
providers where during the SIP handshake our equipment was reinviting
the SIP session: The RTP stream was never setup. We experienced
this after the upgrade from 1.2 to 1.4 (the latest SVN version),
before that it always has worked.
To simulate this problem, I have setup one SIP phone, three identical
Asterisk instances and a connection towards the end-point: A Cisco
Call Manager. The only varying factor in the experiments was the
option "canreinvite": When using "canreinvite=no", it always worked
fine, but when using "canreinvite=yes", it broke down after two
hops.
I have written down the whole setup, the configurations, the scenarios
and the results at http://www.mavetju.org/~edwin/c2-flow.txt.
Attached to each scenario are the SIP packets (captured with ngrep
and processed into a flow visualiser).
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Relationships ID Summary
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duplicate of 0010449 Maximum retries for seqno 102 when re-i...
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Issue History
Date Modified Username Field Change
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08-19-07 19:50 file Relationship added duplicate of 0010449
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