[asterisk-bugs] [Asterisk 0010481]: SIP with canreinvite=yes through multiple Asterisk instances fails

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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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:              11-19-2007 13:18 CST
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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...
related to          0009431 Modify connection: Response 491 not han...
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 oej - 11-19-07 13:18  
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The issue here is really that we're sending a 491 without supporting it on
the receiving end. I might have to disable the 491 totally and just refuse
the offer with another code to disable it until we have proper 491 support,
which isn't easy to do in Asterisk. The re-invite is forced from deep
inside the RTP subsystem. If we stall it in one server - we will stall it
in a range of servers and that will propably cause more havoc.

Food for thought. Thanks for a good a detailed bug report. 

Issue History 
Date Modified   Username       Field                    Change               
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11-19-07 13:18  oej            Note Added: 0073993                          
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