[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:              08-17-2007 10:57 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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---------------------------------------------------------------------- 
 alexcos - 08-17-07 10:57  
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I have the same problem (i think)



Huawei  IpPhone  ->  asterisk1    ->  asterisk2      ->     patton

10.36.2.85      ->    10.36.2.1    -> 10.105.177.21  ->  10.105.177.14



The RTP setup is a mess , the huawei phone is trying to send the rtp to
10.105.177.14 , and the patton expects it from 10.36.2.1
Asterisk2  seems to be the problem , but not using an reinvite..

Asterisk1 and Asterisk2 are the same , version: Asterisk 1.4.8


If you need more info , pls contact me.

Regards 

Issue History 
Date Modified   Username       Field                    Change               
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08-17-07 10:57  alexcos        Note Added: 0068993                          
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