[asterisk-bugs] [Asterisk 0014703]: sip call setup bug

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14703 
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Reported By:                genie
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14703
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.0.6 
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:             2009-03-19 08:04 CDT
Last Modified:              2009-03-19 08:18 CDT
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Summary:                    sip call setup bug
Description: 
I'm testing SIP protocol on Asterisk. While performing tests on performance
with packet loss I came across a bug of Asterisk while call initiation. In
my Asterisk SIP configuration I have 'canreinvite=no' set. That is why
normal call setup looks like that:
student.agh.edu.pl/~genie/Visio-call_setup_no_reinvite.pdf

However, if the first ACK packet of the flow is lost and the ClientA who
does not know it sends Invite_with_authorisation packet the call will never
be set up, ending up with a infinte loop where client sends ACK messages
and asterisk '491 request pending' message. The wireshark files captured on
both ClientA and Asterisk can be found here: 
http://student.agh.edu.pl/~genie/asterisk/

regards,
genie



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 (0101968) genie (reporter) - 2009-03-19 08:18
 http://bugs.digium.com/view.php?id=14703#c101968 
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i wish i could remove links to the files mentioned in the bug report as i
have attached them to the report, but i dont know how to edit the post. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-03-19 08:18 genie          Note Added: 0101968                          
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