[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:                dvossel
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Project:                    Asterisk
Issue ID:                   14703
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
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:              2009-04-02 07:33 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:


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 on bug.


regards,
genie



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---------------------------------------------------------------------- 
 (0102606) snuffy (manager) - 2009-04-02 07:33
 http://bugs.digium.com/view.php?id=14703#c102606 
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I've assigned to you dvossel, as you last worked on 491 recently..
Is it likely that you may have fixed this in your latest patch? 

Issue History 
Date Modified    Username       Field                    Change               
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2009-04-02 07:33 snuffy         Note Added: 0102606                          
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