[asterisk-bugs] [Asterisk 0012322]: SIP reinvite record-route problem after hangup

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12322 
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Reported By:                rolek
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12322
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.18 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             03-28-2008 06:10 CDT
Last Modified:              04-17-2008 08:29 CDT
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Summary:                    SIP reinvite record-route problem after hangup
Description: 
Situation: phone1 - *a - *b - provider - phone2

When making a call from phone2 to phone1, both *b and provider use
re-invites to get out of the RTP stream. After phone1 hangs up, *b tries to
send BYE directly to the RTP server of the provider instead of its SIP
peer. The result is that phone2 does not see that the call has ended.
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Relationships       ID      Summary
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related to          0006240 [branch] Errors in support for SIP stri...
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---------------------------------------------------------------------- 
 rolek - 04-17-08 08:29  
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I'm getting somewhere. The patch of # 11545 ensures that
handle_response_invite() does not call build_route() in case of a reinvite.
However, pouring over debug traces shows that after the hangup, we send a
reinvite, we receive a 200 OK, but when we send the ACK to that sequence,
for some reason we're back to a 'standard' invite (which causes build_route
to be called, which causes the bug). 

Issue History 
Date Modified   Username       Field                    Change               
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04-17-08 08:29  rolek          Note Added: 0085619                          
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