[asterisk-bugs] [Asterisk 0006240]: [branch] Errors in support for SIP strict routing

[noreply at bugs.digium.com]

The following issue has been set as RELATED TO issue 0012322. 
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http://bugs.digium.com/view.php?id=6240 
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Reported By:                Daniel Leeds
Assigned To:                oej
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Project:                    Asterisk
Issue ID:                   6240
Category:                   Core/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases): trunk 
SVN Revision (number only!):  
Disclaimer on File?:        Yes 
Request Review:              
Resolution:                 suspended
Fixed in Version:           
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Date Submitted:             01-14-2006 16:21 CST
Last Modified:              01-31-2008 11:45 CST
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Summary:                    [branch] Errors in support for SIP strict routing
Description: 
Asterisk ignores the information in “Record-Route” headers and thus
breaks compliance with SIP RFC 3261.
This lack of compliance often results in wrong Request-URI after the BYE
and REFER methods.
Consequently the applications HangUp() and Transfer() do not work properly
in many scenarios.

Most acutely Asterisk is not in compliance with RFC-3261, Section:
12.2.1.1 and the following statement: 
“If the route set is not empty, and its first URI does not contain the
lr parameter, the UAC MUST place the first URI from the route set into the
Request-URI…”

Notice that this processing of the route set is MANDATORY in SIP
implementations, and the “route sets” are defined by the
“Record-Route” headers.

Please see quotations from RFC-3261 and Asterisk SIP Debug Output below in
Additional Information:

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Relationships       ID      Summary
----------------------------------------------------------------------
related to          0006284 SIP PROTOCOL VIOLATION:  Route informat...
related to          0007003 SIP routing gets confused
related to          0012322 SIP reinvite record-route problem after...
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Issue History 
Date Modified   Username       Field                    Change               
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04-16-08 06:50  oej            Relationship added       related to 0012322  
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