[asterisk-bugs] [Asterisk 0011823]: RTP gets passed on without early media session

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11823 
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Reported By:                SDamm
Assigned To:                
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Project:                    Asterisk
Issue ID:                   11823
Category:                   Core/RTP
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.4.13 
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:             01-23-2008 09:52 CST
Last Modified:              03-28-2008 04:36 CDT
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Summary:                    RTP gets passed on without early media session
Description: 
When Asterisk sends out an INVITE and receives a provisional response
without SDP, it still passes on RTP packets arriving on this leg to the
other leg of the call getting established. As a consequence, Asterisk does
not generate ringing to the Zap side on the other leg, or it sends out a
183 response to the other leg. 

Discussion about this problem on the list can be found here:
http://lists.digium.com/pipermail/asterisk-dev/2008-January/031660.html

A SIP trace is not needed as it does not show anything unusual. 

Expected behavior is, that Asterisk should drop those RTP packets arriving
without an early media session established.
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---------------------------------------------------------------------- 
 SDamm - 03-28-08 04:36  
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Just tried it with a release 1.4.18.1 and heard a ringing even though the
other end sent RTP packets after the 180. So the patch does work. 

Will the patch make it into 1.4 releases? Or only into 1.6? 

Issue History 
Date Modified   Username       Field                    Change               
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03-28-08 04:36  SDamm          Note Added: 0084681                          
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