[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:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
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:              01-27-2008 05:04 CST
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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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---------------------------------------------------------------------- 
 oej - 01-27-08 05:04  
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This bug is in the wrong category - it's not a SIP issue, more of RTP in
combination with the core pbx. 

Issue History 
Date Modified   Username       Field                    Change               
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01-27-08 05:04  oej            Note Added: 0081231                          
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