[asterisk-bugs] [Asterisk 0012570]: Changing of RTP SSRC between early-session and actual call session

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The following issue has been SUBMITTED. 
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http://bugs.digium.com/view.php?id=12570 
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Reported By:                fcastellano
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12570
Category:                   Core/RTP
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.19 
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:             05-02-2008 05:52 CDT
Last Modified:              05-02-2008 05:52 CDT
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Summary:                    Changing of RTP SSRC between early-session and
actual call session
Description: 
Upgrading Asterisk from 1.4.17 to 1.4.19 I noticed some SIP clients (for
example Siemens Gigaset C450IP), experienced a silence period (4/5 secs)
after the remote party (on PSTN) answered. Looking at the RTP packets, I
concluded this is relate to the change of SSRC between the early session
established with 183 Session Progress, and the actual call session
following the 200 Ok message.
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Issue History 
Date Modified   Username       Field                    Change               
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05-02-08 05:52  fcastellano    Asterisk Version          => 1.4.19          
05-02-08 05:52  fcastellano    SVN Branch (only for SVN checkou => N/A          
  
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