[asterisk-bugs] [Asterisk 0018189]: RFC2833 DTMF generation broken due to SSRC change on bridges channels

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18189 
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Reported By:                marcbou
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18189
Category:                   Core/RTP
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.0 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-10-22 12:13 CDT
Last Modified:              2010-10-28 02:36 CDT
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Summary:                    RFC2833 DTMF generation broken due to SSRC change on
bridges channels
Description: 

Since upgrading to the latest 1.8.0-rc, DTMF digits sent over SIP/RTP to
our service provider were no longer being detected.

We analyzed packet dumps, comparing old and new RTP packets being
generated by asterisk.

The difference was tracked down to asterisk 1.8.0-rc now changing the SSRC
value for RFC2833 DTMF digit packets.

If in main/channel.c:ast_channel_bridge() I comment out 

    ast_indicate(c0, AST_CONTROL_SRCCHANGE);
    ast_indicate(c1, AST_CONTROL_SRCCHANGE);

the SSRC no longer changes for DTMF digits and the provider can detect
them again.

However I am not sure if the change doesn't adversely affect other things.
Please advise.

Kind regards,

Marc Boucher



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 (0128434) langen5 (reporter) - 2010-10-28 02:36
 https://issues.asterisk.org/view.php?id=18189#c128434 
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Yes - this explains, why "constantssrc" has been removed, but I still
cannot figure out, why asterisk receives rtpevents with the same
ssrc-identifier as the rest of the rtp media-stream from the sip-client and
sends out rtpevents and rtp media-stream with different ssrc-identifiers to
the provider. Asterisk stayes in the media-stream and there are no
re-invites send out. 
According to the second link, the ssrc should not change, but it does... 

Issue History 
Date Modified    Username       Field                    Change               
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2010-10-28 02:36 langen5        Note Added: 0128434                          
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