[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:                jpeeler
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Project:                    Asterisk
Issue ID:                   18189
Category:                   Core/RTP
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     assigned
Target Version:             1.8.1
Asterisk Version:           1.8.0 
JIRA:                       SWP-2473 
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:              2010-12-07 14:01 CST
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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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Relationships       ID      Summary
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related to          0018352 SSRC is changing when DTMF sent
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---------------------------------------------------------------------- 
 (0129423) zktech (reporter) - 2010-12-07 14:01
 https://issues.asterisk.org/view.php?id=18189#c129423 
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For us this issue is driven by several factors. The primary is the gateway
of the backend carrier. If the call is hitting a Sonus gateway the delay
between the last rtp audio packet and the dtmf event exceeds 100ms the
Sonus throws out the DTMF event. This sucks. The G729 transcoder pushs the
delay up as well. I have found that if I disable all of the features.conf
on the systems the delay drops and Sonus is happay but this cripples some
features Also if you use the Dial options tT and you hit a Sonus switch the
issue will occure. It appears that anything that is setting in the middle
of the DTMF and incereased the delay between the last rtp audio packet and
the dtmf event will cause this. I need a good fix that will address this
with either features on or off. I am trying cmbaker82 idea to see if it
helps. This issue looks to be at the root of the overal problem
https://issues.asterisk.org/view.php?id=15642 

Issue History 
Date Modified    Username       Field                    Change               
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2010-12-07 14:01 zktech         Note Added: 0129423                          
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