[asterisk-bugs] [Asterisk 0016663]: RTP Timeout is flawed

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Jan 21 07:54:53 CST 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16663 
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Reported By:                falves11
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16663
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            sometimes
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.29 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 238629 
Request Review:              
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Date Submitted:             2010-01-20 17:50 CST
Last Modified:              2010-01-21 07:54 CST
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Summary:                    RTP Timeout is flawed
Description: 
The sip.conf configuration rtptimeout=XX only considers one way, not both.
Suppose the caller is silent for more than XX seconds, on mute, while his
mother issues a long speech. The call drops. This function must consider
audio both ways, and only if there is silence in both channels, drop the
call.

In my particular case, if I call to a voicemail where the greeting is
longer than XX seconds, and I simply keep my microphone silent, the call
will drop exactly at XX seconds, invalidating the whole thing.


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---------------------------------------------------------------------- 
 (0117011) file (administrator) - 2010-01-21 07:54
 https://issues.asterisk.org/view.php?id=16663#c117011 
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Asterisk 1.4 does not support session timers. As for your comment about
rtptimeout not working under pretty normal conditions - these aren't pretty
normal conditions. There's a reason you are the first person to bring this
up and that is because most setups do not use VAD or silence suppression
and continue to send packets even while muted. In those more normal
conditions rtptimeout would operate perfectly fine as intended. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-01-21 07:54 file           Note Added: 0117011                          
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