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

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Jan 21 08:03:10 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 08:03 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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 (0117012) falves11 (reporter) - 2010-01-21 08:03
 https://issues.asterisk.org/view.php?id=16663#c117012 
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I do not use VAD or silence suppression. But I confirmed last night that at
the beginning of call, if I set the rtptimeout to 30 seconds, and the
answering machine salutation is longer than 30 seconds, while the
microphone is mute, the calls drops exactly at 31 seconds. If I extend the
rtptimeout o 60 seconds, it drops at 61 seconds and so forth. This does not
happen if the microphone is open and there is some background noise.
So what is going on here? the RTP should be flowing even with the
microphone muted. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-01-21 08:03 falves11       Note Added: 0117012                          
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