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

[Asterisk Bug Tracker]

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 08:17 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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 (0117014) falves11 (reporter) - 2010-01-21 08:17
 https://issues.asterisk.org/view.php?id=16663#c117014 
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I am going to try again with 1.6. Can you advise on what my sip.conf should
contain for sip-times to work the way I need it? Also,what is the most
stable 1.6 flavor? 

Issue History 
Date Modified    Username       Field                    Change               
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2010-01-21 08:17 falves11       Note Added: 0117014                          
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