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

[Asterisk Bug Tracker]

The following issue has been CLOSED 
====================================================================== 
https://issues.asterisk.org/view.php?id=16663 
====================================================================== 
Reported By:                falves11
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16663
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            sometimes
Severity:                   major
Priority:                   normal
Status:                     closed
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:              
Resolution:                 open
Fixed in Version:           
====================================================================== 
Date Submitted:             2010-01-20 17:50 CST
Last Modified:              2010-01-21 10:51 CST
====================================================================== 
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.


====================================================================== 

---------------------------------------------------------------------- 
 (0117030) lmadsen (administrator) - 2010-01-21 10:51
 https://issues.asterisk.org/view.php?id=16663#c117030 
---------------------------------------------------------------------- 
At this point we're just dealing with a support issue as there is no bug
here. Please take this to the asterisk-users mailing list, or some other
appropriate forum. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-01-21 10:51 lmadsen        Note Added: 0117030                          
2010-01-21 10:51 lmadsen        Status                   new => closed       
======================================================================