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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
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:                     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:              
====================================================================== 
Date Submitted:             2010-01-20 17:50 CST
Last Modified:              2010-01-21 01:05 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.


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

---------------------------------------------------------------------- 
 (0117003) dzajro (reporter) - 2010-01-21 01:05
 https://issues.asterisk.org/view.php?id=16663#c117003 
---------------------------------------------------------------------- 
Sorry for maybe stupid question, but how will you detect the opposite side
is dead and call should be disconnected?

If there is now RTP coming from opposite side, we can assume some
signalling packets (BYE) has been lost and call should be terminated. With
Your update all functionality became meaningless. 

For your scenario, let's consider to use:

Asterisk sip rtpkeepalive = Number : Number of seconds, when a RTP
Keepalive packet will be sent if no other RTP traffic on that connection.
Default 0 (no RTP Keepalive). (New in v1.2.x).

and/or bigger rtptimeout. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-01-21 01:05 dzajro         Note Added: 0117003                          
======================================================================