[asterisk-bugs] [Asterisk 0012688]: serious problems in VAD and CNG support

[Asterisk Bug Tracker]

The following issue has been set as RELATED TO issue 0014168. 
====================================================================== 
http://bugs.digium.com/view.php?id=12688 
====================================================================== 
Reported By:                denke
Assigned To:                file
====================================================================== 
Project:                    Asterisk
Issue ID:                   12688
Category:                   General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.19 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 116466 
Request Review:              
====================================================================== 
Date Submitted:             2008-05-20 06:03 CDT
Last Modified:              2009-02-24 11:02 CST
====================================================================== 
Summary:                    serious problems in VAD and CNG support
Description: 
I know that CNG is in development state ... for a long time ago, but a bug
makes asterisk almost completly unuseable with a cisco voip router vad and
cng enabled.

1. At the beginning, sip and rtp connections are built correctly, but
somewhy the cisco router does not care about the rtp packets we send to it
(could be some protocol handleing error?) then after a while, the cisco
sends us a packet about now he stops sending rtp packets, we should
generate the confort noise ourselves. It that precise monent everything
normalises, and the other end can hear what we are telling.

2. But if we dial() an extension then, (within a few secounds) asterisk
seems to send an event to the cisco router ... something about confort
noise again, and it stops sending rtp packets. Then everthing goes wrong
again, and we can not be heared. BUT if the other side speaks (or makes
some noise), (makeing the other side send rtp packets, keeping the
asterisk's CNG off) asterisk starts sending the rtp packets again ... so
when the other side es knocking the microphone, they can hear us.
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0014168 dahdi timing doesn't work on sip channe...
====================================================================== 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-02-24 11:02 file           Relationship added       related to 0014168  
======================================================================