[asterisk-bugs] [Asterisk 0016014]: Asterisk cuts audio to the internal extension

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16014 
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Reported By:                mrmrmrmr
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16014
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           Older 1.6.0 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-10-02 15:37 CDT
Last Modified:              2009-10-14 16:52 CDT
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Summary:                    Asterisk cuts audio to the internal extension
Description: 
Hi, 

I have a problem with one call scenario on my Asterisk server. 
The call is coming from an external SIP proxy to my server. Asterisk sends
the call to one internal extension. 
Extension rings and then answers. 
The user at extension hears audio for 0.5 seconds. Then audio in that
direction is cut. (no RTP is sent) 
Audio in other direction continues. 

I got network traces for this call scenario. Everything seems normal in
SIP messaging and as RTP. 
I can also see that external audio continues even after the internal
extension's audio is cut.

This problem occurs with one type of CPE on the external side. With other
CPEs this problem is not reproduced.

I believe Asterisk cuts the internal extension's audio because it doesn't
like something in the RTP stream of external CPE.

I will add the network trace for unsuccessful call together with RTP debug
on Asterisk.
On both of these files it is clearly seen that one way audio is cut in a
very short time after call is established.


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 (0112306) mrmrmrmr (reporter) - 2009-10-14 16:52
 https://issues.asterisk.org/view.php?id=16014#c112306 
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I will try that tomorrow (as it is night time now, everybody at home is
sleeping and I don't want to disturb)

aart from that test, I wonder why it is not possible to analyze this issue
with Asterisk traces/logs ?
The RTP on one side is suddenly cut by Asterisk and I get nothing on the
debugs. isn't that weird ?

as a side note, we know that the problem also occurs when there is no
internal client and the external call is answered by echo application.
so, changing the internal client's codec won't make any difference, I'm
afraid. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-10-14 16:52 mrmrmrmr       Note Added: 0112306                          
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