[asterisk-bugs] [Asterisk 0011259]: Jitterbuffer does not work well with injected sound

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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http://bugs.digium.com/view.php?id=11259 
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Reported By:                plack
Assigned To:                putnopvut
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Project:                    Asterisk
Issue ID:                   11259
Category:                   Core/Jitterbuffer
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 89125 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2007-11-15 10:28 CST
Last Modified:              2008-07-25 09:25 CDT
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Summary:                    Jitterbuffer does not work well with injected sound
Description: 
Using the following in the dial plan:
exten => 208,1,Noop
exten => 208,n,Set(LIMIT_WARNING_FILE=beep)
exten => 208,n,Dial(SIP/5001||L(36000000:36000000:15000))

the injected beep file is not passing through the jitterbuffer properly. 
If you look at the additional information, the jitterbuffer is off by the
number of frames required to send the audio.

Audio quality is impacted and there is a loss of packets during the
transmission of the beep tone.
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 (0090693) putnopvut (administrator) - 2008-07-25 09:25
 http://bugs.digium.com/view.php?id=11259#c90693 
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This issue has grown quite old, and so I was assigned to try to bring a
resolution.

The problem I'm having right now is that I am unable to reproduce the
issue. I tried using the latest 1.4 as well as the revision specified in
the original bug report. Neither of them cause the warning message to be
displayed, nor did they cause issues with audio quality. Well, this isn't
the entire truth. The full truth of the matter is that I see no issue when
using a fixed jitter buffer. When attempting to use an adaptive jitter
buffer, I'm having a much worse issue, and that is that I receive no audio
at all on the channel with the adaptive jitter buffer attached. This
happens in both the latest 1.4 svn revision and the revision specified in
this report. Specifically, I'm seeing this messsage

So what I'm going to need in order to try to solve this problem is
1. The jitter buffer settings from your conf files.
2. The channel types involved in the call. I see that you are dialling a
SIP channel, but I'm not sure what type of channel you are dialling in from
(nor do I actually know if that matters).

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
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2008-07-25 09:25 putnopvut      Note Added: 0090693                          
2008-07-25 09:25 putnopvut      Status                   assigned => feedback
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