[asterisk-bugs] [Asterisk 0018900]: Chopping audio into ConfBridge from IAX call when jitterbuffer=yes

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18900 
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Reported By:                gilester
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18900
Category:                   Applications/app_confbridge
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.8.3 
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:             2011-02-28 12:56 CST
Last Modified:              2011-02-28 15:48 CST
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Summary:                    Chopping audio into ConfBridge from IAX call when
jitterbuffer=yes
Description: 
I'm using 2 Asterisk servers (one production machine at 1.6.2, one
development machine at 1.8.3) connected with an IAX trunk across a LAN. One
SIP phone on each server. Each phone calls into a ConfBridge conference on
the 1.8.3 Asterisk.

When the IAX jitterbuffer is enabled (using the default settings in the
iax.conf file), audio into the ConfBridge conference FROM the IAX call is
very choppy and entirely unusable.

Audio in the other direction is fine. Audio between ConfBridge members on
the same Asterisk is fine. Audio from an IAX call directly to the SIP
handset is fine - so it seems to be an interaction between the IAX
jitterbuffer and ConfBridge.

I have no DAHDI modules in this server and "timing test" reveals:

>Attempting to test a timer with 50 ticks per second.
>Using the 'timerfd' timing module for this test.
>It has been 1003 milliseconds, and we got 50 timer ticks

Also, the LAN shouldn't have any jitter on it. During the ConfBridge call,
iax2 show channels alternates between showing:

>Channel               Peer             Username    ID (Lo/Rem)  Seq
(Tx/Rx)  Lag      Jitter  JitBuf  Format  FirstMsg    LastMsg
>(None)                192.168.10.2     (None)      00121/00000 
00001/00000  00000ms  0000ms  0040ms  unknow  Tx:POKE     Tx:POKE
>IAX2/tred51.net-9616  192.168.10.2     tred51.net  09616/18419 
00025/00026  00040ms  0151ms  0200ms  ulaw    Rx:NEW      Tx:ACK
2 active IAX channels

and showing:

>Channel               Peer             Username    ID (Lo/Rem)  Seq
(Tx/Rx)  Lag      Jitter  JitBuf  Format  FirstMsg    LastMsg
>IAX2/tred51.net-9616  192.168.10.2     tred51.net  09616/18419 
00025/00026  00040ms  0151ms  0200ms  ulaw    Rx:NEW      Tx:ACK
1 active IAX channel

The channel (None) in the first example comes and goes. I have no idea
what it is doing, perhaps it's a red herring?
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 (0132455) gilester (reporter) - 2011-02-28 15:48
 https://issues.asterisk.org/view.php?id=18900#c132455 
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Thanks,

Tried that, makes no difference to the choppy audio. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-02-28 15:48 gilester       Note Added: 0132455                          
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