[asterisk-bugs] [Asterisk 0011697]: Low trunk frequency + jitter buffer = broken audio, weird netstats

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11697 
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Reported By:                nermal0
Assigned To:                
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Project:                    Asterisk
Issue ID:                   11697
Category:                   Channels/chan_iax2
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.17 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 96775 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             01-07-2008 03:15 CST
Last Modified:              01-07-2008 14:30 CST
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Summary:                    Low trunk frequency + jitter buffer = broken audio,
weird netstats
Description: 
I'm using the latest Asterisk 1.4 SVN 96775 and speex 1.2beta3. Multiple
speex calls are routed over an IAX2 link between two Asterisks. Trunking is
enabled, with a trunk frequency of 60ms (default: 20ms). (NB: trunk
"frequency" is misleading here as it really defines the time distance
between two packets and not how many packets are sent per second). In
codecs.conf, I've set:

preprocess => true
pp_vad => true

Now as soon as I enable the jitter buffer (and add forcejitterbuffer=yes
as I'm on a jitter-free LAN for testing, but will move to a high delay link
later) frames get lost and the audio is broken. "iax2 show netstat" shows:

machine1*CLI> iax2 show netstats
                                -------- LOCAL --------------------- 
-------- REMOTE --------------------
Channel                    RTT  Jit  Del  Lost   %  Drop  OOO  Kpkts  Jit 
Del  Lost   %  Drop  OOO  Kpkts
IAX2/loadtest14-2            1  121  161  -179  26     0    0      0  122 
162 16777163  18     0    0      0
1 active IAX channel

note the "-179" lost packets on the local side and the "16777163" lost
ones on the remote side (integer overflow?).

In Asterisk 1.2, the very same setup produces fine audio quality and no
dropouts. I believe there is something seriously broken in the jitter
buffer implementation of 1.4.

Thanks for any pointers on how to fix this!
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---------------------------------------------------------------------- 
 jsmith - 01-07-08 14:30  
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If I remember correctly, there are some jitterbuffer debugging tools you
can enable from the Asterisk CLI.  Have you tried turning them on, to see
if that helps debug the problem? 

Issue History 
Date Modified   Username       Field                    Change               
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01-07-08 14:30  jsmith         Note Added: 0076446                          
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