[asterisk-bugs] [Asterisk 0014216]: Random audio dropouts 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=14216 
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Reported By:                Andrey Sofronov
Assigned To:                dvossel
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Project:                    Asterisk
Issue ID:                   14216
Category:                   Channels/chan_iax2
Reproducibility:            random
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.22 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-01-12 10:02 CST
Last Modified:              2009-06-22 01:45 CDT
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Summary:                    Random audio dropouts when jitterbuffer = yes
Description: 
Sometimes (couple times per month) I get one-way audio issue on IAX2
trunks.
iax.conf looks like:
[general]
autokill = yes

bindaddr = xx.xx.xx.xx
disallow = all

jitterbuffer = yes
maxjitterbuffer = 1000
trunktimestamps = yes
transfer = yes

[guest]
type = user
context = guest

[peer1]
type = user
allow = speex
auth = rsa
inkeys = ....
context = peer1_incoming

[peer2]
type = peer
username = tminsk_speex
host = xx.xx.xx.xx
allow = speex
trunk = yes
qualify = yes
auth = rsa
outkey = ....

When the issue occurs, the calling party can hear the remote party, but
the remote party hears silence. The only way that helps is "unload module
chan_iax2.so && load module chan_iax2.so". Also disabling jitterbuffer and
"iax2 reload" helps. 

http://bugs.digium.com/view.php?id=14044 - that patch doesn't help me
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 (0106766) sebycarta (reporter) - 2009-06-22 01:45
 https://issues.asterisk.org/view.php?id=14216#c106766 
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I have the same problem, also on single core boxes. I tried asterisk
1.4.25.1 and 1.6.0.10, the problem is the same, randomly the call is
dropped out, but i noticed that waiting for 1 or 2 minutes the call return
and i can continue to speak. 

Please resolve this issue because jitter buffer is very important on slow
link like adsl connection.

Thank you for your work. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-06-22 01:45 sebycarta      Note Added: 0106766                          
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