[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:                     feedback
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:              2009-06-17 18:04 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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 (0106585) guillecabeza (reporter) - 2009-06-17 18:04
 https://issues.asterisk.org/view.php?id=14216#c106585 
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dvossel, 19450601ms are 324 minutes, it doesn't look like a sign problem.
just by looking at it seems more like a bug on the timestamp calculation.

the problem may be on the originating peer (maybe a sip carrier?) and not
on the asterisk side, because the audio frames timestamp is forwarded
as-it-comes.

andrey, can you please specify what technology are you using on the
originating peer? are incoming or outgoing calls? 

Issue History 
Date Modified    Username       Field                    Change               
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2009-06-17 18:04 guillecabeza   Note Added: 0106585                          
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