[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:              2009-07-16 07:39 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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---------------------------------------------------------------------- 
 (0107840) farisraouf (reporter) - 2009-07-16 07:39
 https://issues.asterisk.org/view.php?id=14216#c107840 
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Unfortunately not only can I not reproduce it on demand, I've found that
jitterbuffer=no didn't seem to resolve things 100% reliably but does seem
to have made things a lot better.

The real problem we have is that it happens far to infrequently, and the
level of our calls is very low, so it is hard to tell what's happening. 


One tiny tiny snippet of info that I can add is this (when we experience
the problem):

When we make outbound calls via IAX, if the problem happens we stop being
able to hear the person we have called after a few mins.

In contrast, when we get incoming calls, the caller can't hear us after a
few mins. 

(And just as the original poster said, if you wait a little while without
hanging up, the audio returns to normal).


In all cases the connection has a latency of 10ms or less. It is from our
own co-lo Asterisk box to a highly respected PSTN gateway provider - not
cheap and cheerful. I've managed to ping the gateway provider when the
problem was in progress, and there was nothing obvious going on. Still 10ms
or less.

I do, in theory, have a TCPdump that covers a call when the problem was
happening, but I don't want to post it anywhere public. I also need to
double-check I still have it. It is quite big -- could be as much as 10Mb
as I had it running constantly. I would not be able to point to the exact
place in the dump that relates to when the problem happened either. So not
all that useful I suppose :-( 

Issue History 
Date Modified    Username       Field                    Change               
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2009-07-16 07:39 farisraouf     Note Added: 0107840                          
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