[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:              2009-06-26 07:01 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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---------------------------------------------------------------------- 
 (0107023) sebycarta (reporter) - 2009-06-26 07:01
 https://issues.asterisk.org/view.php?id=14216#c107023 
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After a lot of try i report this:

I have this architecture:

[FXS]<---SIP(ulaw)-->[asterisk box 1] <---IAX2 (g729) JB----> [asterisk
box 2] <--SIP(ulaw)---> [FXS]

and the problem of the dropped calls.

I modified the architecture in this way:

[FXS]<---SIP(g729)-->[asterisk box 1] <---IAX2 (g729) JB ----> [asterisk
box 2] <--SIP(g729)---> [fxs]

I modified only the codec of the FXSs and the problem disappear!! Then i
think the problem is in the codec translation of the timestamps.

I hope this can help you. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-06-26 07:01 sebycarta      Note Added: 0107023                          
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