[asterisk-bugs] [Asterisk 0015337]: Drawn out and static audio on inbound iax2 calls

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Jun 29 09:27:08 CDT 2009


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15337 
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Reported By:                deitch
Assigned To:                dvossel
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Project:                    Asterisk
Issue ID:                   15337
Category:                   Channels/chan_iax2
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.1.0 
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:             2009-06-16 17:49 CDT
Last Modified:              2009-06-29 09:27 CDT
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Summary:                    Drawn out and static audio on inbound iax2 calls
Description: 
Calls that are inbound on a DID, to which Asterisk has registered using
iax2, have terribly drawn out audio and a lot of static for prerecorded
files (e.g. IVR and Voicemail files in /var/lib/asterisk/sounds). They
sound like:
".... hhhheeeellllllloooooo..... Yyyyyyoooouuuu hhhhaaaavvvveeee " etc.

Specific details:
1) This occurs only on IAX2, not on SIP. I have tried with the exact same
providers, on the same hosts
2) jitterbuffer is off. If I turn jitterbuffer on, no audio at all is
audible.
3) This occurs across two completely separate providers, one in Israel,
one in the US. However, inbound from the ITSPs directly to an IAX2
softphone (e.g. JackenIAX or Zoiper) work perfectly fine.
4) This occurs only when communicating to Asterisk. If the dialer enters
an extension, and is thus connected over Asterisk to a SIP softphone, there
is no problem.
5) This occurs independent of codec. I have the original outbound files
converted to both ulaw and gsm, and it occurs in both cases. The provider
prefers ulaw.
6) CPU usage by Asterisk remains extremely low during the bad output.
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---------------------------------------------------------------------- 
 (0107117) deitch (reporter) - 2009-06-29 09:27
 https://issues.asterisk.org/view.php?id=15337#c107117 
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Oh, sorry, missed the other question. 

I hate to say, I am not sure. I have been having enough problems getting
audio to work, even within the internal network (standard X-Lite phone as a
SIP extension, no NAT), that I cannot confirm it either way. I am trying to
isolate the issues one at a time. The list was a mile long, fortunately, it
is down to "only" the pre-recorded voice on IAX2, and the voice on internal
SIP extensions.... 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-06-29 09:27 deitch         Note Added: 0107117                          
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