Summary:ASTERISK-18955: Voicemail message recording from IAX source sped up and jittery
Reporter:Steven Premeau (premeau)Labels:
Date Opened:2011-12-01 15:42:06.000-0600Date Closed:2016-04-21 14:33:54
Status:Closed/CompleteComponents:Applications/app_voicemail Channels/chan_iax2
Versions: Frequency of
Environment:Asterisk i686 on CentOS 5Attachments:( 0) console.log
( 1) iax.log
( 2) iax.okay.pcap
( 3) iax.pcap
( 4) menuselect.makedeps
( 5) menuselect.makeopts
Description:When asterisk records voice mail on a call received over a IAX connection (from a third party provider) the audio is sped up and jittery.

Calls received by the trunk and forward to a Polycom SIP phone are fine.

This appears to be the same issue reported in http://forums.asterisk.org/viewtopic.php?f=1&t=74149&p=145170 and possibly similar to ASTERISK-18330

Given my connection to the net and NAT, switching to a SIP trunk is not the preferred solution.
Comments:By: Steven Premeau (premeau) 2011-12-01 15:42:58.199-0600

I can replicate this at will, so if I know what type of debugging will help, I can generate it.

By: Leif Madsen (lmadsen) 2011-12-02 13:36:23.371-0600

I suppose a console trace, configuration of the two boxes (dialplan, iax.conf, etc), which options you're using in menuselect (i.e. DONT_OPTIMIZE, DEBUG_THREADS, etc.), and the IAX debug (perhaps as a pcap trace) would all be good places to start.

By: Steven Premeau (premeau) 2011-12-04 22:36:38.775-0600

The results of my menuselect.

By: Steven Premeau (premeau) 2011-12-04 22:46:29.656-0600

Log of asterisk console with and without iax debugging enable.

core debug and core verbose were at 4.

By: Steven Premeau (premeau) 2011-12-08 07:36:25.167-0600

tcpdump capture of affected call.

By: Steven Premeau (premeau) 2011-12-08 07:40:20.554-0600

Capture of call that was answered on Polycom phone.  Audio was fine.

By: Steven Premeau (premeau) 2011-12-08 08:29:15.808-0600

In some of my testing, I was able to get different results by changing the setting for jitterbuffer on the trunk configuration.   I've got a support ticket open with my service provider to understand if they have the jitterbuffer configured for my link or not (and whether they'd be willing to turn it off for me).

However, it may indicate that the issue is how Voicemail records jitterbuffered IAX links.

By: Sean Bright (seanbright) 2012-02-20 11:56:35.711-0600

Are you using trunking with IAX2?  If so, a fix went in the other day for a timing problem.  It will be in 1.8.11 whenever that is released (it's commit [r355746|http://svnview.digium.com/svn/asterisk?view=revision&revision=355746] in the 1.8 branch - here's a [direct link|http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_iax2.c?view=patch&r1=355746&r2=355745&pathrev=355746] to the unified diff).

By: Steven Premeau (premeau) 2012-02-20 19:42:02.482-0600

I made that change in my file (understanding that it may not work because of the other changes between the versions) ... with no success.

The specific link that was having the problem was not a trunk.  (And I don't control the other end.)

I retested on a different trunk that I do control both ends, and it still exhibited the problem if jitterbuffer=yes.

Is there possible there is a similar issue in the jitterbuffer code?

By: Sean Bright (seanbright) 2012-02-21 05:23:35.767-0600

Both of the pcaps attached to this issue contain calls from IAX2 trunking (you can see this in wireshark) and exhibits the bug that r355746 fixes.  You may want to speak with your provider and point them to that particular issue.

By: seb7 (seb7) 2012-09-25 07:44:34.788-0500

I have the same problem on the lastest CentOS Asterisk 1.8 on i686 on a fully up-to-date CentOS 5:
core show version
Asterisk built by root @ 92-139-19-10.digium.internal on a i686 running Linux on 2012-08-31 18:57:50 UTC

If I leave a voicemail on the remote system over an IAX2 trunk, the message is saved with the audio sped up and jittery. If I turn off IAX2 trunking on the server that is sending the call to the remote system, the voicemail message is fine. If I turn off trunking only on the destination server it has no effect: the message is still speed up and jittery to the point of being *indecipherable*.

Why is this bug still unfixed in a stable branch after 9 months when you have all this information and traces?

By: Matt Jordan (mjordan) 2012-09-25 09:34:30.137-0500

Why is this bug still unfixed in a stable branch after 9 months when you have all this information and traces?

As of September 25th, 2012, at 9:33 CST, there are 750 open issues in the ASTERISK project.

The answer to that question should be obvious: this issue will be worked by someone as time and developer resources become available.

If that response is not sufficient, patches are always welcome.  If a patch is provided on an issue, it typically is resolved quicker than issues without patches (although that is a generalization - if a patch does not conform to the coding guidelines, is not well tested, or generally does not appear to fix the issue, then the presence of the patch doesn't really change the response time much).

If that response is not sufficient, please feel free to contact an Asterisk developer on the asterisk-biz list.  There may be a developer who is willing to resolve your issue for you.


By: Sean Bright (seanbright) 2012-11-13 18:27:35.129-0600

If I leave a voicemail on the remote system over an IAX2 trunk, the message is saved with the audio sped up and jittery.

Is the "remote system" also running

By: Leif Madsen (lmadsen) 2016-04-21 14:33:54.792-0500

Closing issue due to lack of feedback for several years :)

CC: @rnewton