|Summary:||ASTERISK-16876: Garbled Audio from IAX trunk|
|Reporter:||Frank Barney (swampfox0866)||Labels:|
|Date Opened:||2010-10-26 14:45:25||Date Closed:|
|Environment:||Attachments:||( 0) config01.zip|
( 1) config02.zip
( 2) iax2_debug.txt
|Description:||When making a call through an IAX trunk I get garbled audio. This happens when only one call is traversing the trunk.|
****** ADDITIONAL INFORMATION ******
I upgraded from version 1.4.36 to 22.214.171.124 so I can take advantage of the trunkmtu configuration option because I need to run 90+ calls through this IAX trunk. I have rolled back to version 1.4.36 and am still able to run up to 80 calls through the trunk using the g729 codec with very few problems. Upwards of 80 calls and I start getting some garbled audio.
The source call originates on a dahdi channel. The call then gets sent over an iax trunk to another asterisk box which then terminates the call to another dahdi channel. As mentioned, the call is compressed using the g729 codec. Compression is done using the TC400B transcoder card.
I have captured some packets while making a call. In the iax2 meta trunk packets I am seeing multiple media frames per packet when I expect to only see one media frame (because only one call is traversing the trunk). Upon closer inspection, each media frame has the same source call number.
|Comments:||By: Frank Barney (swampfox0866) 2010-10-26 15:00:21|
I've uploaded the configuration files from the asterisk machines on either side of the IAX Trunk.
By: Leif Madsen (lmadsen) 2010-11-02 08:41:40
Please upload configuration and debugging information as plain-text and not as a compressed files. It makes it difficult to work with.
By: Adrian Hensler (adrianhensler) 2010-11-03 18:03:14
In case this info helps, I noticed similar problem going from 126.96.36.199 to 1.8.0.
In house asterisk version 188.8.131.52 with an IAX trunk to external 184.108.40.206. Minimal use trunk with rarely more than one call. Source calls are either DAHDI or SIP. Everything works fine.
Changed the external to asterisk 1.8.0 (underlying OS changed to Ubuntu 10.10 from 10.4) and now DAHDI call over iax trunk is garbled and then eventually (60 seconds give or take, sometimes less) gets dropped. I saw these errors "chan_iax2.c: Max retries exceeded to host"
Edit: A single SIP call over the same trunk is perfect. Edit #2: Calls in my case were ulaw.
I thought it must be a timing issue, I was very confused that the SIP call was fine but the DAHDI call was garbled.
I did not troubleshoot much, I just rolled back.
Very simple setup, I can possibly recreate or test or provide more info if required.
By: Adrian Hensler (adrianhensler) 2010-11-04 20:55:10
I just uploaded a debug, again not sure if this is the same issue. First call is sip (home asterisk) ->iax trunk->(remote asterisk)sip provider.
Followed by a (home asterisk) dahdi->iax trunk->(remote asterisk)sip provider.
Sip call may not have been perfect but was much better.
Debug captured on (remote asterisk).
There are a couple IAX tickets that look similar to me, but I don't want to mislead.
Edit: I sanitized phone numbers, IAX Challenge, but that is about it.
By: Sean Bright (seanbright) 2012-02-18 09:20:36.237-0600
I'm 99% certain this is related to a timing error that was recently discovered in the IAX2 channel driver. Take a look at the [last comment|https://issues.asterisk.org/jira/browse/ASTERISK-16258?focusedCommentId=189279&page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-189279] of ASTERISK-16258 and let us know if that resolves the problems you are having.