Summary:ASTERISK-13563: jitterbuffer delays after a meetme finished
Reporter:Christopher Faust (cfaust)Labels:
Date Opened:2009-02-11 10:47:22.000-0600Date Closed:2009-02-12 12:21:13.000-0600
Versions:Frequency of
Description:The version of asterisk is 1.4.  

We need to explain why the jitter buffer is so high after being into a meetme. We don’t have this problem when we are not in a meetme

IAX2/iax_internal_fi   iax_intern  00022/00120  00016/00020  00040ms  0133ms  0183ms  gsm
IAX2/iax_internal_fi   iax_intern  00028/00132  00032/00028  00040ms  0160ms  0200ms  gsm
IAX2/iax_internal_fi   iax_intern  00045/00133  00050/00051  00040ms  0207ms  0247ms  gsm
IAX2/iax_internal_fi   iax_intern  00055/00114  00008/00006  00000ms  0133ms  0185ms  gsm
IAX2/iax_internal_fi   iax_intern  00104/00097  00202/00205  00040ms  0207ms  0250ms  gsm

Voici l’extension qui dial ce meetme remote
exten => _MEETME.,2,Dial(Local/j1${EXTEN}@${CONTEXT}/nj)
exten => _j1MEETME.,1,Meetme(${EXTEN:8},Adqx)
exten => _j1MEETME.,2,Hangup

We need to explain why there is such a huge delay in the jitter buffer when in a meetme room


version 1.4-r11047M   on 64 bit machine
Comments:By: Leif Madsen (lmadsen) 2009-02-11 11:50:15.000-0600

You mention SVN revision 11047 -- which is so old that 1.4 wasn't even branched at that point.

You also mention versions 1.4.17, and 1.4.23 in this bug report.

Could you please clarify which version of Asterisk specifically you are using?

In addition, a few changes to the jitterbuffer code have been put in recently, so I would encourage you to test this with the latest version from the 1.4 branch.


By: Christopher Faust (cfaust) 2009-02-11 14:23:58.000-0600

the version which has this issue is 1.4-r11047M

By: Leif Madsen (lmadsen) 2009-02-11 15:32:56.000-0600

That revision certainly doesn't exist in the 1.4 branch.

What does your 'show version' output?

I suggest you upgrade to the latest from the 1.4 branch.

By: Christopher Faust (cfaust) 2009-02-11 17:37:52.000-0600

the above is from the    show version output    I did not add   built by root @ Lepton on a x86_64 running Linux on 2009-01-06  18:50:57 UTC

By: Christopher Faust (cfaust) 2009-02-11 17:42:36.000-0600

basically i am going through 3 asterisk servers
#1 in montreal
#2 in fedricton
#3 in another location

#1 connects to #2   voice quality is fine
then we do a meetme which brings in #3

when the meetme is finished and #1 is still connected to #3  that is where I see the delay causing extremely poor sound quality

By: Leif Madsen (lmadsen) 2009-02-12 09:18:04.000-0600

We still have no idea what version you are actually running, because the version you supplied does not exist. Revision 11047 is too old to actually even be a 1.4 release.

So either you're not running 1.4, or you're running something quite old, or some other combination of old or corrupt.

I would encourage you to reproduce the issue on a more recent version of Asterisk.

By: Clod Patry (junky) 2009-02-12 12:21:13.000-0600

blitzrage: forget about it.
I know cfaust in real.
I will take care of that.