[Asterisk-Users] Asterisk 1.00 Call quality problem

[Nick Cobley]

On Thu, 30 Sep 2004 09:24:54 +0200 (SAST), steve at daviesfam.org
<steve at daviesfam.org> wrote:
> 
> 
> On Thu, 30 Sep 2004, Nick Cobley wrote:
> 
> > 1 FXS and 1 FXO card.
> > Incoming/Outgoing calls via IAX trunking from our provider. G729
> > running between us and the VoIP provider.
> > Two handsets, one BudgetTone 102 and a Cisco 7940G running the  7.2
> > SIP firmware.
> > Both these phones are using ULAW to the server, and we have plenty of
> > G729 licenses on the server.
> >
> > Now the BudgetTone does not have any problems at all. The Cisco
> > however has severe breakup on the incoming audio from the VoIP
> > provider. The caller can hear me fine. In some cases the call is okay
> > for 10seconds or so then slowly gets worse, but most of the time I
> > just cant make out what they are saying.
> 
> 
> Make sure you aren't trying to use trunking and the iax jitter buffer
> together.  As of now they don't work properly together.  So turn off one
> or the other for the link to your VOIP provider.
> 
> As at 1.0 the jitter buffer can be enabled and disabled per peer/user.
> See iax.conf.sample.
> 
> Steve
> 
> 

Thanks Steve,

Well I tried that to no avail. jitterbuffer=no was already set in
iax.conf... but I specified it for that provider too just to make sure
but same result. I changed trunk=no also for that provider, same
again.

Still the case here is that it only occurs on the Cisco for some
reason. What I did notice during my testing was this. By doing iax2
show channels during the call, I could watch the jitter (not
jitterbuffer) value slowly go down from aroudn 30ms to 0ms. The start
of the call was absolutely fine, but once it hit 0ms it was inaudable.

I appologies in advance if I have done something stupid like not
disable the jitterbuffer correctly!

Cheers
Nick