[asterisk-dev] Jitter Buffer

[John Lange]

As of Revision 45678 there is a small patch that makes it work with
jbforce=yes.

However, as far as I can tell it never works unless you force it.

So for example:

PSTN (T1 via Zap) <-> Asterisk 1.4b3 <-> Voip Device

The Asterisk <-> Voip Device leg never gets jitter buffer unless
forced. 

And yes I'm talking about the received audio at the Asterisk side. It is
not dejittered.

John

On Fri, 2006-10-20 at 16:11 +0200, Pavel Jezek wrote:
> I have exactly same problem, no jb log in tmp, no message on console (I 
> have verbose set to 3),
> and because sound is very choppy using SIP (much worse than with iax), I 
> have suspicion, that jitterbuffer is not working/not activated at all.
> can somebody look to this?
> PJ
> 
> 
> jbenable=yes
> jbforce=yes
> jbmaxsize=2000
> jbimpl=fixed (adaptive tried too)
> jblog=yes
> 
> Asterisk SVN-branch-1.4-r45678
> 
> 
> 
> >  I'm trying to test the new jitter buffer in 1.4beta2 but have not been
> able tell if its working.
> 
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