[Asterisk-Users] Jitter Buffer on chan_sip

[Andres]

On Wednesday 12 November 2003 09:47, Mark Spencer wrote:
> it's implemented on the zap side (which is now configurable with
> "jitterbuffers=foo" in zapata.conf.
Will this work on a SIP to SIP call?  

What does the parameter jitterbuffers=XXX represent?  Is it memory allocation 
or milliseconds of voice?

Thanks,
Andres

>
> Mark
>
> On Wed, 12 Nov 2003, Matteo Brancaleoni wrote:
> > mmmh... I'm not sure ig chan_sip has jitter buffer.
> > I think that there isn't a jb in sip,
> > but correct me if I'm wrong.
> >
> > Matteo.
> >
> > Il lun, 2003-11-10 alle 16:14, Andres ha scritto:
> > > Hi,
> > >
> > > I would like to test chan_sip with a bigger jitter buffer.  Does
> > > anybody know where in the code this is defined?  I looked through it
> > > but could not find where.
> > >
> > > If anybody else can find it please let me know.
> > >
> > > Regards,
> > > Andres
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> > Matteo Brancaleoni
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