[Asterisk-Users] Jitter Buffer on chan_sip

[TeleSIP]

Mark,

So far we have been unable to determine how to set the "jitterbuffers"
parameter in zapata.conf.  We can guess that it represents a multiple of
"bytes" of memory be we are unsure.  We have tried to set it to a value of
"1000" and it makes communication impossible on the zap side.  The zap side
(X100P) can barely hear what somebody on a SIP Phone is saying.  Sound is
all chopped up.

If we set it to "30" then it seems to work ok but we don't know if its
really doing anything.

Can you please share with us how this "jitterbuffers" parameter works and
what methodology we can use to "tune" it properly?

Thanks,
Andres.

----- Original Message ----- 
From: "Andres" <andres at telesip.net>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, November 12, 2003 10:37 AM
Subject: Re: [Asterisk-Users] Jitter Buffer on chan_sip


> 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
> > > Espia System Administrator
> > > Email : mbrancaleoni at espia.it
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