[Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev

[Matt]

I'm going to assume it's still a patch based on what I've seen elsewhere.
As far as responding to arnaud I have to say "uhh why" as well.  Why
would you not want the jitter buffer on all the time for all users? 
It really can only help (if it works correctly) which so far it does
indeed seem to.

On 8/26/05, Matt <mhoppes at gmail.com> wrote:
> I'm confused now.. is the sip jitter buffer currently available in the
> CVS-HEAD or does it still need to be patched in?
> 
> On 8/26/05, Arnaud <arno at directcentrex.com> wrote:
> > Eric Wieling aka ManxPower wrote:
> >
> > > Kevin P. Fleming wrote:
> > >
> > >> AEL (pbx_ael.c) will be included in the 1.2 release, but will be
> > >> clearly marked in the UPGRADE.txt file as experimental. If we don't
> > >> include it in 1.2, we won't get very many testers other than the
> > >> brave souls who will continue to run the development branch :-)
> > >
> > >
> > > The same could be said about the SIP jitter buffer.
> > >
> > I agree, and also it will be better to be able to enable SIP jitter per user
> > --
> >
> > Arnaud Pignard (apignard at frontier.fr)
> > Standard : + 33 1 70 71 50 00 - Fax : +33 1 70 71 50 60
> > MSN : stormfr at hotrmail.com - ICQ : 20946060
> >
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> >
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