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

[Matt]

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)
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> 
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