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

[Matt]

Could we possibly offer a patch to the 1.2 system?  Then brave souls
could patch it and use asterisk-1.2-stable while still using the
jitter buffer.

On 8/26/05, Jerris, Michael MI <mjerris at ofllc.com> wrote:
> > Eric Wieling aka ManxPower
> >
> > 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.
> 
> The sip jitterbuffer was just been reworked, and still has a known
> memory leak at high loads.  I think that this, as much as I would like
> to see it in, can't be considered for 1.2.
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