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

[Arnaud]

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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