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

[Matt]

On 8/26/05, Andrew Kohlsmith <akohlsmith-asterisk at benshaw.com> wrote:
> On Friday 26 August 2005 06:22, Arnaud wrote:
> > I agree, and also it will be better to be able to enable SIP jitter per
> > user
> 
> Uh, why?
> 
> -A.
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