[Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev
Andrew Kohlsmith
akohlsmith-asterisk at benshaw.com
Fri Aug 26 05:17:10 MST 2005
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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