[asterisk-dev] sip/rtp jitterbuffer in 1.4? (Chicken or the egg?)

[Patrick]

On Tue, 2006-05-30 at 18:42 +0300, Zoa wrote:
> Some clarifications,
> 
> Russel and Slav are working hard on getting it to work in trunk, the 
> complete code is ifdef's so can be enabled / disabled on compile time.
> Nobody should worry too much :)
> 
> Zoa

That is good news. Thank you all for all your hard work. I look forward
to test it when it hits trunk.

Regards,
Patrick