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

[Steve Underwood]

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 :)

So, the next one is T.38 passthrough. That's a patch that could have 
gone in safely the day it was submitted. Does it stand any chance of 
getting in a mere 9 months later?

Steve