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

[Jerris, Michael MI]

> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Zoa
> 
> 
> The complete jitter buffer code is ifdefd now, so could be in 
> there and not be in there at the same time. But, the 
> formatting would need to be fixed before that date, and i 
> know several people (including me) that would just put it in 
> the next version.
> 
> Patches are ok, but need to be maintained all the time..
> 

I agree about patches needing to be maintained, Kevin would have to
comment on the possibility of it going in ifdefed, and defaulting to
off.  Can somone get the code cleaned up to match asterisk coding
guidelines and formats?