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

[Greg Boehnlein]

On Fri, 26 Aug 2005, Zoa wrote:
 
> 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 think it should be included in 1.2, but marked as an experimental 
feature that is disabled by default. People then can have the option of 
enabling it to test and work with it.

I agree that a good Jitter Buffer is important for 1.2, and the only way 
that it will get better is if more people try it out and report issues 
with it. Nothing is ever perfect, and no software is ever "done". Open 
Source is a constant evolutionary process that requires Peer Review and 
Field Testing to improve.

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