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

[Jared Smith]

On Tue, 2006-05-30 at 19:57 +0800, Steve Underwood wrote:
> I think anyone who would be happy to ship 1.4 without a solid reliable 
> jitterbuffer would be happy to ship a car with a wheel missing.

I've gotta add my two cents... (and I'm not picking on Steve here, I'm
just replying to his message.)

If you want a solid reliable RTP jitterbuffer in 1.4, then help out!
Jitterbuffers don't invent themselves, and they don't stress-test
themselves, and they don't debug themselves.  In short, it makes me sick
to hear some developers blame other developers for not having done the
work necessary for a solid reliable jitterbuffer.  If *we* really want a
good jitterbuffer, then *we* (myself included) better stop sitting
around wishing it would happen and actually *do the work* to get
something in shape before 1.4 is upon us.  As far as I can tell, very
few people have even tried the existing proposed jitterbuffer for 1.4.

I should also respond to the cry for putting the proposed jitterbuffer
in trunk.  While I'm in no position to speak for anyone but myself, I'm
quite sure that were the proposed jitterbuffer to go into trunk, we'd
get a huge backlash from people running trunk complaining that suddenly
their RTP audio is all messed up!  Seems to me like we can't have it
both ways...

-Jared