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

[Zoa]

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

Jared Smith wrote:

>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
>
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