[asterisk-dev] sip/rtp jitterbuffer in 1.4?

[Roy Sigurd Karlsbakk]

>> the sip/rtp jitterbuffer from http://bugs.digium.com/view.php?id=3854
>> has been in the works for more than a year, and has been in  
>> testing from
>> the last patch since 1.2.1 or so. My testing shows it makes G.711A  
>> work
>> well with crystal-clear audio even on an overloaded 704/128kbps link.
>> since asterisk is quite unusable for large-scale itsp rollouts  
>> without a
>> jitterbuffer for RTP-based protocols, SIP in particular, it would be
>> really nice to get slav's code into 1.4.
>
> Yes, it would be very nice.  But, have you even tested the code  
> that is
> the candidate for 1.4?  While testing the 1.2 branch of this code  
> may be
> useful to you, it does nothing for the project to help get it into  
> 1.4.
>
> When Joshua Colp and I started testing this last week, we ran into
> serious problems immediately.  When we tried the 1.2 version,  
> everything
> worked great.  If the trunk version gets into shape in the next few
> days, it is going in for 1.4.
>
> It's like Kevin said in another thread about this, "An unreliable,  
> buggy
> or unmaintainable jitter buffer is worse than none at all."

Can you at least please put it in there so people can start testing  
it? Trunk code usually _is_ buggy, and I somehow doubt this code is  
unmaintainable. According to zoa/slav the code is very thoroughly  
tested on 1.2, so it should be no more than porting it to 1.4. They  
may perhaps even have time to help.

I know there was much argument about allowing the jitterbuffer into  
1.2 by release or not, and that it was decided against it. PLEASE do  
not let this happen with 1.4. The PLC and jitterbuffer is is a  
feature much missed in asterisk, and they're both in these works.

best regards

roy

--
Roy Sigurd Karlsbakk
roy at karlsbakk.net
(+47) 98013356
---
"Ubuntu: Ancient African word for ''I'm sick of compiling Gentoo''"  
-- Jeff Waugh