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

[Steve Underwood]

Russell Bryant wrote:

>Roy Sigurd Karlsbakk wrote:
>  
>
>>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."
>  
>
A feature that doesn't get into the trunk is never going to get tested. 
Its as simple as that. If you won't put it in there before it is 
polished it will stay out forever. If the jitter buffer had sanely been 
stuck into the trunk, along with T.38 and other important stuff, just 
after 1.2 was released it would all have been shaken out by now. As it 
is, these important features have been sidelined into half baked alleys 
to rot.

I tried testing OEJ's test-this-branch last week, and had to get through 
about 20 obstacles to get it to run at all. Then it crashed at the least 
provocation. How many potential testers have the competance to get 
anywhere with a mess like that?

Regards,
Steve