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

[Anton]

IMHO - RTP jitterbuffer and PLC should be the first prio 
over many new features of the 1.4 and no exiting features 
for instance as expectable as JB in asterisk among service 
providers, since we just excessively need that

On 29 May 2006 20:10, 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."
>
> Russell
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