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

[Russell Bryant]

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