[Asterisk-Users] SIP jitter?

[Steve Kann]

To be totally honest:

I wrote the thing.

I don't think it's ready to go into HEAD, until the core people can at 
least agree on the overall structure of the implementation and integration..

There's at least one major fork that one could take with it's 
architecture (basically, whether it should be applied separately to each 
VoIP technology, or in common channel handling code), and there are pros 
and cons to be weighed there.

If we choose to continue down with this fork (presenly, it's set up to 
apply to each technology separately, with a sample integration point for 
IAX2), we could make things configurable, so you can switch between the 
old and new jitterbuffers (perhaps with some limitations, like only at 
start-time, or for new calls, etc), get it applied, and work on 
perfecting it.

It _is_ being used in iaxclient right now, so most of the up-to-date IAX 
softphones have it, BTW. But, in that case, the integration is much 
simpler, and I have more control over the project..


-SteveK