[Asterisk-Users] SIP jitter?

[Roy Sigurd Karlsbakk]

>> how can I tune SIP jitter? is it possible today in asterisk?
>
> I assume you are asking for how to alleviate the effects of jitter on 
> the
> RTP audio streams initated by SIP? Asterisk currently only has a jitter
> buffer for IAX, not for RTP streams. There are pland for the next
> generation jitter buffer code to hook into RTP as well.
>
> There is an entry on the bug tracker that touches on this topic.

thanks

is this in HEAD yet?

roy