asterisk users - Mar 2006 - status on jitter buffer for SIP/RTP? (OT?)

This might be a better question for the dev list, but does anyone know 
the status of a jitter buffer for SIP channels?

I know they created a generic jitter buffer and implemented it for IAX 
channels.  Does it work yet for SIP?  Like is it there and disabled or 
not there at all?