[Asterisk-Dev] SIP RTP JitterBuffer in Asterisk

[Olle E. Johansson]

Matt wrote:
> Hi,
> I heard talk that there was a SIP RTP JitterBuffer which was either in
> asterisk CVS, or was being made as a patch here on the dev list.   Can
> anyone confirm this or deny it?  And if it exists, what is the current
> status of it?
At this point, a lot of work is going on to fix this. We do not know if
we can fix it in due time for 1.2. There are non-working patches in the
bug tracker, but rumour (a less than one hour old rumour) tells me that
something is working out there. Nothing is included in CVS at this point.

A bit longer answer than "no" this second time :-)

/O