[Asterisk-video] The new jitterbuffer and the RTCP patch

[Olle E Johansson]

Hello,

After one week of training in New Jersey, I'm back home and doing  
development again. We're finalizing the 1.4 beta
now and work hard to get ready.

Yesterday, the new Jitterbuffer for SIP/RTP was added (as well as  
many other RTP-based channels). I was wondering
whether this will affect the RTCP patch that I want to merge a.s.a.p.  
Anyone that has any ideas on that?

/O