> I am using CVS latest > > Is it correct there is no jitter buffer for SIP (RTP) > > Are there any plans for this? > > prob a stupid question: > Is it required / do the endpoints handle this - if the > src and destination are both SIP and there is no > transcoding but asterisk is still in the media path?
My understanding is the new jitterbuffer code (in cvs-head) has been applied to iax connections, and the objective is to make it available for sip/rtp (and possibly other channel types) after things are cool in iax. A jitterbuffer is only required when the delivery of rtp packets is inconsistent (eg, jerky). Its my understanding that sip phones have at least some sort of jitterbuffer built into firmware. Don't know how effective they are for large variations in packet delivery though. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
