Hi 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? Thanks Jack __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ 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
