On Tue, 8 Feb 2005, Roy Sigurd Karlsbakk wrote: > >> how can I tune SIP jitter? is it possible today in asterisk? > > > > I assume you are asking for how to alleviate the effects of jitter on > > the > > RTP audio streams initated by SIP? Asterisk currently only has a jitter > > buffer for IAX, not for RTP streams. There are pland for the next > > generation jitter buffer code to hook into RTP as well. > > > > There is an entry on the bug tracker that touches on this topic. > > is this in HEAD yet?
See http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002532 There isn't even any code for SIP yet. However the iax integration works wonders for a link with just a bit of packet loss and jitter. Voice conversations are nice and crisp and without the pops associated with lost packets or growth of the jitter buffer. Peter _______________________________________________ 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
