1 sep 2009 kl. 08.17 skrev James Mutuku: > Hello, > > From http://www.voip-info.org/wiki/view/Asterisk+new+jitterbuffer, > it says that there For Asterisk 1.2 there was no jitterbuffer in the > RTP-based channels (i.e. chan_sip). > > I am using 1.2 and Ind there is no reason to upgrade. Are there any > developments on this?
Well, the development ended up being named Asterisk 1.4 which included jitter buffers. That's a good reason to update! /O _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
