Hello all! I dont know how affect this issue (jitter buffer) on a SIP implementation with a VOIP trunk and I want to know how to setup this item to get a good IP quality calls without voice delay. thanks for any help.
_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
