On Tue, 13 Jul 2004 [EMAIL PROTECTED] wrote:
> Qualify will only stop the call going through if for example the ping > is above 200ms. I find most of my problems come from fluctuating > ping times (~100ms) than from a stable high ping. I agree that the overall delay isn't really the problem - jitter and packet loss are what causes the trouble. There really isn't currently anything in Asterisk which measures this - especially not when there is no active call using the path. The IAX2 jitter buffer code does know the amount of jitter - and could probably make this measurement available in a variable or something. And I propose to add similar jitter buffer code for SIP and other RTP-using protocols too. But I'm not really sure how the measurement can then be used effectively for call routing. I'd be interested in your ideas. Note that I observe that in my environment jitter and packet loss come and go over a timescale of seconds - this a result of sharing a narrowish pipe with a bunch of other traffic without any shaping to help the VOIP traffic. For this environment the real fix is to improve the network rather than do anything too complicated with *. (Not to say that *s jitter handling and packet-loss-concealment can't be improved - I've been working on that and I'm still busy). I'm about to ask for some help in gathering jitter stats from a bunch of users - perhaps you'd like to help with that. Steve _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users