After some very dry reading (RFC3261 - SIP, RFC1889 - RTP), I found some interesting (ha!) information in section 6.3 of RFC1889 on RTCP Sender/Receiver Reports -- with some good keywords in hand, I googled through the asterisk mailing lists to discover that this feature is coming Real Soon Now(TM), since 2004:
http://bugs.digium.com/view.php?id=2863 So, now I'm reading through the rather lengthily feature report, and seeing how far along things are. There's a fairly substantial patch attached to the bug report for rtp.c that looks like it will provide debug output for RTCP. With luck, I can shoe-horn that into Asterisk 1.0.9, without moving to -HEAD. Cheers, Simon P. Ditner On 11/2/05, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > On Wednesday 02 November 2005 16:56, Simon P. Ditner wrote: > > Elvis brings up a good point about QoS which should help me with > > scoring calls -- are per-call values available for jitter, packet > > loss, and delay available from within asterisk? > > IAX2, yes. SIP, no. I was going to actually meter this out and spit that > information into some kind of event, whether it be manager or not I'm not > sure. Basically the end of each IAX2 call would have an event telling what > the stats were (local and remote). > > -A. > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [EMAIL PROTECTED] > For additional commands, e-mail: [EMAIL PROTECTED] > >
