Like you point out its not SIP specific but there are certain aspects of SIP that we need to take care of and RAQMON does that. let me clarify; Since it is SIP the session monitoring is media agnostic and need to accomodate that. Thats what RAQMON doesin a media gnostioc fashion. Voice Over IP, Video over IP or Fax over IP and many other apps fit into the Framework. See it helps you.
Since it is in WG LAst call, would be very happy to see what kind of needs you have and ensure that it serves the purpose. Anwar -----Original Message----- From: Romascanu, Dan (Dan) Sent: Monday, November 01, 2004 2:36 AM To: [EMAIL PROTECTED]; Madabhushi Pramod; [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED] Cc: Siddiqui, Anwar A (Anwar) Subject: RE: Collecting media statistics for SIP calls? I agree. This is not a SIP domain specific issue. See my previous answer pointing to the real-time application QoS monitoring (RAQMON) work in the RMON MIB WG. Regards, Dan > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > Behalf Of Thomas Gal > Sent: 31 October, 2004 11:00 PM > To: 'Madabhushi Pramod'; > [EMAIL PROTECTED]; > [EMAIL PROTECTED]; [EMAIL PROTECTED] > Subject: RE: Collecting media statistics for SIP calls? > > > I don't think that's really a SIP domain issue, though > this may have > been adressed somewhere that I'm not aware of. > If you're using RTP to carry the audio than these statistics are > derived from the RTP stream in the context of the sender > (with you being the > receiver) and the sender sends SR (Sender Report) RTCP packets with > interarrival jitter included. Packet count, fraction lost, > and cumulative > number of packets lost are also transmitted in these packets. > Latency would > probably be derived from the NTP timestamp if anything and is > not directly > addressed in RTP to my knowledge. So if you wanted this data > after the fact > the server would have to maintain that information, and it > would probably be > a query at the application level, not really anything to do with SIP. > > -Tom > > [EMAIL PROTECTED] > > > -----Original Message----- > > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On > > Behalf Of Madabhushi Pramod > > Sent: Friday, October 29, 2004 4:09 PM > > To: [EMAIL PROTECTED]; > > [EMAIL PROTECTED]; [EMAIL PROTECTED] > > Subject: Collecting media statistics for SIP calls? > > > > Is there any way by which I call query a SIP endpoint for > > media statictics after call termination. I would like to know > > details like Jitter, latency, packet loss, packets received, > > packets sent etc. > > > > Thanks in advance. > > > > Pramod Madabhushi > > ShoreTel communications. > > > > ===== > > Pramod Madabhushi > > email: [EMAIL PROTECTED], > > [EMAIL PROTECTED] > > Phone:001-408-204-8077 > > > > > > > > __________________________________ > > Do you Yahoo!? > > Y! Messenger - Communicate in real time. Download now. > > http://messenger.yahoo.com > > > > _______________________________________________ > > Ietf mailing list > > [EMAIL PROTECTED] > > https://www1.ietf.org/mailman/listinfo/ietf > > > > > _______________________________________________ > Ietf mailing list > [EMAIL PROTECTED] > https://www1.ietf.org/mailman/listinfo/ietf > _______________________________________________ Ietf mailing list [EMAIL PROTECTED] https://www1.ietf.org/mailman/listinfo/ietf
