ietf
[Top] [All Lists]

RE: [Sipforum-discussion] RE: Collecting media statistics for SIP calls?

2004-11-01 13:22:12
I would like to thank every one for their responses.

Pramod Madabhushi 

--- Henry Sinnreich <Henry(_dot_)Sinnreich(_at_)mci(_dot_)com> wrote:

 Folks,

Thats what RAQMON doesin a media gnostioc fashion.

RAQMON is an excellent approach for network elements

The media statistics for SIP calls are intended at
the application level in
endpoints, such as when using an Instant Messenger
client for voice or a
soft phone, or a SIP phone, or a game with VoIP. 

Let's not be dogmatic and force techniques that have
been designed for
network elements such as routers, gateways or
servers on to the application
level in endpoints, where many good practical
reasons have shown the RTCP
extensions to be the better approach. Live and let
live!

Thanks, Henry

-----Original Message-----
From: sipforum-discussion-admin(_at_)lists(_dot_)su(_dot_)se
[mailto:sipforum-discussion-admin(_at_)lists(_dot_)su(_dot_)se] On
Behalf Of Siddiqui, Anwar
A (Anwar)
Sent: Monday, November 01, 2004 8:01 AM
To: Romascanu, Dan (Dan); ThomasGal(_at_)LumenVox(_dot_)com;
Madabhushi Pramod;
sip-implementors-requesto(_at_)cs(_dot_)columbia(_dot_)edu;
sipforum-discussion(_at_)lists(_dot_)su(_dot_)se;
ietf(_at_)ietf(_dot_)org
Subject: [Sipforum-discussion] RE: Collecting media
statistics for SIP
calls?

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: ThomasGal(_at_)LumenVox(_dot_)com; Madabhushi Pramod;
sip-implementors-requesto(_at_)cs(_dot_)columbia(_dot_)edu;
sipforum-discussion(_at_)lists(_dot_)su(_dot_)se; ietf(_at_)ietf(_dot_)org
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: ietf-bounces(_at_)ietf(_dot_)org
[mailto:ietf-bounces(_at_)ietf(_dot_)org]On 
Behalf Of Thomas Gal
Sent: 31 October, 2004 11:00 PM
To: 'Madabhushi Pramod'; 
sip-implementors-requesto(_at_)cs(_dot_)columbia(_dot_)edu; 
sipforum-discussion(_at_)lists(_dot_)su(_dot_)se; ietf(_at_)ietf(_dot_)org
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

thomasgal(_at_)lumenvox(_dot_)com  

-----Original Message-----
From: ietf-bounces(_at_)ietf(_dot_)org
[mailto:ietf-bounces(_at_)ietf(_dot_)org] On 
Behalf Of Madabhushi Pramod
Sent: Friday, October 29, 2004 4:09 PM
To: sip-implementors-requesto(_at_)cs(_dot_)columbia(_dot_)edu; 
sipforum-discussion(_at_)lists(_dot_)su(_dot_)se; ietf(_at_)ietf(_dot_)org
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: mpramod(_at_)hotmail(_dot_)com,
       madabhushi_p(_at_)yahoo(_dot_)com
Phone:001-408-204-8077


          
__________________________________
Do you Yahoo!?
Y! Messenger - Communicate in real time.
Download now. 
http://messenger.yahoo.com

_______________________________________________
Ietf mailing list
Ietf(_at_)ietf(_dot_)org
https://www1.ietf.org/mailman/listinfo/ietf



_______________________________________________
Ietf mailing list
Ietf(_at_)ietf(_dot_)org
https://www1.ietf.org/mailman/listinfo/ietf

------
To unsubscribe from the Sipforum-discussion mailing
list visit

https://mbox.su.se/mailman/listinfo/sipforum-discussion




=====
Pramod Madabhushi
email: mpramod(_at_)hotmail(_dot_)com,
       madabhushi_p(_at_)yahoo(_dot_)com
Phone:001-408-204-8077


                
__________________________________ 
Do you Yahoo!? 
Check out the new Yahoo! Front Page. 
www.yahoo.com 
 


_______________________________________________
Ietf mailing list
Ietf(_at_)ietf(_dot_)org
https://www1.ietf.org/mailman/listinfo/ietf


<Prev in Thread] Current Thread [Next in Thread>