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

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I would like to thank every one for their responses.

Pramod Madabhushi 

--- Henry Sinnreich <Henry.Sinnreich@xxxxxxx> 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@xxxxxxxxxxx
> [mailto:sipforum-discussion-admin@xxxxxxxxxxx] On
> Behalf Of Siddiqui, Anwar
> A (Anwar)
> Sent: Monday, November 01, 2004 8:01 AM
> To: Romascanu, Dan (Dan); ThomasGal@xxxxxxxxxxxx;
> Madabhushi Pramod;
> sip-implementors-requesto@xxxxxxxxxxxxxxx;
> sipforum-discussion@xxxxxxxxxxx;
> ietf@xxxxxxxx
> 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@xxxxxxxxxxxx; Madabhushi Pramod;
> sip-implementors-requesto@xxxxxxxxxxxxxxx;
> sipforum-discussion@xxxxxxxxxxx; ietf@xxxxxxxx
> 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@xxxxxxxx
> [mailto:ietf-bounces@xxxxxxxx]On 
> > Behalf Of Thomas Gal
> > Sent: 31 October, 2004 11:00 PM
> > To: 'Madabhushi Pramod'; 
> > sip-implementors-requesto@xxxxxxxxxxxxxxx; 
> > sipforum-discussion@xxxxxxxxxxx; ietf@xxxxxxxx
> > 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@xxxxxxxxxxxx  
> > 
> > > -----Original Message-----
> > > From: ietf-bounces@xxxxxxxx
> [mailto:ietf-bounces@xxxxxxxx] On 
> > > Behalf Of Madabhushi Pramod
> > > Sent: Friday, October 29, 2004 4:09 PM
> > > To: sip-implementors-requesto@xxxxxxxxxxxxxxx; 
> > > sipforum-discussion@xxxxxxxxxxx; ietf@xxxxxxxx
> > > 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@xxxxxxxxxxx,
> > >        madabhushi_p@xxxxxxxxx
> > > Phone:001-408-204-8077
> > > 
> > > 
> > > 		
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> > 
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> 


=====
Pramod Madabhushi
email: mpramod@xxxxxxxxxxx,
       madabhushi_p@xxxxxxxxx
Phone:001-408-204-8077


		
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