Subject: RE: [Sipping] New internet draft on media loopback

From: "Hedayat, Kaynam" <[email-masked]>
Date: Fri, 8 Oct 2004 09:54:41 -0400

Jonathan,

Thank you for your comments. My reply is in-line below.

Regards,
Kaynam

-----Original Message-----
From: Jonathan Rosenberg [mailto:[email-masked]]
Sent: Thursday, October 07, 2004 7:03 PM
To: Hedayat, Kaynam
Cc: [email-masked]'
Subject: Re: [Sipping] New internet draft on media loopback

Thanks for writing this up.

I like the general idea of adding mechanisms in sip to facilitate 
testing and monitoring of an active network, but there are a lot of 
higher level questions about this specific proposal.

Firstly, I think you need to step back and really understand the problem 
that is being solved. Is the problem that you want a way to have network 
operator that deploys sip figure out the quality of calls made in the 
network? Is the problem that someone in the field wants to test if the 
system is "working"? That is, if a call can be made?

[KH] The problem is pro-active monitoring of a VoIP service to make sure
that it is working per quality standards set by the provider. The
proposed solution is making "test" calls to endpoints and collecting
performance statistics for such calls.

I'm assuming it's probably the latter. Do you need the phone that is 
acting as the mirror to also automatically answer the sip invite, or is 
another human going to do that? If its the former, then you need more 
things than just SDP mirroring. You would need a way for an automated 
answering of the call. That has all kinds of interesting security 
issues; being able to send someone a message that causes them to 
automatically answer calls can be troubling and used as a source of dos 
attacks (consider a phone with a single line).

[KH] The intervention of a human being is not going to be scalable and
manageable solution. Therefore the calls should be answered
automatically. Furthermore, some of the Voice Quality measurements
will not be possible without the loopback of the original media. However,
our goal is not to mandate the loopback of media and/or automatic
answer for "test" calls. Our goal is to introduce a loopback media 
attribute and the notion of the "test" call. The caller decides
how media should be handled (loopback, someone talking, etc). The callee
decides how to handle a "test" call. As you suggested there is more than 
SDP mirroring that needs to be done. We are currently working on changing 
this draft to address SDP issues only (e.g. new attributes) and a new draft 
for establishing "test" calls that will address your concerns.

If you assume that there is a human that will answer the phone acting as 
the mirror, what is the need for actually mirroring the media? Isn't it 
enough to merely receive some kind of media, and you can collect 
statistics from the RTCP? Why isn't someone saying 
"testing-1-2-3-testing" sufficient?

[KH] The need for mirroring the media is the reception of the sent media
for voice quality measurement and voice recording (for offline quality
assessment). That said in some situations someone saying "testing-1-2-3"
is sufficient. Our goal is to leave this as an option.

So, at a high level, I still don't understand the specific diagnostic 
problem which this is attempting to solve. I think you need to 
articulate that as a problem statement and requirements in order for the 
group to evaluate it better.

[KH] We will do what you have suggested. As far as the IETF process, should
we post a new draft based on comments on the list prior to the IETF meeting?

Thanks,
Jonathan R.

Hedayat, Kaynam wrote:
> I would like to bring the attention of the group to a new internet draft
on
> media loopback capability in SIP sessions. The draft can be found at
> http://www.ietf.org/internet-drafts/draft-hedayat-media-loopback-00.txt
> 
> Regards,
> Kaynam
> 
> -----------------------------------------
> Kaynam Hedayat
> Chief Technology Officer
> Brix Networks
> 285 Mill Road
> Chelmsford, MA 01824
> 
> tel.   978.367.5611
> fax.   978.367.5700
> [email-masked]
> www.brixnet.com
> 
> See it before they hear it: Test the performance and quality of your VoIP
at
> www.TestYourVoip.com
> 
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use [email-masked] for questions on current sip
> Use [email-masked] for new developments of core SIP
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
[email-masked]                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use [email-masked] for questions on current sip
Use [email-masked] for new developments of core SIP

---

From: "Hedayat, Kaynam" <[email-masked]>
Date: Fri, 8 Oct 2004 10:20:30 -0400

The service will provide loopback points in specific locations of the
network. The goal is to provide loopback points in all locations of the
network. As stated in the draft what you suggested has been implemented in
some networks and has proven to be not scalable and unmanageable.

Regards,
Kaynam

-----Original Message-----
From: Christian Jansson [mailto:[email-masked]]
Sent: Friday, October 08, 2004 4:03 AM
To: Jonathan Rosenberg; Hedayat, Kaynam
Cc: [email-masked]
Subject: RE: [Sipping] New internet draft on media loopback

Why not build a service and give it a SIP-URI like
sip:[email-masked] that answers all calls and loops back the
audio? There is no need for any new SDP attributes to be able to build
such a service. 

/ Christian Jansson, Hotsip

---

From: "Hedayat, Kaynam" <[email-masked]>
Date: Fri, 8 Oct 2004 10:24:32 -0400

Rajnish,

This is a very good point. What are the requirements for the responder of
the 102, 105, 108 tests? Is looping back the media sufficient?

Regards,
Kaynam

-----Original Message-----
From: Jain, Rajnish [mailto:[email-masked]]
Sent: Friday, October 08, 2004 10:18 AM
To: Hedayat, Kaynam; [email-masked]
Subject: RE: [Sipping] New internet draft on media loopback

Kaynam,

In many ways, this draft is analogous to continuity test (COT) and test
calls 102, 105, 108 of traditional telephony. While there are some
fundamental differences between a TDM continuity test and an RTP media
loopback test, there are many similar goals. 

While the draft at this time seems to emphasize user end-points, the
idea can also be extended to SIP-enabled media gateways. Currently, in a
SIP bridging scenario (RFC 3372, RFC 3398), a continuity test between an
SSP and a media gateway tests only the local TDM path (not the
end-to-end TDM-RTP-TDM path). If SIP bridging were to be truly
transparent to the two SSPs with respect to the continuity test, the
media gateways could possibly receive a continuity test request on the
TDM side and propagate to the SIP side (and vice-versa). 
 
Rajnish

-----Original Message-----
From: [email-masked] [mailto:[email-masked]] On
Behalf Of Hedayat, Kaynam
Sent: Monday, October 04, 2004 3:22 AM
To: [email-masked]'
Subject: [Sipping] New internet draft on media loopback

I would like to bring the attention of the group to a new internet draft
on media loopback capability in SIP sessions. The draft can be found at
http://www.ietf.org/internet-drafts/draft-hedayat-media-loopback-00.txt

Regards,
Kaynam

tel.   978.367.5611
fax.   978.367.5700
[email-masked]
www.brixnet.com

See it before they hear it: Test the performance and quality of your
VoIP at www.TestYourVoip.com

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP Use
[email-masked] for questions on current sip Use
[email-masked] for new developments of core SIP

---

From: "Hedayat, Kaynam" <[email-masked]>
Date: Fri, 8 Oct 2004 10:43:18 -0400

The goal is to:

- Measure availability of an endpoint (at the SIP level)
- Measure Call Signaling performance (PDD, etc.)
- Measure Media performance (voice quality)
- Identify causes of poor Call Signaling & Media performance (packet loss,
jitter, etc.)

Regards,
Kaynam

-----Original Message-----
From: Jonathan Rosenberg [mailto:[email-masked]]
Sent: Friday, October 08, 2004 8:31 AM
To: Christian Jansson
Cc: Hedayat, Kaynam; [email-masked]
Subject: Re: [Sipping] New internet draft on media loopback

Before jumping into solutions, I think it is important to understand the 
specific diagnostic and testing requirements. What is it that you wish 
to discover with the test?

-Jonathan R.

---

From: "Hedayat, Kaynam" <[email-masked]>
Date: Fri, 8 Oct 2004 11:02:04 -0400

RTP loopback will apply to 108 test. 102 and 105 tests would require
additional signaling at the signaling layer.

Regards,
Kaynam

-----Original Message-----
From: Jain, Rajnish [mailto:[email-masked]]
Sent: Friday, October 08, 2004 10:59 AM
To: Hedayat, Kaynam; [email-masked]
Subject: RE: [Sipping] New internet draft on media loopback

108 is a simple loopback test. However, 102 and 105 require responder to
generate special tones.

-----Original Message-----
From: Hedayat, Kaynam [mailto:[email-masked]] 
Sent: Friday, October 08, 2004 10:25 AM
To: Jain, Rajnish; Hedayat, Kaynam; [email-masked]
Subject: RE: [Sipping] New internet draft on media loopback

Rajnish,

This is a very good point. What are the requirements for the responder
of the 102, 105, 108 tests? Is looping back the media sufficient?

Regards,
Kaynam

-----Original Message-----
From: Jain, Rajnish [mailto:[email-masked]]
Sent: Friday, October 08, 2004 10:18 AM
To: Hedayat, Kaynam; [email-masked]
Subject: RE: [Sipping] New internet draft on media loopback

Kaynam,

 
Rajnish

-----Original Message-----

Regards,
Kaynam

tel.   978.367.5611
fax.   978.367.5700
[email-masked]
www.brixnet.com

---

From: "Jain, Rajnish" <[email-masked]>
Date: Fri, 8 Oct 2004 10:17:36 -0400

Kaynam,

 
Rajnish

-----Original Message-----

Regards,
Kaynam

tel.   978.367.5611
fax.   978.367.5700
[email-masked]
www.brixnet.com

---

From: "Jain, Rajnish" <[email-masked]>
Date: Fri, 8 Oct 2004 10:58:49 -0400

-----Original Message-----
From: Hedayat, Kaynam [mailto:[email-masked]] 
Sent: Friday, October 08, 2004 10:25 AM
To: Jain, Rajnish; Hedayat, Kaynam; [email-masked]
Subject: RE: [Sipping] New internet draft on media loopback

Rajnish,

Regards,
Kaynam

-----Original Message-----
From: Jain, Rajnish [mailto:[email-masked]]
Sent: Friday, October 08, 2004 10:18 AM
To: Hedayat, Kaynam; [email-masked]
Subject: RE: [Sipping] New internet draft on media loopback

Kaynam,

 
Rajnish

-----Original Message-----

Regards,
Kaynam

tel.   978.367.5611
fax.   978.367.5700
[email-masked]
www.brixnet.com