Subject: RE: [Sip] RE: [rohc] Reduced processing overhead for SIP compression

From: [email-masked]
Date: Fri, 17 Aug 2001 18:19:38 -0500

Hi Hans,

Since the SIP Call set up latency is one of the main (if not the main)
problem we want to solve with signaling compression, I propose we add a
requirement to that effect:

The compression scheme MUST be such that when applied to SIP, the call
set up latency is less than X seconds, for a representative set of
scenarios to be defined. Scenarios specify the messages exchanged, their
contents, as well as the channel bit rates. 

We need to agree on X and the scenarios, but I believe this requirement
is useful, because compression ratio alone does not adequately capture
the problem of call set up latency. Scenarios can be based on those
typically envisioned in cellular cases like 3GPP.

Khiem

-----Original Message-----
From: ext Hans Hannu [mailto:[email-masked]]
Sent: 15 August, 2001 1:49 AM
To: Liu Zhigang.C (NRC/Dallas); [email-masked]
Cc: [email-masked]; [email-masked]
Subject: RE: [Sip] RE: [rohc] Reduced processing overhead for SIP
compression

Hi,

I'm currently doing some work on the requirements document to fit the
comments that I received from the list before the London meeting. But
I'll
wait for the meeting minutes to see if there is a need for additional
changes.

/Hans H

> Hi,
>
> I wasn't present during the ROHC WG meeting, so I don't
> know if this is a moot question already. But what happened
> to the requirements? Specifically, is this going to be a
> generic protocol or SIP-only?
>
> (I guess this will be clarified in the new WG charter
> as well as in the meeting minute.)
>
> BR,
> Zhigang
>
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---
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Archive: http://www.cdt.luth.se/rohc/

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use [email-masked] for questions on current sip
Use [email-masked] for new developments on the application of sip

---

From: Lawrence Conroy <[email-masked]>
Date: Sat, 18 Aug 2001 13:12:52 +0100

>typically envisioned in cellular cases like 3GPP.
To which I reply:
Hi Folks,
   There is a GERAN meeting at the end of the month, and I will canvas
to see if there is a HARD limit such as that proposed as a requirement.

However...
I think that the assumption that the main goal of SIP Comp is to reduce
(or, as described above, =>LIMIT<= ) the call set up time is not obvious
(to me, at least) and I'm not yet happy to have it included in the
requirements document as proposed.

For GERAN at least, the HARD requirement is on the mid-call Re-Invite
sequence, NOT on the call set up. That's probably true for UTRAN as well,
but that's not my area of interest.

I have already sent a posting on my understanding of the GERAN mid-call
problem and the figures that seem to apply there (and the reasons).

Of course one benefit of SIP Comp is that we reduce the transmission
time for messages sent during call set up for latency reasons, and so
we may avoid too many re-transmissions of SIP messages. However, I don't
see how this is a HARD requirement and would be interested in understanding
the logic here.

Do we have a specific limit (like the DCS one that is, I believe, regulated
by the FCC)?

BR,
  L.
-- 
[email-masked]: +44 1794 833666::<my opinions>:

---

From: "Hans Hannu" <[email-masked]>
Date: Mon, 20 Aug 2001 18:45:49 +0200

Hi,

I like the idea of having some defined scenarios to be able to compare the
scheme against and those scenarios could be included in the requirements
document, but perhaps not under the requirements section, but under the
"assumptions" section.

We have already some message sequences (thanks to Richard and Ishita) that
we could use. What we need to do now is just find out some representative
channel bitrates to run these against. Maybe someone could provide some
channel bitrates?

Are there other "message exchange sequences" that someone thinks are more
representative than those already used?

BR
/Hans H

---

From: "Dean Willis" <[email-masked]>
Date: Mon, 20 Aug 2001 23:01:00 -0500

I would argue that there are two goals of SIP or UDP compression.

1) To reduce the bandwidth used by SIP messaging, especially non-session
messaging such as REGISTERS and the like.
2) To reduce the serialization delay through very slow connections such as
wireless. This impacts call setup time, mid call event processing, and call
teardown. It further impacts the jitter induced by mixed signaling and media
on a non-interleaving connection such as traditional PPP.

--
Dean

----- Original Message -----
From: "Lawrence Conroy" <[email-masked]>
To: <[email-masked]>; <[email-masked]>
Cc: <[email-masked]>; <[email-masked]>;
<[email-masked]>; <[email-masked]>
Sent: Saturday, August 18, 2001 7:12 AM
Subject: RE: [Sip] RE: [rohc] Reduced processing overhead for SIP
compression