Subject: RE: [Sipping] RE: draft-stein-great-00.txt

From: "Yaakov Stein" <[email-masked]>
Date: Thu, 6 Oct 2005 20:52:32 +0200

> I find this statement hard to reconcile with my daily 
> experience using SIP phones, and other VoIP such as Skype, 
> Google Talk, Damaka, etc. between Dallas, Boston, Washington, 
> Berlin, Amsterdam and Tokyo over the public Internet 'as is'. 

Actually, at least two of these services use several advanced mechanisms
of the type I am talking about - but completely nonstandard.
What I want is more such mechanisms, 
and for them to be open and standardized.

BTW, I was just on the phone with someone who works in Cisco,
ie. calling PSTN to VoIP. There were two DTMF talk-offs in a ten minute
conversation. Even were the voice a solid 4, that would
take points off.

> In all such calls, using the GIPS broadband codec (40
> kb/s) the quality is clearly superior to the PSTN and the MOS 
> would be higher than 5.0 if telecom minds could grasp such a 
> wild notion. 

Yes, broadband codecs could conceivably be rated
at MOS 5, and indeed are quite good when the Internet holds up.
But the peak MOS is not a very good indicator of call quality.

I personally use Skype all the time - always with a broadband connection
-
but would not think to call it "better than PSTN".
In fact, I frequently have to fall back to the IM when the going
gets rough.

The typical delay is MUCH too long for comfortable
conversation, and the drop outs too frequent.
And when using SkypeIn or SkypeOut you drop down
to G.711, there no special mechanisms,
and congestion is frequent. So the ONLY
reason to use it is that it is CHEAP.

> A good reminder is also the Internet did not 
> fail, nor any of these services (including Vonage) during 
> recent diasters - in stark contrast to the "five nine" 
> legendary telecom networks that all failed for days and weeks.

In PSTN terms, the public Internet "services" are in
a constant state of failure - check any standard measure
of availability, reliability, error-rate, etc.

And of course you can't be five-nines realibility
when you rely on your local (three nines) electric
company to power your PC. I am wondering how
your claim of Internet availability really held up
when the lights were out.

> Maybe your Internet access link is the problem?

I use Skype from work (VERY broadband connection)
from home (1.5M), at IETF meetings (reasonable speed),
at the ITU (>1M), etc. The results are quite consistent. 

> 
> As for Dean's comments that Internet communications are 
> different, I would only add the observation that confirms it: 
> Most decision makers I happen to know would rather give up 
> the $40k/year office receptionist who uses the various call 
> transfers mentioned by Dean than give up their $400 (plus usage
> fee) Blackberry, even for 15 minutes. Blackberry has the 
> clear win over call transfer for the decision makers in most 
> well and alive industries. I did not mean to include legacy 
> telecom though...

Precisely. 

> 
> > IETF is NOT the proper forum to do this work.
> 
> We agree on this.

Well, I certainly agree that SIP/SIPPING are not the WGs
to do this work.

Y(J)S

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---

From: Jonathan Rosenberg <[email-masked]>
Date: Thu, 06 Oct 2005 16:54:00 -0400

Yaakov,

I for one thank you for this draft. I, like others, believe that VoIP 
can ultimately produce a better user experience than PSTN, and its good 
to see folks write down some of the problems that need to be solved to 
get there.

What I am unsure about, is what specific work you are proposing for 
ietf. Much of what is discussed are implementation tricks and 
algorithms; i.e., jitter buffer adaptation appraoches. Are you proposing 
that IETF should somewhere standardize those?

Some of the other problems you describe - high packet loss and delay - 
are resolved either through overprovisioning or through the various QoS 
mechanisms IETF has proposed over the years. None of the QoS mechanisms 
are present on the public Internet, but the barriers there are business 
related, not technical. So, I am not sure whether you think there is 
IETF work to be done there or not.

Another item you mention in your draft and in the thread is a 
standardized wideband codec. Are you proposing that IETF standardize one?

Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
[email-masked]                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

---

From: "Dale R. Worley" <[email-masked]>
Date: Fri, 07 Oct 2005 15:46:40 -0400

> IETF work to be done there or not.

The longer this discussion progresses, the more I think that it should
be in another arena, such as the SIP Forum.  The problems that Yakov
mentions are quite real, at least in substantial areas of use.  But I
think that they need to be analyzed first at a whole-system level -- not
only codecs, protocols, routing, QoS, but also the human context, both
in regard to user psychology but also the business realities of
networking.  Only after looking at the whole-system situation can we
discern that "this would be much improved if we exploited the
possibilities of variable bit-rate codecs", or "if we tweaked RSVP in
certain manners, network operators would embrace it (because of xyz)",
or some other technical direction as being an effective solution to the
problem as a whole.  Once we identified technical requirements to a
piece of the overall solution, then we could make an IETF work item out
of it, especially in regard to ensuring that there is a standardized
solution to the problem, rather than a host of competing proprietary
ones.

Dale