Subject: Re: [Ieprep] draft-ietf-ieprep-framework-02.txt

From: Ken Carlberg <[email-masked]>
Date: Fri, 01 Nov 2002 11:44:05 +0000

> No disagreement with any of that.  Have you, however, been able to find
> any IP telephony (by your definition, SIP or equivalent) using site
> pairs which are experiencing interconnection performance worse than
> that?  The only connection we've been able to find which failed to
> exceed PSTN quality was actually a satellite modem connection, so it
> was, in effect, tunnelled through the PSTN.

yes, through parts of the transatlantic link.  some connections between
the US and Chile (locations other than Santiago), and between the US and
New Zealand.  the caveat is that this loss of PSTN quality is over the 
busy period of the day (as opposed to 24/7), and at times asymetric. 
New Zealand & transatlantic cases carried other traffic, but still stayed 
within the classic ~90%-to-10% ratio of TCP/UDP.  

> And that was the link adjacent to the end-user, and was carrying no
> other traffic. Thus no amount of QoS, dif-serv, SLA, et cetera would
> have been of any help.
>
> My operational experience tells me that that's a generalizable truth.
> That unlike in the PSTN, in the Internet, if there's a path, it'll be
> sufficient, and if there isn't a path, no amount of QoS will fix the
> basic lack of ability to forward packets.

I would qualify your statement as "usually" the case. its the peak times 
(and transient problems) that are of concern.

As a side note, i try to listen to a few jazz stations in the states,
and its nasty in my afternoon, with streaming audio dropping on occasion
because of loss.  American football VoIP radio stations on Sundays are
also frustrating for me at times.

>    > This problem is further compounded when toll-quality service is
>    > expected because it assumes a default service model that is
>    > better than best effort.  This in turn can increase the
>    > probability that a form of call-blocking can occur with VoIP or
>    > IP telephony traffic.
>
> What's meant by "toll-quality service?"  I'm perhaps incorrectly
> reading that as "voice quality similar to that experienced by users of
> the PSTN."  If that interpretation is as intended, 

your interpretation is correct.  its a term used in the telco world, and 
I've heard it used in IP telephony groups of the IETF.  However, I shall
provide an explanation in the draft.

> it implies that the
> authors believe that IP telephony on the commodity Internet doesn't
> universally exceed that threshold.

the word universally is quite a strong word.  in terms of 24/7 and the
five 9's that the PSTN aims at, no, i don't believe the non-US portion
of the Internet "universally" exceeds the threshold for IP telephony.

>    > 3.1.  Alternate Path Routing
>    > This objective involves the ability to discover and use a
>    > different path to route IP telephony traffic around congestion
>    > points and thus avoid them. Ideally, the discovery process would
>    > be accomplished in an expedient manner (possibly even a priori to
>    > the need of its existence).  At this level, we make no
>    > requirements as to how the alternate path is accomplished, or
>    > even at which layer it is achieved -- e.g., the network versus
>    > the application layer.  But this kind of capability, at least in
>    > a minimal form, would help contribute to increasing the
>    > probability of call completion of IEPS traffic.
>
> I cannot begin to express the depth of my disagreement with this
> assertion.  _Nothing short of a backhoe_ will have a more adverse
> effect upon the likelihood of call completion than routing it on a path
> other than that of the production traffic.

then, I would assume you have expressed your total objection to TRIP.

-ken

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From: Henning Schulzrinne <[email-masked]>
Date: Fri, 01 Nov 2002 07:38:00 -0500

There have been approximately half a dozen published papers that have 
attempted to characterize VoIP quality over various parts of the 
Internet. Not surprisingly, there are generally three observations:
- some pairwise sessions almost always work well, with < 2% packet loss;
- some have intermittent "hick ups" with bursts of very high or 100% 
packet loss (which are sufficiently long that no FEC or other 
compensation mechanism can deal with them);
- some are persistently bad to unusable, typically but not exclusively 
for sessions between North America and Latin America and parts of Asia, 
as well as between non-NA regions.

Search in netbib (http://www.cs.columbia.edu/~hgs/netbib) for 
references, if you'd like more details. This is probably only marginally 
related to the WG, but I can dig up the citations. It's obviously pretty 
hard to deduce from these limited experiments involving maybe at most a 
dozen sites what the precise likelihood of success is. All reasonable 
estimates would put it significantly below 99.9%.

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From: Bill Woodcock <[email-masked]>
Date: Fri, 1 Nov 2002 09:38:58 -0800 (PST)

> of the Internet "universally" exceeds the threshold for IP telephony.

I'll try to be very clear: I do not mean that every IP telephony call
which I've experienced has exceeded the quality of the best PSTN call.

I mean that the distribution curve of IP telephony call quality, if you
will, is at or above that of that of PSTN calls.  Thus the best IP
telephony calls exceed the quality of the best PSTN calls, the 90th
percentile IP telephony calls exceed the 90th percentile PSTN calls, the
10th percentile IP telephony calls exceed the quality of 10th percentile
PSTN calls, et cetera.  Obviously the worst calls fail to complete, in
both cases.

As I said, this is my personal qualitative experience from daily use
consisting of many calls on a global SIP network.

                                -Bill