Subject: Re: [Ieprep] Qualitative statement about quality

From: [email-masked]
Date: Fri, 15 Nov 2002 14:36:41 EST

In a message dated 11/14/2002 2:01:08 PM Eastern Standard Time, [email-masked] 
writes:

>       On Thu, 14 Nov 2002 [email-masked] wrote:
>     > We really need a clarification of this claim. What do you mean be 
> "latency"?
>     > Do you mean one-way or round-trip delay of the voice? Or something 
> else.
> 
> 

[MAP] Yes, I presumed that you were referring to round-trip speech path 
delay, but I had to check, since I can't imagine how anyone could claim that 
a voice conenction with 600 ms was "better" than the PSTN. I am quite certain 
that no where on the face of the earth does the PSTN have a round-trip speech 
path delay anywhere approaching 600 ms. (You can only get that with a 
satellite hop, which few use anymore.) As Henning correctly said, 'G.114 says 
150 ms one way delay as upper limit for "regular" service'. Further, I.380 
(Y.1530) defines a parameter for packet delay as one-way and Y.1541 gives 
100ms is the objective for "Class 0" and 400ms for "Class 1". Class 0 is for 
delay sensitive applications such as "high interactive speech and video 
teleconferencing", while Class 1 is for less interactive speech and VTC. As 
Henning pointed out, the application you were referring to sounded like an 
esentially one-way presentation, in which delay is less important.

In a message dated 11/15/2002 4:07:43 AM Eastern Standard Time, [email-masked] 
writes:

>  The quality of the VoIP calls is much, much higher than that
> of PSTN calls, and it's difficult to keep the PSTN calls up...  Initial
> call-completion rate is only about 80%, and most calls are dropped within
> twenty minutes.  We've never had a failed connection or a dropped call
> between this pair of VoIP phones, by contrast.
> 

[MAP] Now I'm seeing it. You're comparing with the normal quality (Connection 
Setup Error Probability and Connestion Premature Disconnect, which are 
important) for calls to Nairobi. Yes, I'm sure there are local factors in 
Nairobi that cause higher failure rates than what is generally expected in 
other places. But please don't extrapolation that result to apply to the 
whole "PSTN".

And An - the delay for G.711 is practically non-existent. It samples the 
voice every 0.125 ms, turns it into 8 bits, and sends it on its way. The 
decode does the reverse in .125 ms. Any delay (in the PSTN) is due to time 
slot assignment in TDM multiplex systems and time switching in nodes. These 
are on the order of 10 ms per switch. Then there is the speed of light, which 
we can't change (yet). If you are talking of other delays, you are probably 
referring to a packetization delay, that is, time to gather up enough .125 ms 
samples to make a reasonably size packet (so that the IP overhead doesn't 
kill you). So typically this takes 20 ms for 160 samples (160 octets). This 
packatization (and delay), of course, does not occur in circuit switched.

How about the other part of the claim - that the IP connection had about 6% 
loss? I presume you mean packet loss ratio. My information tells me that many 
codecs fall apart at that loss rate, or at least the degradation in audio 
quality is very noticable. The criteria (which is normally met) for a digital 
connection through the PSTN is for a bit error rate much less than that, so 
again, I can not possibly imagine how one could claim that the Voice over IP 
connection you cited was better than the PSTN. Y.1541 suggests that the 
objective should be 0.1%.

You referred to the comparison being to a call from San Luis Obispo to 
Berkeley. I imagine that the one end is on the Berkely campus, i.e., probably 
not the "public" PSTN all the way. Maybe a lot of old, bad wiring or analog 
switches on campus with lots of interference. Yes, when you deal with some 
real long, old loops, the voice quality can be worse, but this is not a fair 
comparision. (I, like many others, have only several hundred feet of analog 
loop to the digital concentrator.)

I (and the world) would certainly be interested in any fair comparison that 
shows that a voice over IP call achieves better quality than the PSTN (taking 
into account all of the parameters defined in ITU-T Y.1530 and Y.1540).

Mike Pierce
Artel

---

From: Bill Woodcock <[email-masked]>
Date: Fri, 15 Nov 2002 12:07:11 -0800 (PST)

On Fri, 15 Nov 2002 [email-masked] wrote:
    > I am quite certain that no where on the face of the earth does the
    > PSTN have a round-trip speech path delay anywhere approaching 600
    > ms.

Dang, you mean that all that faster light that ISPs have been clamoring
for has all been hoarded by the PSTN?  I knew this wasn't a level playing
field!

    > (You can only get that with a satellite hop, which few use anymore.)

Let me give you another hint, Mike, before Rex and some of your colleages
rip your head off: your use of the word "few" is perhaps, um, injudicious.

The prevalence of Satcom tail-circuits is one of the pillars of your
argument justifying your need to be special, remember?  You don't want to
downplay it.

    > The application you were referring to sounded like an

I believe it was characterized by all parties concerned as a
"conversation."  It was fully interactive, as most telephone conversations
are.

                                -Bill

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---

From: RJ Atkinson <[email-masked]>
Date: Fri, 15 Nov 2002 21:20:19 -0500

On Friday, Nov 15, 2002, at 14:36 America/Montreal, [email-masked] 
wrote:
> How about the other part of the claim - that the IP connection had 
> about 6% loss? I presume you mean packet loss ratio. My information 
> tells me that many codecs fall apart at that loss rate, or at least 
> the degradation in audio quality is very noticable. The criteria 
> (which is normally met) for a digital connection through the PSTN is 
> for a bit error rate much less than that, so again, I can not possibly 
> imagine how one could claim that the Voice over IP connection you 
> cited was better than the PSTN. Y.1541 suggests that the objective 
> should be 0.1%.

Bill,
	Which specific codec(s) were you using ?

Mike,
	I would not want my life depending on a codec that falls apart at 6%
packet loss.  Please don't deploy that one, pick something more robust.

> Berkeley. I imagine that the one end is on the Berkely campus, i.e., 
> probably not the "public" PSTN all the way. Maybe a lot of old, bad 
> wiring or analog switches on campus with lots of interference. Yes, 
> when you deal with some real long, old loops, the voice quality can be 
> worse, but this is not a fair comparision. (I, like many others, have 
> only several hundred feet of analog loop to the digital concentrator.)

Mike,

I actually think it is a pretty fair comparison -- unless you want to 
agree that
the Internet should not include access links in its performance 
measurements
(since those are the equivalent of "not the public PSTN all the way" 
for the IP
world).

Ran
[email-masked]

---

From: Bill Woodcock <[email-masked]>
Date: Fri, 15 Nov 2002 18:25:16 -0800 (PST)

> Which specific codec(s) were you using ?

g711ulaw both ends, which is the default for the Cisco phones.

    > probably not the "public" PSTN all the way.

Actually the Berkeley end was in my house which, for voice, has a
seven or eight year old 26-gauge 50-pair back to the CO, about 4kfeet
away.  SLO end was on campus, and may well have been through analog
switches or wiring even worse than the PSTN.  Nairobi end was residential.

                                -Bill

---

From: Bill Woodcock <[email-masked]>
Date: Fri, 15 Nov 2002 18:27:53 -0800 (PST)

> [email-masked] wrote:
    > about 6% loss? I presume you mean packet loss ratio.

Correct.  If you look at the subsequent ping, you'll see a somewhat higher
loss rate.

                                -Bill