Subject: [Ieprep] Qualitative statement about quality

From: Bill Woodcock <[email-masked]>
Date: Wed, 13 Nov 2002 15:29:49 -0800 (PST)

For what it's worth, the CALREN2 folks asked us to talk about the INOC-DBA
phones at their meeting this afternoon, and we presented the talk by phone
from Nairobi.  The CALREN2 guys did a qualitative comparison between voice
quality on the VoIP phones between San Luis Obispo and Nairobi versus
voice quality on PSTN phones between San Luis Obispo and Berkeley, and
determined that the VoIP connection was far better, qualitatively.
Latency on the VoIP connection was hovering in the 600ms range, with about
6% loss.  Conversation was about 30 minutes long.  These were their
measurements and evaluation, not mine, and these are people who know what
they're talking about in both data and voice arenas.

                                -Bill

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From: "Nguyen, An" <[email-masked]>
Date: Fri, 15 Nov 2002 15:22:22 -0500

Mike,
 
        I meant G.711 being transported in IP networks. 
 
        An

-----Original Message-----
From: [email-masked] [mailto:[email-masked]]
Sent: Friday, November 15, 2002 2:37 PM
To: [email-masked]
Cc: [email-masked]
Subject: Re: [Ieprep] Qualitative statement about quality

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. 

Round-trip, same as anyone else means when they say latency.

[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