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Path: utzoo!mnetor!seismo!kitty!larry
From: larry@kitty.UUCP (Larry Lippman)
Newsgroups: comp.dcom.modems
Subject: Re: Pending FCC ruling threat to modem users
Message-ID: <1496@kitty.UUCP>
Date: Thu, 25-Dec-86 00:25:08 EST
Article-I.D.: kitty.1496
Posted: Thu Dec 25 00:25:08 1986
Date-Received: Thu, 25-Dec-86 02:48:34 EST
References: <1572@brl-adm.ARPA> <3454@curly.ucla-cs.UCLA.EDU> <403@pixar.UUCP>
Organization: Recognition Research Corp., Clarence, NY
Lines: 120
Keywords: modems, bandwidth, carrier, sex (no, not really)
Summary: Arrgh!  Misinformation strikes again...

In article <403@pixar.UUCP>, aaa@pixar.UUCP (Tony Apodaca) writes:
> In article <3454@curly.ucla-cs.UCLA.EDU> stiber@zeus (Michael D Stiber) writes:
> >>First: MODEM calls DO NOT cost the phone company the same amount as
> >>other calls.

	For all practical intents and purposes, this statement is INCORRECT.
Read on for more details.

> >2) Modems use the same lines as voice.  The assertion that they tolerate
> >noise less well is irrelevant, since they do not get special priveleges.

	This statement is correct.

> I'm sure that this discussion is going hot and heavy in net.dcom.etc but
> I'll answer here anyway.  Flame off, Mike.  The point is true even if the
> rationale is messed up.  Modem calls DO cost the phone company more, for
> several reasons:
> 	1) They are continuous.  The dual carrier never stops.  Therefore,
> the phone company must supply bandwidth to the call continuously even if
> there is no "valid" data.  They cannot time-multiplex their signals.
> There are small breathing and thinking pauses in all voice conversation,
> and 99% of voice is half-duplex, even a teenage girl's conversations.

	If you are alluding to most telephone calls being statistically
multiplexed (for lack of a better simple term) by a process known as
TASI (Time-Assigned Speech Interpolation), you are wrong.
	TASI was first developed by Bell Labs during the 1950's to improve
the effective circuit capacity of transoceanic cables where the number of
channels was limited, and the per channel cost was very high.  TASI does
make use of the fact that an intertoll circuit is 4-wire, with separate
transmit and receive channels; during normal human conversation each
of the two channels is used only about 45% of the time.  Therefore, TASI
apparatus can almost double the effective channel capacity by assigning
a given intertoll trunk to a transmit or receive channel only when speech
energy is detected.
	HOWEVER, TASI apparatus is complex, expensive, and has some
objectional human factors (like clipping of syllables).  Operating telephone
companies have used TASI only for transoceanic circuits, and for some early
satellite circuits where TASI was cost-effective.  As far as I know, TASI
has never been used within the continental U. S. by an operating telephone
company.
	In recent years, there has been a resurgence in interest for TASI
apparatus, and a few vendors have offered TASI apparatus for use on
private line communication facilities - however, this does NOT apply to use
by operating telephone companies for the DDD switched network.
	So the point is: operating telephone companies do NOT vary the
"bandwidth" of telephone call by TASI or other means (with the exception of
transoceanic cable use).

> 	2) They are high bandwidth.  The phone line was designed with human
> voices in mind, and they are pretty low bandwidth, as everyone knows.  Also,
> everyone knows that their modems strive to get the most out of it, so they
> use it all up (if they didn't you'd buy a new one!).  However, the phone
> company "counts on" the signals being voice-like, so they can cram as many
> signals into one wire as possible, and a modem transmission screws up their
> frequency-division multiplexing.

	This is sick.  For all intents and purposes, the bandwith of any
dialed telephone call is _limited_ to being well under 4 KHz for several
reasons:

1.	Many central office subscriber loops are loaded to minimize the
	effects of capacitive loop attenuation.  The result of placing
	loading coils in the subscriber loop is a distributed L-C low-pass
	filter which drastically limits bandwith to less than 4 KHz.

2.	Frequency-division multiplex apparatus (like N-carrier) effectively
	limits channel bandwith to less than 4 KHz by virtue of bandpass
	filters.

3.	Time-division multiplex apparatus (like T1-carrier) effectively
	limits channel bandwith to less than 4 KHz since the T1 sample
	rate is 8 KHz.

4.	Digital central offices effectively limit channel bandwith because
	they too sample at typically 8 KHz.

	So the point is: there just ain't no way to achieve more than 4 KHz
in bandwidth on the switched DDD network (you're lucky to get 300 to 3,000
Hz).  There is no bandwith allocation scheme being used by an operating
telephone company that is somehow defeated by a modem.  Furthermore, the
bandwith required to transmit intelligible speech is the same as is used by
any dial-up modem.

> 	3) The carrier on some modems just happens to overlap a critical
> region of part of the phone company's equipment's frequency allocation.
> The circuits known as "echo suppression" use it to monitor themselves, and
> kick in higher bandwidth and better circuits if there is too much echo on the
> line.  If they didn't have it, you'd bitterly complain about the quality of
> your long-distance connections.  Voice doesn't have much of these freqs,
> but the carrier flips it out, causing it to allocate too much signal to
> your call.

	Echo supressors do NOT affect channel bandwith.  On a voice call
an echo suppressor detects speech energy (say on an E-W direction) and
inserts a large amount of attenuation (as much as 40 dB) in the opposite
direction (say, W-E in the above example).  This is why one person may
not be able to interrupt another until they stop talking on a DDD circuit 
with an echo suppressor.
	Echo suppressors are used sparingly in the DDD network, and generally
only on regional intertoll trunks.  Only ONE echo suppressor can ever be in
a given DDD connection.
	Echo suppressors are intentionally designed to be disabled by a
SF tone between 2,000 Hz and 2,200 Hz which lasts for at least 400 ms; it is
a necessary requirement that echo suppresssors be disabled for any data
call.  The primary purpose of the "answer tone" furnished by a modem is to
disable any echo supressor for the duration of the call.
	Your description of echo suppressor operation is totally wrong.

> 	I don't work for the phone company ...

	Perhaps you should - then you'd learn something about telephony.
I don't enjoy being harsh on you - but your article contained some serious
misinformation than has to be corrected.  I try to consider Usenet as an
educational forum, and it always troubles me to see articles such as yours.

<>  Larry Lippman @ Recognition Research Corp., Clarence, New York
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