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From: goldstein@carafe.enet.dec.com (Fred R. Goldstein)
Newsgroups: comp.dcom.telecom
Subject: Re: Data Lines vs. Voice Lines
Message-ID: <12708@accuvax.nwu.edu>
Date: 27 Sep 90 18:15:23 GMT
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In article <12623@accuvax.nwu.edu>, motcid!benyukhi@uunet.uu.net (Ed
Benyukhis) writes...

>> Voice is not packet data.  It is not treated in a packet manner.
                ^^^^^^^^^^^
>It could be.  VSCS (FAA) at Bell Labs is implementing just that i.e.
>Packatizing voice for air traffic controllers communications.  Voice
>packatezation perhaps warrants some discussion/explanation by someone
>more familiar with the process.  How about it Pat????

[Moderator's Note: Fred Goldstein comes to mind.  PAT]

This discussion sounds like a rerun of one we had last year, around
TASI, but I'll jump in anyway.  I've even sent a contribution to ANSI
T1Y1 for their next meeeting explaining why Digital voted NO on a
proposed packet voice standard (syntactic and checksum matters), but
that's beyond the scope of this thread.

Packet voice does occur on the public switched telephone network, but
it's not common.  Old-fashioned Time Assignment Speech Interpolation
using analog gear has gone the way of the FDM open wire carrier.  But
newer digital interpolation gear does exist, mostly in private nets
and in international calls.  It's not worth the effort for domestic
calls, since raw transmission is cheap enough and any packetization
adds delay, making echo cancellation (not so cheap) necessary.

AT&T uses a device of their own manufacture called IACS (Integrated
Access & Cross-Connect Switch, if I recall) to compress international
telephone calls.  (Undersea cables aren't cheap!)  An effect of the
fax explosion, which they reported at a T1S1 meeting, was that they've
had to reduce the number of derived channels from each physical pipe,
since there are no gaps in fax modem tones.  So yes, modems do add a
little to the cost of calls, but only overseas.

IACS uses a technique called Embedded Adaptive Delta Pulse Code
Modulation.  This is like ADPCM except that the low-order three bits
of a five-bit sample are not used for predicting the next sample.  So if
the network is particularly busy, it can throw away the lowest-order
bit or two from each speech sample, by truncating the last 32 bytes in
a frame which carries speech samples arranged by bit significance.
Thus the tail end of the frame is all low-order bits.  On average you
may get 30 kbps or so, but during the busy hour it may drop and still
sound okay (just not quite as good) and at real off-hours you may get
over 32 kbps and sound better than normal.

BTW, the ISDN service definitions for "telephony" and "3.1 kHz audio" 
differ in that the former permits the use of speech processing, TASI, 
etc., while the latter requires that the network listen for the 2100 Hz 
disable tone that modem calls begin with.  ISDN interworks with the 
analog network using the 3.1 kHz audio service.

But again, for the bulk of domestic toll and essentially all intra-LATA
and local calling, you get raw circuit mode and the network doesn't
care one whit about whether you have a modem or microphone.

Fred R. Goldstein   goldstein@carafe.enet.dec.com 
                 or goldstein@delni.enet.dec.com
                    voice:  +1 508 486 7388 
opinions are mine alone; sharing requires permission