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From: goldstein@delni.enet.dec.com
Newsgroups: comp.dcom.telecom
Subject: Re: 7kHz Voice and ISDN
Message-ID: <2002@accuvax.nwu.edu>
Date: 8 Dec 89 16:55:39 GMT
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X-Telecom-Digest: Volume 9, Issue 564, message 6 of 8


In article <1933@accuvax.nwu.edu>, ttidca.TTI.COM!jackson%sdcsvax@ucsd.edu 
(Dick Jackson) writes...
 
>As I understand it, work is afoot to implement a standard for the ISDN
>defined 7kHz voice service, wherein audio sampled (presumably) at 16
>ksamples per second is encoded (using cunning modern techniques) at
>the ISDN bearer channel rate (64 kbps).
 
>Any comments? From those who know how the technology is progressing?
>From potential owners?

Funny you should ask.  Yes, there's a new ISDN 7 kHz audio bearer
service.  It makes use of 64 kbps ADPCM encoding.  (Digression:
Standard PCM uses 64 kbps to do 3.1 kHz audio.  ADPCM is more
efficient, so 32 kbps is essentially adequate for 3.1 kHz audio, with
only minimal distortion (modems might complain, humans won't).  So if
you use the ADPCM principle on the usual 64 kbps bandwidth, you can
get better audio.)

The network uses PCM to generate tones and announcements for ISDN
telephones in the telephony bearer service.  The 7 kHz standard says
that you begin all calls in standard 3.1 PCM mode, specifying that
it's really a 7 kHz call.  Once the two ends are connected to each
other, they do a handshake to confirm that they're ready to switch
into 7 kHz mode.  That way the terminals are in 3.1 PCM mode when
doing call setup (talking to the network) and in 7 kHz ADPCM mode when
actually communicating with each other.

This hack makes it essentially transparent to the network, which will
speed implementation.  You just need the chips in your telephones.  I
don't personally see much use for it in "handsets", given their cruddy
mic/speaker combos, but it could be very nice for speakerphones, audio
dial-up program services, remote broadcast feeds, etc.

     fred  (member, ANSI T1S1, speaking for himself)