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Message Digest
Volume 29 : Issue 11 : "text" Format
Messages in this Issue:
Re: AT&T asking FCC for "end date" of switched network..
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Date: Sun, 10 Jan 2010 10:55:38 -0800 (PST)
From: "harold@hallikainen.com" <harold@hallikainen.com>
To: telecomdigestmoderator.remove-this@and-this-too.telecom-digest.org.
Subject: Re: AT&T asking FCC for "end date" of switched network..
Message-ID: <0bd98547-b41c-43cc-beb7-f5a30f603fa4@s31g2000yqs.googlegroups.com>
> In the old analog days, a group was not 12 circuits on undersea cable,
> but 16 circuits (3 kHz bandwidth instead of 4 kHz) a few groups would
> be assembled into a set for TASI. The circuit gain was well above
> 100%.
How is the circuit gain well over 100%? If it's a two way conversation
with each person talking 50% of the time, it seems that you'd get a
gain of 100% by dropping in other circuits during the idle times in
each direction. There are, of course, other pauses, and people don't
start talking right when the other person talks (though my wife says I
start before she's finished, but that's another story). So, the
circuit gain COULD go a bit above 100%. I was assuming they would
leave some reserve circuits to handle the times when more than 100% of
the people were talking in one direction, but your mention of dropping
the encoding level in digital TASI makes sense. They could, I suppose,
drop bandwidth per channel on analog TASI, but that seems complex. So,
for analog TASI, did they leave some excess circuits to handle the
peaks?
This sort of multiplexing is interesting. The data stream is bursty
(even in analog, since there are pauses with no speech). Without
multiplexing, you're stuck with allocating a full circuit to the data.
With multiplexing, though, the bursts of one stream can fill in the
valleys of another stream, decreasing the required bandwidth. With
real time streaming, though, there are times where the peaks coincide
and you run out of bandwidth. Moving from real time to near real time
by adding buffers, you can shift the peaks such that they do not
coincide, then restore the timing at the receive end. I think this
works very well for stuff like satellite television where a bunch of
channels can be put in one datastream. A scene change in the video
results in a spike in the data rate since the entire screen has to be
redrawn, instead of just the changes. But, this spike can be time
aligned with a static scene where no data is to be sent. With a large
enough buffer, you can get real high usage of the available bandwidth.
Any excess can, of course, be used for non - realtime data, such as
program guide info, subscriber authorizations, etc. With voice,
however, latency is really a problem since the speakers become
confused. I often get confused on a digital cellphone and start
talking over the person I'm talking to (especially my wife, but,
again, that's another story).
Moving away from TASI a bit, but still staying with statistical
multiplexing, I've been giving more thought to network neutrality and
traffic shaping. If indeed ISPs are suffering from network congestion,
prioritizing of traffic may make sense. Email probably does not have
to be delivered as quickly as interactive video. I believe IP has bits
allocated to packet priority. ISPs could charge different rates based
on the packet priority. They could also go to time of use metering to
try to even out the load. Economics could provide a method of
allocating limited resources.
Anyway, thanks for the TASI discussion. It's been a long time since I
saw that system at the AT&T underground building in San Luis Obispo.
They had a pamphlet entitled "San Luis Obispo, Communications Center
of the World" showing the various Pacific undersea cables.
Harold
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End of The Telecom digest (1 message)
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