[e2e] Skype and congestion collapse.
RJ Atkinson
rja at extremenetworks.com
Sat Mar 5 08:16:03 PST 2005
On Mar 5, 2005, at 05:32, Jon Crowcroft wrote:
> if we all took 576kbps DSL lines and sent 24*7 MPEG2, we would (today)
> cause trouble - but in a few years time, who's to say what the minimum
> entry point is?
Datum:
With DOCSIS cable modems, which are pretty widely used today throughout
Europe, North America, Australia, Japan, and other parts of Asia, a
given
cable modem subscriber has a single upstream frequency available. On
that
shared upstream frequency, the standard permits either QPSK or QAM
modulation.
In practice, European cable plant will permit QAM to work yielding
rather
more usable bandwidth. Outside Europe, cable plant rarely supports QAM
(from an RF S/N perspective), so one is stuck with QPSK. This means
that
the shared upstream can typically get (outside Europe) about 1.5 Mbps.
It is unlikely that the cable plant would be reworked to support QPSK,
because that is relatively expensive. It is practical, however, to
space
(georgraphically) division multiplex upstreams so that fewer customers
are sharing a given upstream.
Even so, I would not be optimistic that a cable modem end user would
ever
really be able to use 512 Kbps (or more) of upstream capacity (except in
the much smaller geographic area, nearly all fibre, cable plant in
selected
parts of Europe where QAM could be used upstream).
[Typically, the DOCSIS upstream is operating in the lower "roll off"
region
of the RF spectrum (e.g. ~27 MHz), where the CATV RF transmission gear
is
becoming marginal. This is done because there is (today) more revenue
from
carrying an additional TV channel than from giving that same bandwidth
to
upstream cable modem use. The economics could change at some point,
though right now that seems unlikely.]
With DOCSIS, the shared downstream is significantly less of an issue,
because RF S/N ratio is much better downstream and because the cable
operator normally allocates one TV channel (not in the roll off region)
for that purpose.
> you know what: users do NOT like variable quality -if you aregoing to
> support a given rate, dont go ABOVE it if you are later going to have
> to go back down
> to that rate 0 if you have a lower rate, only support that. this is
> critical
> for audio (but less so for video) - so all the work on fancy codecs
> and user/channel/codec adaption we all did 2 decades back for 10 years
> - you know,
> was all misguided.
Very interesting result, IMHO. Thanks for describing both test regime
and results. Is there a suggested bibliographic citation or two to
read ?
> so if you look at the Book we wrote on all this stuff (Internetworking
> Multimedia, Handley/Crowcroft/Wakeman -
> morgan kauffman pubs), we were wrong (though all the stuff on rtp and
> sip and realtime on IP and multicast there is
> is still pretty up-to-date:)
Sounds like time for a 2nd Edition. :-)
Cheers,
Ran
rja at extremenetworks.com
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