Nothing here is an emulator or a museum piece. The call is a real call,
the carrier is a real carrier, and the pages come from the live Internet
a few seconds before you read them.
A real number on the real PSTN
Asterisk 22.10.1, built from source, answers the call over SIP and
hands the raw audio to the modem process through
app_audiosocket. The codec is locked to
ulaw with transcode:prevent, because a
transcoder in the path destroys modem tones. A live PSTN call carried
13.2 s of clean audio: 661 RTP packets for 660 frames, zero loss.
V.32bis at 14,400, with V.42bis
The working configuration is V.32bis at 14,400 bps; V.34 does not
train over this path. Incompressible data moves at 1,444 B/s. The
same 20 KB of zeros moves at 9,979 B/s, which is the V.42bis
compressor doing its job and not a throughput claim. A 2,400 bps
V.22bis path also exists and measures 275 B/s.
The retroproxy transform
Every page is parsed, stripped of what a period browser cannot render,
scored for its actual article, downgraded and re-serialised. The byte
budget is set in seconds of line time rather than in bytes, so the
same page is allowed to be larger on a fast line than on a slow one.
Nothing to configure
The transform is transparent. It listens at the PPP gateway,
10.64.0.1:80, and port 80 is redirected
to it. There is no proxy host and no port number to type into Trumpet
Winsock, MacTCP or the Netscape preferences dialog. You dial, you open
a browser, it works.
Sixteen lines
Sixteen calls can be up at the same time, each with its own modem
worker, its own PPP session and its own address out of
10.64.0.0/16. The negotiated line rate of
every session is written down as it trains, and the transform reads it
to pick that session's budget.
Two numbers, two states
Pennsylvania on
+1 610 922 1996 and
Boston on
+1 857 342 5877, which
spells 857-DIALUPS. Both land on the same trunk group and the same
sixteen lines. See coverage for who can reach
them without paying for the call.