Topband: Remote SDR Receive only setup

[DF3CB]

> I have found that this limits latency to something like 50 ms and the
> audio is relatively high fidelity because the standard 56k codex is
> quite good, compared to any kind of VOIP, which is optimized for, guess
> what, voice, not weak signal CW.  In general, you cannot get this kind
> of latency over the internet, and if you could, it would require BOTH
> the remote internet and the control point internet to have low latency.

I can agree to some point. However, it's a fact that most of the latency is
not generated on the internet line itself but on your own (client) computer
and in particular with soundcard processing of the remote audio. I am also
operating my station fully remote-controlled for DX, the station is 30km
away from home. I've gone very deep into the problems around latency (it's
almost a science) and developed my own VoIP software called RemAud
(available at http://df3cb.com/remaud/). Another important factor are the
right audio buffer sizes. More to read about that at
http://df3cb.com/remaud/concept.php. 

The internet line latency can be as low as 20ms in my environment but can be
as high as 80 or 100ms in the evening time. Not much to do about that. But I
was surprised how long soundcard processing can take. I measured it and
found out that there is a steady soundcard buffer overrun on my computers
adding another 50 or even 200 or 300ms latency. The soundcards are not able
to process a high number of small audio chunks - necessary for low latency -
in an appropriate time. What I do with my software is to watch the number of
audio chunks going into the soundcard and counting the number of processed
audio samples at the same time. If a certain buffer overrun, set by a
software parameter, is exceeded I begin to omit some of the incoming audio
chunks and let them **not** process by the soundcard. This doesn't play any
matter for CW or SSB. It might for digital modes. You might not even notice
the dropped-out audio samples but latency can be absolutely reduced and
minimized. (To whom it may concern, the next step are ASIO soundcard
hardware drivers).

I've also experimented with other VoIP software and different Codecs,
compressed and uncompressed and my favorite is PCM, 8kHz needing a rather
low internet bandwidth and reaching an estimated radio audio quality of 95%.

73 Bernd DF3CB