Section 10: POTS Operation and Usage Tips
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T E C H N O L O G Y
Section 10
Section 10.
POTS Operation and Usage Tips: How it works
All analogue audio inputs are digitized by a high quality stereo 24-bit analogue-to-
digital (A to D) converter. This serial data stream is then fed to an extremely fast 32-
bit Digital Signal Processor (DSP) that encodes the audio data using the
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proprietary digital audio compression algorithm. After compression in the DSP the
resulting low bit rate serial data stream is fed to a data modem when in POTS mode,
or to an ISDN data modem when in ISDN mode.
At the receiving
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codec, the reverse process takes place. The audio data is
decompressed, fed to a digital-to-analogue (D to A) converter, then amplified to line
level and fed to a balanced output connector.
ISDN operation is very similar to POTS operation except that a fixed bit rate of
connection is used. The quality of sound at ISDN bit rates is near CD quality and the
delay is an insignificant 100 milliseconds.
10.1.
Historical Reflections
10.1.1.
Coding Algorithms
Over the past two decades, there have been great improvements in the way
that digital audio data can be condensed, while retaining the quality and
subtle nuances of the original signal. Many different algorithms have been
developed, including well-known MPEG layer 2 and 3.
These algorithms require a reasonably high bit rate, typically 64 kbps or
higher, for high quality wide band mono operation. Many ISDN codecs use
MPEG layer 2 and 3 for mono communications at 64 kbps and 128 kbps.
However, for stereo, the inherent MPEG encoding and decoding delays can
become a problem for real time operation.
While most MPEG algorithms may be suitable for ISDN operation, they are
generally inadequate for the low bit rates available with POTS operation.
Some manufacturers have tried to use MPEG for low bit rate POTS
operation but have found the results unsatisfactory. A typical POTS line will
achieve bit rates of less than 28,800 bps and few algorithms can deliver full
bandwidth, high quality audio at these very low bit rates.
Fortunately, significant advances in the development of coding algorithms
have made the design of digital audio codecs like the
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G
3
range
possible. The proprietary music and voice coding algorithms used by
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achieve compression factors of the order of twenty times or more. This
enables 15 kHz bandwidth high quality bi-directional audio to be transported
at a bit rate as low as 24,000bps. 7 kHz bandwidth voice quality audio can
be transported at bit rates down to 9,600bps. All bit rates have an end to end
latency of only 100 milliseconds. This insignificant delay provides the kind of
codec performance required for real time operation.
