ISDN for Studio Call-in Talk Systems


ISDN has become widely used for remote broadcasting. But does it offer anything to stations who want to improve the quality of call-in programming?

The answer is yes, most definitely. When the telephone network is extended to studios in pure digital form, quality is improved in a number of dimensions. Studio telephone interfaces using ISDN are being introduced now. These are especially suitable for installations which intend to keep audio signals as much as possible in the digital domain, and for the combined studio facilities which are becoming common under consolidation.

ISDN for Call-in Systems?

Broadcasters have discovered that ISDN service offers valuable digital access to the dial-up telephone network. High-fidelity ISDN MPEG codecs are widely used for remote and inter-studio feeds, and most major-market stations have ISDN lines installed for this purpose. The wide use of ISDN is one expression of the natural progression of analog to digital connectivity and electronics for moving and processing audio signals.

ISDN can also be used to enhance the quality of call-in shows. While not apparent to the many users of ISDN for MPEG codecs, the telephone network is able to cross-connect Plain Old Telephone Service (POTS) analog lines and ISDN. (Telos has had a POTS calling feature in our Zephyr codec since its introduction, but, according to our support engineers, most users are surprised to discover that it can be used to connect to analog phones.)

The cost of ISDN service is not a barrier. In most parts of the USA, the benefits of ISDN can be had for about the same cost as analog. The usual ISDN service has two channels and costs about twice as much as a POTS line. (Pricing varies around the country, at from a 20% discount to a 30% premium. The average is probably around a 10% premium.)

The Telco network is almost entirely digital–except for the "last-mile" copper connections from the central office to the customer site. Telephone switches are digital, long-distance calls travel over digital fiber strands, and many local phone paths use digital T1 lines. Despite this, the vast majority of users interface to the network via an analog technology that is little different from that employed in Alexander Bell’s days. To support the immense installed base of analog telephones, central office equipment includes a stage which converts the digital signals to analog for connection to POTS subscriber lines.

Within the Telco network, calls are routed over 64 kbps channels. A sampling rate of 8kHz is used, with a word length of 8 bits. The 8kHz sampling rate supports a Nyquist (audio cut-off) frequency of 4kHz. In practice, telephone systems are designed to have audio frequency response extending to 3.4kHz in order to allow relatively simple roll-off filters to be used. The word length is what determines dynamic range–and 8 bits would only permit 48 dB were it used in standard PCM linear fashion. A primitive kind of compression is used to stretch the dynamic range: µLaw in North America and much of Asia, and A-law in Europe. This is a scheme that equalizes the step-size in dB terms across the dynamic range–a smaller step-size on low-level signals reduces quantization noise and improves effective dynamic range to the equivalent of about 13 bits.



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