The core design element of a traditional digital PBX is the local transmission bus that connects to a port circuit card. Many port circuit cards may share a common local transmission bus, and a PBX system may have many local buses dedicated to designated port circuit cards housed in different port carrier shelves and/or cabinets. Port circuit cards are used to connect peripheral equipment devices, such as telephones and telephone company trunk circuits, to the internal circuit switched network, where the local transmission bus is the point of entry and exit. Voice signals transmitted from the port circuit card onto the transmission bus are in digital format. The transmission and coding standard used by all current circuit switched PBX systems is known as Time Division Multiplexing/Pulse Code Modulation (TDM/PCM). To fully understand the workings of the PBX circuit switched network, it is necessary to define the basic terminology (Figure 1).
Multiplexing is the sharing of a common transmission line (bus) for transport of multiple communications signals. A communications transmission bus is a collection of transmission lines used to transport communications signals between endpoints. TDM is a type of multiplexing that combines multiple digital transmission streams by assigning each stream a different time slot in a set of time slots. TDM repeatedly transmits a fixed sequence of time slots over a single transmission bus. In a PBX system, the transmission bus is usually referred to as the TDM bus.
A PBX TDM bus is used to transport digitized voice signals that originate as continuous (analog format) sinusoidal waveform signals. Digital sampling of a continuous audio signal is a technique used to represent the analog waveform in digital bit format. The sampling technique that has become the accepted standard for circuit switched communications is PCM.
Pulse Code Modulation
PCM is a sampling technique for digitizing the analog voice-originated audio signals. PCM samples the original analog signal 8,000 times a second. This is more commonly referred to as 8-KHz sampling. The sampling rate used to code voice audio signals is based on the frequency range of the original signal. To accurately represent an analog signal in digital format, it is necessary to use a sampling rate twice the maximum analog signal frequency, a calculation based on the Shannon theorem. The maximum frequency of human voice is about 3.1 KHz. This frequency was rounded up to 4 KHz for ease of engineering design, resulting in an 8-KHz (2 × 4 KHz) sampling rate for digitizing voice audio signals. An 8-KHz sampling rate translates into a one sample every 125 microseconds (8 KHz–1; Figure 2).
Each digital sample is represented by an 8-bit word (28 = 256 sample levels) that measures the amplitude of the signal. The amplitude of the signal is based on the power (expressed in units of voltage) of the electri- cal signal generated by the telephone transmitter/receiver in the handset. This signaling technique has become known as Digital Signaling 0 (DS0), or 64-Kbps (8 bits × 8 KHz) channel transmission format. The term DS0 was defined based on the Digital Signaling 1 (DS1) format used to describe a digital T1-carrier communications circuit supporting 24 64-Kbps communications channels.
The PCM samples generated from each communications system port are transmitted onto the TDM bus in a continuously rotating sequence based on the time slot assignments given to each port circuit interface (see below). Only a single PCM word sample is transmitted at a time; that is the entire electrical transmission line is reserved for use by only one port circuit for transmission of its sample signal. The PBX processing system monitors each port circuit’s transmission time assignment in the rotating sequence, controls when the sample is transmitted, and coordinates transmission of the sample between the originating and destination endpoints.
There are two standards for coding the signal sample level. The Mu-Law standard is used in North America and Japan, and the A-Law standard is used in most other countries throughout the world, although each uses the 64-Kbps transmission format. For this reason, PBX systems must be designed and programmed for different geographic markets. Using firmware downloads, system vendors and customers can program their PBX systems to support the local PCM standard. The early digital PBX systems used different hardware equipment based on the location of the installation.
To summarize the fundamentals of PCM:
1. 4-KHz analog voice signals are sampled 8,000 times per second (8-KHz sampling rate)
2. Each sample produces an 8-bit word number (e.g., 11100010)
3. 8-bit samples are transmitted onto the TDM bus at a 64-Kbps transmission rate
4. The samples from each port circuit are transmitted in a continuously rotating sequence
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