Basic Technology and Services

CHAPTER1 BASIC TECHNOLOGY AND SERVICES 1.1 PULSE-CODED MODULATION Voice has been one of the primary services in the communications industry. Voice, by nature, is an analog signal. First, an acoustic wave is generated as the vibrating vocal cords and the mouth cavity modulates it into recognizable and distinguishable compounded sounds that we call words. This acoustical signal is converted to an electrical signal by a transducer known as the microphone. The generated electrical signal is also analog; that is, it changes value in a con- tinuous manner with respect to time. At the receiving end, this electrical signal activates the electromagnetic coil of a speaker, another transducer, which re- produces the original acoustical signal. Initially, telephony entailed few basic functions such as ringing and call initiation (and number dialing), and an analog signal was transmitted over the telephone network. This service became known as Plain Old Telephone Service (POTS) and the telephones were known as POTS telephones. Soon thereafter, the analog signal was converted to a digital one, known as pulse-coded modulation (PCM), to form a binary (or digital) bit stream at 64,000 bits per second, known as digital signal level 0 (DSO). The circuitry responsible for converting an analog electrical signal to PCM and vice versa is known as a coder/decoder, abbreviated CODEC. Figure 1.1 illustrates an analog voice signal propagated over a twisted pair of wires (left side), also known as tip-and-ring (T&R); it passes through the CODEC circuit and on the right side we obtain a digitally encoded signal. A CODEC periodically samples the analog signal, and based on a conversion table, it translates each sampled value into a binary representation. There are two different representations or conversion tables. The one, known as the /x-law (mu-law), is used in the United States, and the other, known as the ce-law (alpha-law), is used in Europe. The acoustical signal of speaking voice, for all practical purposes, has a maximum frequency of under 4 kilocycles per second, or kilohertz (3.4 kHz). 3 4 CHAPTER 1 BASIC TECHNOLOGY AND SERVICES Digital Signal (PCM) Analog Signal 10100110001010100 A t Twisted Pair (T&R) INPUT OUTPUT PCM = Pulse-Code Modulation DS0 = Digital Signal Level 0 T&R = Tip and Ring Figure 1.1 Analog to PCM. Although voice (depending on the speaker) may contain higher frequencies, filters remove the frequencies above 3.4 kHz. It is proven that in order to decode PCM perceptually back to the (almost) same voice signal, the analog signal must have been sampled at least twice its maximum frequency content. This is known as Shannon's theorem, developed and proven by Shannon while work- ing at Bell Laboratories. Thus, a CODEC samples the electrical equivalent of analog voice 8000 times per second (2 X 4000), or every 125 yu.s, and it converts each sample into 8 bits PCM. Consequently, in every second there are generated 8000 X 8 = 64,000 bits, or a bit rate of 64 kilobits per second (64 Kbps); see Figure 1.2. This is a 64-Kbps channel and the bit rate is termed DS0. 1.1.1 Adaptive PCM Besides the 64 Kbps, methods have been developed that use sophisticated digital signal-processing algorithms to compress the 64 Kbps to 32 Kbps, or to 16 Kbps, or even to lower than that. These methods, known as differential ljoiooi loojoioioiooi A CODEC : : 0123456789 10 t Time Slot Time Slot —». t 125}Jsec 125,jisec Sampling analog signal Converting each sample to 8,000 times per second 8 binary PCM bits {0,1} and (ONE sample/125 Jisec) place them in contiguous time slots of 125 jisec Figure 1.2 DS0 rate. SECTION I 1.2 TIME-DIVISION MULTIPLEXING 5 PCM (DPCM), adaptive DPCM (ADPCM), and sigma-delta PCM (2APCM), are compression techniques, each identifying the particular algorithm used. 1.1.2 Local Loop In traditional telephony, the user's equipment is a POTS telephone that trans- mits an analog signal over a pair of twisted copper wires to the service provider equipment, where the CODEC is located. This pair of copper wires is also known as a local loop cable. Copper wires may be placed underground or on poles. These cables are susceptible to environmental electrical interfer- ence and noise, and in long loops and legacy systems chokes or filters had been placed to filter out the high-frequency content (above 3400 kHz) of the analog signal. 1.2 TIME-DIVISION MULTIPLEXING Since the "digitization" era, all communications systems and networks that support voice transport are based on the 8-kHz sampling rate (or an integer multiple of it), on 64-Kbps channels, and on the 125-/xs "quantum" interval. In a traditional digital network, the POTS telephone converts the acoustical signal into an electrical signal. The electrical signal is then transmitted over a pair of copper wires to a communications system where the CODEC function is performed. From that point on, the network does not know of analog signals but only of digital, and this is what constitutes an all-digital communications network. Now that the network has become all-digital, in addition to voice, we can also pass through it digital data (raw digital data, encoded video, encoded sound, etc.). An earlier data service is the Digital Data Service (DDS). This takes advantage of the 64-Kbps channel and uses 56 Kbps for data and the re- maining 8 Kbps for "in-band" signaling, that is, bandwidth allocated to the network for its needs and not delivered to the end user. Another service is the Basic Rate Integrated Services Digital Network (BRISDN or BRI). The BRI uses two 64-Kbps channels, known as channels B, and a 16 Kbps subrate channel, known as channel D, to support a combination of voice and/or data services over a single pair of wires. BRI is defined to support, at the end user's request, any of the following modes: two B channels for voice and a D channel for data; one B channel for voice and the B + D channels for data; two B channels for data and the D for signaling; all channels for data. Notice, however, that BRI can only be supported if both the user has Integrated Services Digital Network (ISDN) equipment and the communications system supports ISDN. The CODEC in this case is located at the end user's equipment. Table 1.1 lists the DSO services and the various formats per service. The formats listed in the table will be explained in a subsequent section. 6 CHAPTER 1 BASIC TECHNOLOGY AND SERVICES Table 1.1 DSO Rates and Line Codes Source Format, Overhead Line Code Voice: 64 Kbps Frame, bit robbed DDS 56 Kbps + 8 Kbps signaling 64CCC B8ZS B-ISDN 64 Kbps 64CCC B channel (2B + D) B8Zs Abbreviations: CCC, clear channel capability; B = 64 Kbps voice or data channel, DDS, digital data services; D = 16 Kbps signaling or data channel. The user-to-network interface (UNI) is the interface where the user signal, a 64-Kbps data stream (organized in 8-bit bytes), first meets the network. At the UNI, however, signals from many users arrive. These signals are synchro- nized with the system clock, 8 kHz or a multiple of it. Then, based on a round- robin principle (a periodic, sequential and circular polling scheme), the signals are sequentially polled one byte at a time and placed one after the other in a fixed order, a process known as byte interleaving, and the bit rate is upped. The location of each byte source in this ordered digital signal is known as time slot. This process is known as time-division multiplexing (TDM), see Figures 1.2 and 1.3. At the receiving end, time-division demultiplexing takes place. That is, based on the round-robin mechanism, the time slots in the received TDM signal are extracted and each is distributed to its destination. ;0 1 0 0 1 1 0 0 1 125 usec TS1 TS23 TS24 Time Slot !0 10 0110 0! jl 010010 ill 1000111F! DSO il 01 00 101; 23 Fs=100011011100 ; Time Slot Fe=DCDFDCDFDCDFDCDFDCDFDCDF |1 1 0 0 0 1 1 1 l>=Data Link, C=CRC, F=Framing 24 Time Slot !•-• rClock & Sync Figure 1.3 DS1 rate—Ml Mux. SECTION I 1.3 DS1 RATE 7 1.3 DS1 RATE When 24 bytes (a byte from 24 different signals) are time-division multi- plexed, the beginning of the byte sequence must be marked to distinguish it from the previous 24 bytes. Adding a single bit, the F-bit, does this. The 24 bytes and the F-bit together construct a DS1 frame. In a DS1 frame there is a total of 24 X 8 + 1 = 193 bits and all 193 bits are transmitted within 125 /xs, or at a bit rate of 1.544 Mbps, known as digital signal level 1 (DS1). The F-bit constitutes a subrate channel of 8 Kbps. The function of this time-division multiplexer is known as a Ml multiplexer (Figure 1.3). Although a single bit (the F-bit) cannot mark the beginning of a single frame, 24 bits over 24 frames, or a superframe, can. Thus, the F-bit becomes meaningful when it is collected over 12 or 24 frames (depending on the system). The basic function of the F subrate channel is threefold: to mark the beginning of a frame (a function known as framing), to provide a data link over which network data are sent between the transmitting and receiving end, and to provide error control.

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