Basic Concepts of Communications: An Introduction · Basic Concepts of Communications: An...

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DPRO-89990Richard A. Costello, Ray Horak

Technology Overview24 September 2003

Basic Concepts of Communications: An Introduction

Summary

To develop a solid understanding of communications technology, one must be firmly grounded in a widerange of basic concepts in both the voice and data domains.

Table of Contents

Technology Basics

Multiplexing Techniques

Transmission, Switching and Internet Telephony

Intelligent Networks (INs)

Mobile Communications

Wide-Area Data Networks

Local Area Networks

Communications Equipment

Management and Control

Communications Software

Technology Analysis

Business Use

Technology Leaders

Insight

List Of Tables

Table 1: Steps in an Internet Telephony Call

List Of Figures

Figure 1: The OSI Reference Model

Figure 2: HDLC Framing

Figure 3: Types of Data Transmission

Figure 4: Start and Stop Bits

Figure 5: Transmission Methods

Figure 6: Transmission Signal Attributes

Figure 7: Multiplexing Techniques

Figure 8: Pulse Code Modulation (PCM)


© 2003 Gartner, Inc. and/or its Affiliates. All Rights Reserved.DPRO-8999024 September 2003 2

Figure 9: Telephone Network Architecture

Figure 10: Voice on IP Networks

Figure 11: Basic Intelligent Network (IN) Including SS7 Signaling

Figure 12: Mobile Communications Network

Figure 13: Cabling Types

Figure 14: LAN Topologies

Figure 15: Bridges, Routers and Gateways

Figure 16: Network Management Functions



Technology Basics

Telecommunications is defined as the transfer of information over a distance. The information can beaudio (for example, voice), image, video, computer data or any combination. The voice or videoinformation can be transferred in its native analog format over an analog network, or it can be transformedinto a digital format for transfer over a digital network. Similarly, computer data can be transferred in itsnative digital format over a digital network, or it can be transformed into an analog format for transfer overan analog network.

In the voice and video domains, the telephones accept acoustic analog audio inputs, and the camerasaccept reflected analog light inputs on the transmit side of the data transfer, and telephones and TVmonitors recreate them on the receive side. In the domain of data communications, the data may take theforms of text, graphics and images in a wide variety of formats including documents, spreadsheets, e-mail, customer records, performance records, traffic tallies, still images and video. The originating anddestination computing devices can be dumb terminals, PCs, high-performance workstations,minicomputers or mainframes.

The network connecting the terminal devices may be either analog or digital in nature or a combination ofthe two. It may be a dedicated network, connecting the terminal devices directly, or it may be a switchednetwork. The network may operate over a local area, a metropolitan area or a wide area. The networkmay be may relatively simple in nature or quite complex, perhaps comprising a great number of networkelements, the most prevalent of which are transmission and switching systems. There may be a greatmany switching devices interconnected by a great many circuits, thereby offering a considerable numberof alternative physical routes or paths for the data to take as it moves between the communicatingterminal devices.

Prerequisites for Communications

Specific conditions must be met before communications can occur. Regardless of the nature of theinformation exchange, certain standards for conduct must be understood, agreed on and followedcarefully. For example, two people attempting to exchange ideas must agree on a common language forcommunications to take place. Otherwise, no communication occurs, even though words are spoken byone and heard by another. The communication is more complete and effective if the two people can seeeach other, for facial expressions and body language can be quite meaningful.

Certain rules and conventions must be followed in order that the communication is as effective aspossible. For example, if the communication takes the form of an interactive conversation rather than alecture, the two people must take turns talking, and the listener must be attentive. The voices must bemodulated so that the volume and frequency levels are acceptable—it is difficult to hear a whisperedword, yelling and screaming can be very unpleasant for the listener, and a monotone may put the listenerto sleep. The pace of the speaker must take into consideration the ability of the listener to absorb andunderstand the data transfer. If there is excessive background noise, the volume levels may have to beincreased, the pace may have to be slowed and words may have to be repeated in order to compensatefor communications errors. These rules and conventions for communications can be characterized asprotocols.

Data communications networking involves a great many protocol considerations. Devices must alsospeak the same language and follow the same rules, and mechanisms must be in place to ensure thatdata travels from one device to another without errors. Unfortunately, commercially available data



communications devices speak a variety of tongues and follow a number of different rules, therebycausing real confusion among data communications users.

However, various organizations, including the International Telecommunication Union-TelecommunicationStandardization Sector (ITU-T), the International Organization for Standardization (ISO), the ElectronicIndustries Alliance (EIA) and the Institute of Electrical and Electronics Engineers (IEEE) have developedstandards that are widely recognized and used throughout the industry in the development anddeployment of networks, and in their interconnection.

Open Systems Interconnection

To communicate at all, and certainly to communicate effectively, devices must be compatible on variouslevels. The ISO’s Open Systems Interconnection (OSI) reference model consists of a set of internationalnetworking standards known collectively as X.200. The model defines a set of common rules andconventions that computers of disparate origin can use to exchange information (that is, communicate).The model is organized into seven distinct layers in order to segment software responsibilities. Althoughthe OSI model does not describe a specification for any particular network or network element, it doesserve as a reference point for the establishment of a standard data communications system.

Each layer of the OSI model defines a particular function involved in the transfer of data from onemachine to another. The model is structured in an upwardly compatible manner. If there is compatibilityon one level, it is assumed that compatibility exists on the levels beneath. These levels are briefly definedin figure “The OSI Reference Model.”

Figure 1: The OSI Reference Model

The ISO OSI reference model defines a seven-level hierarchy for data communications. Depending on the protocol,certain levels will not be implemented.



In very basic data communications configurations, the prerequisites are relatively few and involvecompatibility on only the first two layers of the model. In support of end-user applications across complexnetworks, however, hundreds of conditions must be met before communications can occur, andcompatibility must be established on all seven levels.

Interfaces

A physical interface (layer 1) defines the physical connection between two network elements, specifically,between a device and a transmission medium, such as twisted pair, coaxial cable or optical fiber. It iscomparable in some respects to an electrical plug-and-socket connection with “male” and “female”components. Data and control lines from a device terminate in a connector with pins that handle assignedsignal functions, such as carriage return, line feed and request to send (information). For example, EIA-232, more commonly known as RS-232, is the industry-standard interface for connecting data terminalequipment (DTE) and data communications equipment (DCE). Communications hardware and softwaredrivers also are found at this level.

Each pin in a 25-pin connector represents a standard specification. Pin assignments are explicit andunalterable, except for those that are unassigned. Unassigned pins can be used to handle specialfunctions, such as “busy out,” on a modem, a condition that causes a modem to go “off the hook.” V.24,another common data communications industry standard, is functionally compatible with RS-232 and RS-449. It specifies standards for expanded transmission speeds, longer cable lengths and additionalfunctions.

Universal Serial Bus (USB) is a 12-Mbps serial bus developed by several leading PC andcommunications vendors to facilitate a “plug and play” capability for devices (for example, printers,modems, mice, telephones and joysticks) that connect to the communication ports on new PCs. The idea



behind “plug and play” is to have the PC automatically detect and configure any devices that areconnected to its ports. USB also includes a “hot attach” capability, which enables devices to beconnected, detected and configured (and disconnected) even when the power is on.

Coding Schemes

Information is created, stored, output from and input to computers in binary format, that is, 1s and 0s,each of which is known as a bit (binary digit). A coding scheme, which resides in layer 6 of the OSIreference model, defines a set of characters, including alphanumeric characters, punctuation marks andcontrol characters. Each character in the set comprises a unique bit pattern of a standard length known asa “byte format.” A byte commonly comprises an octet of eight bits, although some coding schemes specifybyte lengths of four, seven or even 16 bits.

For example, the widely used American Standard Code for Information Interchange (ASCII) is a seven-bitcode. Seven bits of 1s and 0s represent each character in the code set. The ASCII code consists of 128(27) characters, including 95 graphics characters (that is, letters, numbers and punctuation marks) and 33control characters (for example, tab, escape, backspace, line feed and carriage return). As the commonunit for storage and processing is an eight-bit byte, the eighth bit is used in various ways. In a pure ASCIIenvironment, it is used as a parity bit for error control—very poor error control. In DOS-based PCs, theextended ASCII code set of eight bits yields 28, or 256, unique combinations of bits, each of which is usedfor a character. Specifically, the additional 128 characters are used for foreign language symbols (forexample, Ö, ä and è) and graphics symbols (for example, a, b and m). In Macintosh computers, the 128additional values are user-definable.

Other commonly used codes are Baudot and Extended Binary-Coded Decimal Interchange Code(EBCDIC). Baudot, named after Emil Baudot and first established in 1874, is a five-bit code used onvintage teleprinter terminals, such as those made by Telex. It does not, however, have an error-checkingcapability or means of checking that the information is received. EBCDIC, the code used on synchronousIBM equipment, consists of eight-bit coding, which yields 28, or 256, characters. An end-user device thatoperates with one type of code (for example, ASCII) cannot accept data from a device using a differentcode (for example, EBCDIC) unless a conversion is performed to make one code compatible with theother. Code converters handle this relatively simple protocol conversion function in software.

Unix- and DOS-based operating systems, except for Windows 2000 and its predecessor Windows NT,use ASCII for text files. These systems use the newer Unicode, a 16-bit coding scheme that yields 65,536(216) unique character definitions. The value of Unicode is in its ability to support complex alphabets, suchas Japanese and Chinese. With over 40,000 defined characters, Unicode can support multiple alphabetssimultaneously.

Primarily for historical reasons, however, this largely is an eight-bit world. So Unicode TransformationFormat (UTF-8) was developed by the ISO to allow ASCII and Unicode to coexist on the same machine.This is relatively simple, as ASCII matches the first 128 characters of Unicode. Aside from UTF-8 andrelated UTF standards, various conversion programs allow different operating systems to change a filefrom one code to another.

Communications Protocols

Communications protocols cover a wide spectrum and range from single character-by-charactertransmissions with no error checking to complex rules for moving large amounts of data among manydevices. In general, communications protocols comprise three major areas:

• Data representation and coding



• Structure and meaning of overhead information

• Sequencing of information so that two devices can establish control, detect failures or errors, andinitiate corrective action

Prior to transmitting the data, protocol-specific software in the originating device chunks the data intodatasets, or payloads, of specified size and adds overhead bits for various control purposes. Thereceiving device checks the overhead bits as it receives the payload bits, perhaps checking for errorscreated in transit. If no errors are detected, the receiving device may send an acknowledgement,indicated that the dataset was received correctly. If, on the other hand, it detects errors, it generallyrequests a retransmission through a negative acknowledgement.

High-Level Data Link Control (HDLC) is one of the most commonly used layer 2 framing protocols. HDLCencapsulates layer 3 (Network Layer) packet data, adding data link control information. Variations ofHDLC are used in X.25 public networks and in frame relay public and private networks.

Figure 2: HDLC Framing

The beginning eight-bit flag alerts the receiving device that a frame of data is about to be received. The end flagsignifies that the transmission is complete. The address field names the station to which the information is beingsent (or in some cases the sending station). The control field identifies the type of frame and provides sequencenumbers of frames transmitted and received. The frame check sequence provides an error control mechanism.

Beginning and end flags are actually the same 8-bit sequence (1000 0001). In a typical sequence offrames, there is only one flag between frames, which serves to end one frame and begin the next. If oneor more bytes of data anywhere in the frame look like a flag, the receiver will get out of sequence and willbe unable to rebuild the payload. To avoid this, the sending end looks at every byte being coded andconverts it to a defined “escape” sequence if it detects a flag character in the payload. At the receiver, flagcharacters perform their task and are stripped away; the receiver then restores any detected escapesequences to the original coding.

Communications protocols are either bit-oriented or byte-oriented:

• Byte-oriented protocols communicate value strings in byte format, generally with eight bits in a byte.A string of bytes is organized into a block and encapsulated in a block, with overhead informationadded through the use of a header and trailer. Byte-oriented protocols are overhead intensive,generally are asynchronous in nature and generally operate in half-duplex (HDX) mode over dial-upcircuits. Binary Synchronous Communications (Bisync or BSC) is a byte-oriented protocol thatdefines specific characters for specific functions.

• Bit-oriented protocols transmit data value strings in bit format, with the data organized into frames ofany length up to a specified maximum; an acknowledgment can take place after one or several



frames have been sent, depending on the protocol specifics. Data (payload) is normally sent inblocks that range from 80 to 512 characters, with multiple blocks comprising a frame. Bit-orientedprotocols are less overhead-intensive than byte-oriented protocols, are generally synchronous innature and usually operate over high-speed full-duplex (FDX) circuits. Synchronous Data Link Control(SDLC) is a bit-oriented protocol from IBM.

As there is a great number of protocols, there are significant issues of protocol incompatibility at all sevenlayers of the reference model. Where these issues surface, protocol conversion must be accomplished insoftware, which often resides in devices known as “gateways.”

Data Transfer

There are two types of data transfer—parallel and serial. Parallel data transfer employs a communicationsinterface with a sequence of dedicated wires, each serving one purpose. A typical parallelcommunications cable has eight data wires, each of which is dedicated to transmitting a given bit in aneight-bit character byte. Thereby, the entire byte can be transmitted in one strobe, in parallel.

This form of transmission is, however, limited by distance. If one wire is slightly shorter or longer than theothers, the data from the eight separate channels will not arrive at the same instant in time and cannot bematched exactly. This phenomenon is referred to as “skew.” In addition to the data wires, there typicallyare a number of wires for various signaling and control purposes. The large number of wires makesparallel transmission impractical over distances greater than a few feet.

Most data communications devices transmit in a serial fashion. Used for long-haul communications, serialdata transfer operates bit-by-bit over a single wire rather than in parallel over multiple wires. Therefore,the data must be converted from parallel to serial bitstream prior to transmission. This protocol conversionprocess is accomplished through a shift register.

Figure 3: Types of Data Transmission

In a parallel transmission, each bit in a character is transmitted simultaneously on a separate circuit. In a serialtransmission, bits are transmitted in sequence over one circuit.

Synchronization

Data transmission can be asynchronous or synchronous in nature. Asynchronous transmission, oftencalled “start-stop transmission,” transmits one character at a time. Each character has its own “timing”device (that is, a start bit and a stop bit), which alerts the receiver to the beginning and end of eachcharacter. The time between the transmission of each character is referred to as idle time. Asynchronoustransmission is inefficient, due to the overhead required for start and stop bits and the idle time betweentransmissions. With asynchronous communications, transmission speeds as high as 33.6 Kbps can besupported in most equipment today.



Figure 4: Start and Stop Bits

During transmission, a data character consisting of seven or eight bits is preceded by a start bit and ended with astop bit that lets the receiving device know where a character begins and ends. A parity bit sometimes is used forerror control.

Synchronous transmission sends a set (that is, block, frame, packet or cell) of characters over acommunications link in a continuous bitstream. Data transfer is controlled by a timing device called a“clock.” Initiated at the sending device (for example, terminal, modem, multiplexer, switch or router), theclock runs at a frequency equal to the transmission rate. Each dataset is preceded by sync bits or aunique character pattern to allow synchronization, and special idle characters are transmitted if no data isbeing sent.

To accommodate the transmission of a large number of blocks, terminals involved in synchronous datatransfer must have buffers for storing the character blocks, as they wait to be transmitted or processed atthe receiving end. Synchronous transmission is generally used for higher-speed data transfers. In manycases, a device can operate both asynchronously and synchronously.

Transmission Mode

Three modes of data transmission are defined: simplex, half-duplex and full-duplex:

• Simplex transmission occurs in one direction only, with no possibility of response. Aside fromapplications such as alarm circuits, simplex transmission mode is unusual in contemporary terms.

• HDX transmission occurs in both directions, but only in one direction at a time. Therefore, there issome idle turnaround time as the direction of communications is reversed. HDX mode is commonlyused for relatively slow-speed transmission over dial-up circuits for applications such as modemcommunications, polling of remote buffers and transaction processing.

• FDX transmission occurs in both directions simultaneously. FDX circuits generally are four-wire, high-speed circuits, with examples being T-carrier and E-carrier.

Figure 5: Transmission Methods

Simplex transmission is unidirectional. Half-duplex transmission is bidirectional, but in only one direction at a time.Full-duplex transmission is simultaneously bidirectional.



Transmission Signal Attributes

All telecommunications systems and networks make use of electromagnetic energy to support thetransmission of data. Whether in the form of electricity, radio or light, all electromagnetic energy travels inwaves. The signal waveform has three fundamental attributes—frequency, amplitude and phase.Frequency is the number of cycles in a given time, usually a second, and is measured in hertz (Hz).Amplitude refers to the strength of the signal as illustrated by the height of the peaks of the waveform and,in the case of sound, determines the volume or “loudness.”

The figure “Transmission Signal Attributes” shows two sinusoidal waveforms in which B lags behind A.They are said to be out of phase with one another. Phase refers to the position of the waveform as it risesand falls and is of particular importance in modulation theory.

Figure 6: Transmission Signal Attributes

An electromagnetic signal has three fundamental attributes—frequency, amplitude and phase.

Analog vs. Digital

All communications can be characterized as either analog or digital in nature. Analog communications ischaracterized by the presence of a continuous electromagnetic waveform that varies in terms offrequency and amplitude. Digital communications is characterized by the representation of information inbinary form (1s and 0s) and is accomplished by the manipulation of the waveform in a series of pulses orblips of discrete value at specific points in time.



There are a number of ways that the electromagnetic signal can be manipulated, or modulated, to supportdigital transmission. The simplest approach is On-Off Keying (OOK), whereby the signal is switched on torepresent a 1 bit and switched off to represent a 0 bit. The signal is switched at precise points in time andremains on or off for precise periods of time, known as “bit times,” so that the receiving device knowswhat to expect and so that multiple adjacent 1s and 0s can be represented easily. OOK originated intelegraphy and is the technique used in most fiber optic systems. There are a great many much moresophisticated digital signal modulation techniques that involve various combinations of amplitude,frequency and phase modulation.

Bandwidth

Bandwidth refers to the information-carrying capacity of a communications circuit or channel. At theanalog level, bandwidth is defined as the range of frequencies that the channel is capable of transmittingwithout interference or signal loss and is measured in hertz (Hz). The greater the range of frequencies amedium can handle, the greater its information-carrying capacity. A broadcast-quality entertainment TVsignal, for example, requires bandwidth of 6MHz, whereas a voice signal requires a relatively modest4kHz.

In digital terms, bandwidth is generally specified in bits per second (bps). A 2-Mbps channel can support atransmission rate of two million bits per second. This rate can be accomplished through simple OOK, aswe discussed above, or through sophisticated modulation schemes that involve manipulation of theanalog carrier, with “carrier” referring to the frequency or frequency range that carries the communicationssignal.

Bits and Bauds

Baud is an old term used in the days when teletypewriters were considered cutting-edge terminaltechnology. Baud rate refers to the number of signal events (that is, signals or signal changes) occurringper second over an analog circuit. In contemporary terminology, baud rate refers to signaling rate of amodem and is roughly equivalent to Hz.

Bits per second (bps) refers to the number of binary bits transmitted per second. The impression of bits onan analog signal is the essence of a digital signal modulation scheme. In a modem communication, thebaud rate refers to the signaling rate, while the bit rate refers to the rate of information transfer.

Signaling Rate, Transmission Rate and Throughput

In the digital domain, there are even more subtle distinctions in terminology, which are no less important.A high-level discussion of T1 will serve to illustrate the differences between signaling rate, transmissionrate and throughput.

The signaling rate refers, once again, to the number of signal events occurring per second over the circuit.In an electrically based T1 circuit running over unshielded twisted pair (UTP), the signaling rate is 1.544Mbps. Although the specifics of the modulation technique can vary, the simplest technique involves ananalog carrier modulated at a nominal rate of 1.544 Mbaud. Therefore, it can be said that a baud equals abit in the case of this simple unibit modulation scheme and therefore that the baud rate equals the bit rate.

The actual transmission rate, which refers to the rate of information transfer, is 1.536 Mbps. Thedifference is 8 Kbps, or 0.008 Mbps, which is attributable to the fact that the signaling rate includes 8,000framing bits per second. These framing bits are not information bits (that is, payload bits). Rather, they arecontrol bits used variously for purposes of synchronization, circuit monitoring and error control, with thespecific uses being sensitive to the T1 framing convention involved. This transmission rate is under thebest of circumstances, as older circuit termination equipment requires additional signaling and control



bandwidth, which it gains by seizing some data bits and inserting signaling and control bits in their place.This process of bit robbing further reduces that actual data transmission rate, or payload rate.

Throughput refers to the actual amount of good data put across the circuit per second. If there are errorsin transmission caused by inherent circuit noise or some external source of electromagnetic interference(EMI), the integrity of some data may be affected, in which case the affected data must be retransmitted,which fact affects throughput negatively.

Transmission Quality

The quality of the transmission can be thought of as the conveying of information without the loss ofinformation or the addition of any unwanted information (noise). The quality of a transmission system isusually measured in terms of the signal-to-noise ratio, which is usually expressed logarithmically in termsof decibels (dB) and bandwidth. The equivalent measure of the quality of a digital system is the bit errorrate (BER).

Multiplexing Techniques

Multiplexing allows multiple channels to be derived from a single circuit, thereby allowing multiple lower-bandwidth transmissions to share a single higher bandwidth circuit. Although the high-bandwidth circuitgenerally is more expensive than a low bandwidth circuit, multiplexing allows a single multichannel circuitto be shared among many transmissions. The advantages, therefore, are increased efficiency and lowerassociated circuit costs. There are two basic approaches to multiplexing:

Frequency Division Multiplexing (FDM)

FDM was the earliest and most fundamental method of multiplexing. It divides the frequency spectrum ofan analog circuit into multiple independent, lower-speed subchannels, each of which operates on aseparate and distinct frequency band within the available spectrum. The bandwidth of each channel issensitive to the width of the frequency band. Therefore, the slower the transmission rates, the moresubchannels can be derived from the circuit. FDM supports multiple simultaneous transmissions, one perchannel, to coexist on the circuit, which is a distinct advantage in many applications.

FDM was the basis for early voice networks but has been replaced in large measure by time-divisiontechniques. FDM continues to find important applications, though, such as in the deployment of DigitalSubscriber Line (DSL) technology and in optical fiber media, where the technique is known as“wavelength division multiplexing” (WDM).

Figure 7: Multiplexing Techniques

Multiplexing increases the efficiency of communications links, allowing a business to lease a single shared high-speed circuit for less than it costs to lease multiple low-speed circuits.



Time Division Multiplexing (TDM)

Rather than divide a communications link into frequency-separated channels as FDM does, TDM dividestime into slices called “time slots.” With TDM, each inputting terminal takes its turn at transmitting andreceiving data in a continuous fashion. Depending on the multiplexer type, the device accepts only onebit, byte or packet of data from each input line, puts it into a specifically allocated time slot on the high-speed circuit and then moves on to the next terminal in the sequence. The process of accepting data frommany terminals in succession is called “interleaving.”

At each stage of TDM, the ensemble of groups of bits from respective channels, plus a flag, is called a“frame.” A flag is a special pre-defined pattern of bits that indicates the beginning of the frame andenables the receiver to work out which bits belong to which channel. Input signals are sampled one afterthe other at high speed; only one sample of a specific signal occupies the channel at any one time.

Pulse Code Modulation

Pulse code modulation (PCM) is a standard, fundamental process for converting analog data into thedigital format required for transmission over a digital circuit. Standardized by the ITU-T as G.711, PCMinvolves three steps:

• Sampling

• Quantizing

• Encoding

Figure 8: Pulse Code Modulation (PCM)

The analog waveform is sampled every T seconds. The samples are then converted to integer numbers using eightbits per sample. Finally, the integer numbers representing the samples of a given channel are sent in sequence as abinary bitstream to the TDM process shown in the lower part of the previous figure.



Sampling measures the amplitude of the signal at the given sampling rate in order to extract all theinformation. According to the Nyquist Theorem and Shannon’s Law, the two theorems that define thespecifics of PCM, the sampling rate needs to be at least twice the highest frequency of the signal in orderto encode the analog signal, send it across a digital circuit and reconstruct a high-fidelity facsimile.

If we take the bandwidth of speech to be 4kHz, then the signal must be sampled at least 8,000 times persecond (2×4,000Hz) to preserve the original information for transmission in digital mode. Further, thesamples must be taken at a precise pace of 1/8000 second, or every 125 microseconds. Each analogsample is then encoded into an 8-bit binary approximate value through a process known as quantization,based on a table of 255 standard values of amplitude. As this process takes place 8,000 times a second,the basic bandwidth requirement for high-fidelity voice is 64 Kbps.

Digital Transmission Systems (T1 and E1)

T-carrier was designed specifically to support digitized voice communications based on PCM. To achievebackward compatibility with established analog carrier systems, T1 specifies 24 channels. In traditionalPCM applications, each channel is 64 Kbps wide, consisting of 8 bits sent 8,000 times per second. Each8-bit sample is interleaved with 23 others to form a frame of 192 bits. A framing bit is added to indicate thebeginning of the frame and, variously, to support synchronization, circuit monitoring and error controlfunctions. The result is a frame of 193 bits. The signaling rate, therefore, is 1.544 Mbps (1,544,000 bitsper second). This consists of the aggregate rate due to speech (24 speech channels × 64 Kbps perchannel = 1,536,000 bps) plus 8,000 framing bits.

A collection of 12 frames is called a superframe (SF), corresponding to the format defined in D4, thefourth generation of channel banks, a type of circuit terminating equipment. A collection of 24 frames isused in the extended superframe (ESF) format. T-carrier is a North American standard.

The European and international version is E-carrier. E-1, the fundamental level of E-carrier, supports 30information channels, each of 64 Kbps. Two additional channels of 64 Kbps are used for synchronizationand signaling, respectively, yielding a total 32 of channels requiring aggregate bandwidth of 2.048 Mbps.The Japanese J-carrier standard is similar to T-carrier, but different enough to be incompatible.



Although T-carrier and its offspring were developed to support PCM-encoded voice, they are agnosticwith respect to the underlying applications. The transmission facility can’t tell the difference between voicebits, data bits, video bits or any other kind of bits—a bit is a bit is a bit. So, digital carrier systems are usedin support of the transmission of all varieties and combinations of data in digital format. Further, thetraditional channelization conventions often are ignored, as some applications require smaller channels(for example, compressed voice), some require larger channels (for example, high-speed data) and some(for example, frame relay- and broadcast-quality video) don’t lend themselves to channelization at all.

Voice Compression

There exists a number of techniques for compressing voice in order to reduce the bandwidthrequirements. In some cases, there is no perceptible loss in fidelity, while in other cases the loss in qualitycan be extreme.

Adaptive differential PCM (ADPCM) is a form of waveform encoding—one of the main techniques used inspeech compression. ADPCM takes advantage of the fact that human speech tends to be modulated inamplitude and frequency in a relatively smooth and gradual manner. ADPCM involves the digitalexpression (that is, encoding) of only the differences between samples, in the form of a four-bit digitalvalue, which results in a bandwidth requirement of only 32 Kbps. The use of ADPCM enables 48 speechchannels, rather than 24, to be carried on the 1.544-Mbps T1 facility.

A considerable number of other compression standards exist, some of which are standardized and othersof which are proprietary in nature. For example, the ITU-T G.728 standard defines 16-Kbps audio. Evenlower bit rates can be found in the wireless world, where radio channel bandwidth is precious and must beconserved at all costs.

The G.723.1 standard defines 6.3-Kbps audio for use on the “air interface” of cellular and PCS networks.The G.723.1 standard is also an implementation option when voice over Internet Protocol (VoIP)techniques are used. Naturally, there are quality issues at such low-bit rates; nonetheless, thesophisticated coders used in cell phones and VoIP gateways generally do a remarkably good job.

Digital Signal Processors (DSPs)

Special-purpose microprocessors with instruction sets designed to manipulate digital signals are termed“DSPs.” Originally developed to support encryption for military applications, DSPs currently are used in awide range of applications, including modem signal processing, video compression, TV enhancement andstraightforward speech compression.

Transmission, Switching and Internet Telephony

Transmission facilities, also known as transmission systems, comprise the links and circuits that serve tointerconnect devices. In the local loop, transmission systems most frequently are in the form of telephonecables made of copper wire. Long-haul systems increasingly are in the form of fiber-optic cables, althoughorbiting satellites or microwave-radio also are used.

Regardless of the specifics of the transmission systems, there are three primary means of establishingconnectivity: dedicated circuits, circuit switching and packet switching. Dedicated circuits are dedicated tothe interconnection of two or more devices and are not otherwise shared. Switched circuits allow thenetwork resources to be shared, and often highly shared, thereby eliminating the need for every device tohave a dedicated line to every other device. The design of networks takes into account the volume oftraffic generated by the users of the network and is usually designed to provide a specified grade ofservice (GOS), or network availability.



Dedicated Circuits

A dedicated circuit is a nonswitched circuit that directly interconnects two or more devices, such asconcentrators, switches or routers. Whether analog or digital in nature, the dedicated circuits do not gothrough any intermediate switching device. Therefore, it is not shared with other devices, attached endusers or user groups or applications. Such circuits, therefore, can be rented or leased exclusively to onecustomer and can be used for voice, data, video, multimedia or virtually any other sort of communicationsapplications. For example, a PBX tie trunk (see the figure “Telephone Network Architecture,” above)directly interconnects two or more PBXs via dedicated leased circuits. Companies with high levels oftraffic commonly prefer this arrangement, as it can be more economical than the pay-per-call structure ofthe public long-distance network.

Although a leased line does not go through a central switch, it nonetheless may go through a centraloffice, in which resides the network transmission equipment consisting of wire and fiber optic distributionframes, digital cross-connect equipment, multiplexing equipment and so on.

Circuit Switching

Circuit switched connections are temporary, continuous and exclusive in nature. That is to say thatconnections are established between links, or circuits, through a switching matrix on an “as available”basis, and only temporarily (that is, only for the duration of the call). During the course of the call, theconnection and all of the associated bandwidth are available continuously and exclusively in support ofthat call.

In a digital network involving a digital carrier system (for example, T-carrier or E-carrier), each voice call issupported by a specific channel comprising 8,000 specific time slots per second, as defined in theconnection setup process. Those time slots are dedicated to the call in progress, even during periods ofsilence (pauses between words and sentences, staying quiet while the other person talks or other lapsesin talking). The channel and its associated time slots remain assigned to that call and are released onlywhen the parties disconnect.

Circuit switching is defined as connection-oriented in nature. That is to say that a physical and logical pathis established prior to the commencement of the information transfer, and all data follow the same pathduring the course of the call.

Packet Switching

Packet switching was developed specifically as a cost-effective means of supporting interactive datanetworking, which is highly bursty in nature. That is to say that the communications are characterized byperiods of intense activity interspersed with periods of total silence. Circuit switching is not cost-effectivefor such communications, and dedicated circuits don’t provide the necessary flexibility.

Packet refers to a set of data organized in such a way that it can be accepted, switched and delivered asa unit through an internetwork. Each packet is independent as it works its way through the network, eventhough it may be part of a stream of packets. In the case of a data transfer involving a large text file orimage file, for example, the file is fragmented into multiple payloads of a defined maximum size and eachis formed into a packet. The packets are presented to the network in sequence. In the case of voice overa packet network, a number of PCM-encoded voice samples are gathered to make up the payload andformed into a packet. Packets are presented to the network in a sequential stream as long as the voiceconversation is active.

In any case, each packet in the sequence works its way through the network independently of others. Ifsome packets are lost, damaged or excessively delayed in transit, measures may be taken to rectify the



failure. Alternatively, the failures may simply be accepted, with the specifics of any error control measuresbeing sensitive to the nature of the application being supported.

To facilitate the handling of the packet through the network, various control data is appended. Dependingon the specifics of the network protocol or protocol suite involved, the control data is in the form of aheader, which precedes the payload, or both a header and a trailer, which trails the payload. In any case,the control data is overhead variously including originating and destination addresses, sequencing, formatdescription, payload type, priority level, path selection and error-control mechanisms.

Developed in the early 1960s in support of the Advanced Research Project Agency Network (ARPANET),X.25 was the first packet data network standard. Although it is fairly primitive by contemporary standards,X.25 remains heavily used. The most popular packet protocol is the Transmission ControlProtocol/Internet Protocol (TCP/IP) suite developed in the 1970s for what has become the Internet.

Note: Packet is a generic term for data organized into datagrams and formatted as payloadsencapsulated with control information in the form of a header and, perhaps, a trailer. In technology-specific terms, a packet refers to a unit of data (that is, datagram) carried in an internetwork (for example,the Internet), and a frame is a unit of data in a local network (for example, frame relay or LAN). The termcell describes a unit of information carried in a Switched Multimegabit Data Service (SMDS) orAsynchronous Transfer Mode (ATM) network.

Public Switched Telephone Network (PSTN)

The PSTN is a circuit-switched network designed to support analog voice communications. ContemporaryPSTNs are largely digital in nature in terms of both transmission and switching systems, although analoglocal loops from the customer premises to the network edge remain predominant in residential and small-business applications.

Medium and large businesses generally connect to a central office, or end office, at the network edge viadigital facilities, commonly in the form of T1 or E-1. Contemporary PSTNs in developed countries usedigital facilities to interconnect the switching centers. See the figure “Telephone Network Architecture.”

Figure 9: Telephone Network Architecture

Many facilities and switches are involved in a typical switch train. Designation (4) represents tandem trunks or entireinterexchange carrier (IXC) networks. In the U.S., end users pre-subscribe to exactly one IXC, although it can bebypassed in favor of a different IXC on a call-by-call basis through a “dial around” process. Tie trunks (6) arededicated leased line facilities that bypass the switched network to interconnect Private Branch Exchanges (PBXs)in a multisite enterprise.



A traditional PBX is a privately owned circuit switch located on the customer premises. A PBX enablespeople within the enterprise to talk to each other (station to station), to gain access to the public networkfor purposes of placing outgoing calls and to receive incoming calls from the public network. The latter twoare accomplished through the previously mentioned analog or digital trunk facilities.

Note: A trunk is defined as a link that interconnects switches. A PBX trunk interconnects a PBX switchand a PSTN central office switch. A tandem switch is a PSTN switch that interconnects other PSTNswitches, such as central office switches.

Internet Telephony

Internet telephony refers to communications services—voice, facsimile and voice-messagingapplications—that are transported over the Internet, a packet network based on the TCP/IP protocol suite.In general, use of the Internet in this manner bypasses significant portions, but not all, of the PSTN. Seethe figure “Voice on IP Networks.”

Figure 10: Voice on IP Networks

The top half of the figure shows a voice call routed to the Internet. The lower half shows a company’s private internalnetwork (intranet) carrying voice calls and data across an IP-based wide area network (WAN), as an example ofvoice/data convergence.



Phone to Phone on the Internet

The five steps involved in an Internet telephony call, performed by the originating network and reversed atthe destination, are listed in the table “Steps in an Internet Telephony Call.” Also listed are the entities atthe respective ends of the transmission facilities as the call progresses. For simplicity, calling and calledparties are assumed to be using analog telephones.


Description Entity at One End Entity at Other End

[1] Call origination in the normal manner from an

analog telephone

Calling party PSTN originating end

office, which routes the call

to the originating Internet

Service Provider (ISP)

[2] Conversion of analog voice signal to digital

(PCM) format

PSTN originating end office —

[3] Local transmission of PCM voice signal on

interoffice trunk (T1 or E-1)

PSTN originating end office Originating ISP’s gateway

equipment

[4] Compression/translation of PCM voice signal into

Internet Protocol (IP) packets

Originating ISP’s gateway

equipment

—

[5] Transmission of IP voice signal on the Internet Originating ISP’s IP packet-

switching infrastructure

(packet switch, router)

Internet backbone




Description Entity at One End Entity at Other End

[5] Transmission of IP voice signal on the Internet Internet backbone Destination ISP’s IP

packet-switching

infrastructure

[4] Decompression/translation of IP voice signal into

digital (PCM) format

Destination ISP’s gateway

equipment

—

[3] Local transmission of PCM voice signal on

interoffice trunk (T1 or E1)

Destination ISP’s gateway

equipment

PSTN terminating end

office, which routes the call

to the called party

[2] Conversion of PCM voice signal to analog format PSTN terminating end

office

—

[1] Call delivery in normal manner to an analog

telephone

PSTN terminating end

office

Called party

PBX to PBX on an Intranet

In a traditional PBX environment (lower half of the figure “Voice on IP Networks”), employees place IPtelephony calls using the same steps (in the table “Steps in an Internet Telephony Call”), but are precededby a dial access code (usually “9” in the U.S.) to reach an outside line. Should the PBX be IP-enabled, itwill perform the necessary gateway functions (that is, protocol conversion) to provide compatibility withlegacy end-office circuit switches over traditional analog or digital channelized circuits.

Terminology

Terms such as Internet telephony, IP telephony and VoIP are often used, or misused, interchangeably todenote similar types of calls and services. But Internet telephony refers to calls that are transportedspecifically over the Internet, while IP telephony typically refers to calls that are carried on any public orprivate network running the IP protocol. And VoIP is more properly a technology banner, implying defined(open) implementation standards. By convention, the public Internet is capitalized, while private intranetsare not, even though both are IP-based and intranets generally run over the Internet.

Packet Telephony Pitfall

As described above, circuit switching assigns time slots and dedicates them to the switch train,guaranteeing that talk paths are there when someone talks.

This very obvious and comforting situation is compromised when speech is chunked into packets andsent into a network designed to carry data packets—not digitized speech packets—and switched ontodifferent paths to or toward the destination. Time slots are not fixed along the same physical path for theduration of the call; indeed, it is possible that each packet could take a different physical path. As thelength of the physical path has a direct impact on the length of time that it takes a signal to travel betweentwo points, the result of multiple physical paths is jitter, which is variability in packet latency, or delay.Delay and jitter also can be the result of variations in network congestion levels from packet to packet.

As packet data communications are bursty in nature and as a great many transmissions commonly aresupported over the Internet at any given moment, congestion levels are variable and unpredictable. Ascongestion increases, packets may be held in buffer space in switches and routers at various points in thenetwork until the congestion eases and they can be forwarded. If a given packet is lost or errored intransit, it typically must be retransmitted. Variations in physical paths, congestions levels, and packet lossand error all contribute to latency, jitter and loss of sequencing.



Typical data communications applications, such as e-mail and file transfers, can deal with these issuesquite effectively, as there is always time to recover from latency, jitter and loss of sequencing in order toreconstitute the data in its original form. Stream-oriented communications, such as real-timeuncompressed voice and video, have a difficult time dealing with these issues, however, as time is of theessence and data must be played immediately and in the order in which it is received.

Intelligent Networks (INs)

The concept of INs—based on digital switching, database intelligence and ITU-T SS7 (Signaling System7) signaling—provides customers with a host of advanced network features and services, such asdistributed call processing, one-number services, class of service (CoS), private network services andnetwork billing enhancements, to name a few. As shown in the figure “Basic Intelligent Network (IN)Including SS7 Signaling,” the elements of an IN are:

• Service Switching Points (SSPs)

• Signal Transfer Points (STPs)

• Service Control Points (SCPs)

• SS7 Signaling Links

Figure 11: Basic Intelligent Network (IN) Including SS7 Signaling

SS7 signaling links connect SS7 network nodes. SSP voice paths (to PBXs, subscriber phones and other SSPs) areomitted for clarity.

SSPs are typically end offices, but can be tandem switches in sparsely populated areas with capabilitiesthat go beyond straightforward switching—enhanced software and hardware enable them to communicatewith application databases. The SSP formats and sends a request (an SS7 message) to an SCP, wherean application database is stored, and suspends call progress until a response is received that providesthe routing information needed to complete the call. An SSP may communicate with many different SCPs,depending on the number and variety of applications available.

STPs are essentially packet-switching systems used to transport messages—call setup messages androuting request messages—between SS7 nodes. As a cost-effective alternative to interconnecting allnodes directly to one another, STPs serve as centralized hubs in the SS7 network. Many nodes are linkedto a single STP, and in turn all STPs are interconnected. Messages sent to an STP are routed to thecorrect destination node. Because an STP connects to many SS7 nodes, it must be capable of handlinghigh-message throughput and must be equipped to support many signaling links. Redundancy andreliability are also key attributes.



SCPs are the centers of intelligence in the network. The main function of an SCP is to accept a query forinformation, retrieve the relevant information from the appropriate database and send a response to theoriginator (typically an SSP) of the request. Adding or updating databases, without affecting any othernode in the network, can increase the functionality of an SCP.

Integrated Services Digital Network (ISDN)

The ISDN—defined and described in ITU-T Recommendations—extends to the customer premises thecapabilities and benefits of the IN to support a wide range of voice and nonvoice applications over thesame digital network. ISDN is an end-to-end (customer premises-to-customer premises) digital networkthat integrates enhanced voice and image features with high-speed data and text transfer.

ISDN provides two interfaces—basic rate interface (BRI), also known as basic rate access (BRA), andprimary rate interface (PRI), also known as primary rate access (PRA). Both interfaces consist of Bchannels and D channels. The B channels, or bearer channels, are information-bearing channels thatprovide transparent digital channels for voice or high-speed data transmission at 64 Kbps per channel.The D channel, or delta channel, provides a nontransparent channel for signaling, telemetry and low-speed packet data at 16 Kbps or 64 Kbps:

• BRI provides two 64-Kbps B channels and one 16-Kbps D channel (2B + D).

• PRI provides thirty 64-Kbps channels and one D channel (30B + D) in Europe, which is backwardcompatible with E-1. In North America and Japan, PRI supports twenty-three 64-Kbps B channelsand one 64-Kbps D channel (23B + D), which is compatible with T1 and J1.

Mobile Communications

Modern mobile communications revolve around two main ideas. The first, “smaller is better,” breaks arelatively large geographic area into many contiguous cells. So a single macrocell is broken into manymicrocells, which can be further subdivided into many picocells. Radio towers situated approximately atthe center of cells communicate with mobile telephones (cell phones) and other devices (such as mobilepersonal digital assistants, or PDAs).

The second concept, “less is better,” reduces the power transmitted from respective towers, which is thebasic reason that the cells can be reduced in size. Taken together, reduced transmitted power and smallcellular areas enable radio frequencies to be reused. This has a multiplication effect, enabling many moreusers to be served using the same range of frequencies.

Cellular/Mobile Wireless Basics

The centerpiece of cellular/wireless networks is the mobile switching center (MSC), which interconnectssmall radio coverage areas into a larger system. See the figure “Mobile Communications Network.”

Figure 12: Mobile Communications Network

Base stations communicate by radio with mobile phones or other wireless devices. Land lines connect base stationsto an MSC, which tracks the location of devices that have been turned on.



Call Origination

When a device such as a mobile phone is turned on, it scans for an unused command channel, locks onand sends a registration request. To originate a call:

• The mobile phone sends a call request to the MSC by way of the base station. Individual basestations can simultaneously transmit and receive on many different radio channel frequencies andthus can be in contact with a large number of mobile devices at any given time.

• The MSC connects the call to the PSTN, which then connects the call to its destination (for example,home phone, PBX or perhaps another mobile phone in another locale).

• The MSC commands the mobile phone to switch to a talk channel (radio channel selected by theMSC) to hear call setup and, ultimately, call answer by the called party. When the parties hang up,the MSC instructs the mobile phone to switch to the command channel.

Call Delivery

To receive calls (known as “call delivery”), the mobile phone must be turned on and locked to a commandchannel. Assuming the call originates from a residence:

• The PSTN switches the call to the MSC, which has been tracking the whereabouts of the mobilephone from the moment it was turned on.

• The MSC sends a page (bearing the phone’s mobile telephone number) to the cell site, whichbroadcasts it to all phones in the cell site.

• When the target phone responds to the page command, it is commanded to switch to the talkchannel (radio channel selected by the MSC) for the call. The MSC then commands the phone toalert the user (ringing tone).



Handoff to the Same MSC

When a mobile phone is turned on (“available”) or is turned on and engaged in a conversation (“busy”), itmay traverse a cell site and find itself in a different cell site. At some point, the received signal strength ofthe command channel or the talk channel will decrease to the point where reception is no longer viable.The mobile phone monitors its received signal and sends signal strength measurements to the MSC.Signal strength below a predetermined level triggers handoff (of the mobile phone) to a base station witha stronger signal. A change in frequency occurs, coordinated by the MSC, as adjacent cell sites operateon mutually exclusive frequency “lists.”

Some cellular networks make “hard handoffs” through a “break and make” process that involves breakingthe connection with the original cell prior to making the connection with the next. This hard handoff isimperceptible in voice communications, but may cause a data session to be terminated. Other networksmake “soft handoffs,” which are acceptable in data communications as the connection is made to the nextsite before the connection is broken at the original site.

Handoff to a Different MSC

Conceptually, the situation is similar when the new cell site is served by a different MSC. The details arecomplex, though, as the MSCs must exchange messages (IS-41 or SS7 messages depending on thenetwork) to coordinate many actions:

• Selection by the first MSC of an interoffice trunk to the second MSC

• Radio channel selection by the second MSC

• Notification to the mobile phone to switch channels

• Teardown of the path to the old base station

• Establishment of a path to the new base station

• Confirmation to the second MSC that the mobile phone is operating on the new channel

After successful handoff, the talk path consists of PSTN switches and facilities, a trunk to the first MSCand the first MSC’s switch, a trunk to the second MSC and the second MSC’s switch, a trunk to the newbase station, a radio channel to the mobile phone and the mobile phone itself. Theoretically, the processcan continue indefinitely, the talk path accumulating additional MSCs (and trunks) during the mobilephone’s apparent odyssey. Practically, calls are finite, and even the longest ones terminate before theswitch train grows beyond several MSCs.

Roaming

Users sometimes transport mobile phones out of the home service area, a condition known as “roaming.”When the phone is turned on, emitting its mobile phone number, the MSC serving the mobile phone at thedistant location communicates with the home location register of the subscriber’s service provider to findout whether or not to provide service to the visiting phone (credit worthiness), as well as the serviceoptions and phone features that it should (or should not) honor. In this way, the service provider knowsthe whereabouts of the mobile phone, specifically, it knows the identity of the visited MSC, and the visitedMSC knows how to provide service as if the subscriber resided there. Call origination is handled by thevisited MSC and is the same as described above (that is, the visited MSC treats the roaming phone as if itwere one of its own). Call reception, however, is complicated by the fact that the phone is roaming.



Assume the call comes from a residence in Chicago, for example. It will be directed, via the PSTN, to thehome location register of the subscriber’s service provider (say, Dallas), which will notice that the mobilephone is out of the home service area. The following steps now occur:

• The home location register sends a message to the visited MSC (say, New Orleans) requesting thatthe visited MSC supply a temporary local directory number (TLDN). In North America, this is a 10-digit number bearing the area code of the visited MSC.

• The visited MSC selects a TLDN from its pool of numbers, associates the TLDN with the mobilephone’s telephone number (as sent in the original request from the home MSC) and sends the TLDNto the home MSC.

• The home MSC originates a new call, via the PSTN, using the TLDN as the destination address.

• The PSTN routes the call to the visited MSC, which matches the received TLDN to the mobile phonecurrently visiting.

• The MSC issues a page command bearing the mobile phone’s telephone number.

• When the mobile phone responds, the MSC commands commencement of an alerting signal (ringingtone) and commands the phone to switch to the channel that will carry the call. The subscriberanswers and converses with the originating party.

• When the call finishes (one or both parties hang up), the visited MSC releases the TLDN to the poolof available numbers, for use on another call.

TLDNs are not published, as they are reserved for internal network routing only. During a random callorigination, if a TLDN is dialed in error, the PSTN will route the call to the MSC (assumed to be in NewOrleans), which will provide an announcement that the call cannot be completed as dialed.

Wireless Communication Standards

There are a number of different mobile radio systems ranging from pagers to the pan-European digitalcellular radio system known as Global System for Mobile Communications (GSM). There also exists anarray of mobile communications technologies that focus on a variety of sub-segments of the mobilecommunications marketplace. That marketplace can be further subdivided into segments that addressdifferent aspects, characteristics or needs of its customer base.

There exist a great number of mobile phone systems and standards, all of which are incompatible. Thenarrowband cellular voice systems currently in use span 1G (first generation) analog and 2G (secondgeneration) digital systems, the most significant of which include:

• Advanced Mobile Phone System (AMPS): analog cellular in the 800MHz band

• Personal Digital Cellular (PDC), aka Japanese Digital Cellular (JDC): digital cellular running in the800MHz, 900MHz, 1400MHz and 1500MHz bands

• Digital-AMPS (D-AMPS), aka United States Time Division Multiple Access (US TDMA) and NorthAmerican-TDMA (NA-TDMA): digital cellular running in the 800MHz band

• Global System for Mobile Communications (GSM): digital cellular running in the 800MHz and900MHz bands

• Digital Cellular System 1800 (DCS 1800), aka Personal Communications Network (PCN): digitalcellular upbanded version of GSM, running in the 1800MHz band



• Personal Communications System (PCS), also known as cdmaOne and Code Division MultipleAccess (CDMA) Digital Cellular: spread spectrum digital cellular running in the 800MHz band

Data communications is difficult in 1G and 2G systems. The only datacom technology that hasexperienced any real success is Cellular Digital Packet Data (CDPD) in the U.S. Based on TCP/IP, CDPDis a packet data communications technique that operates over established AMPS analog networks attheoretical rates up to 19.2 Kbps, which generally translates into a practical data rate of no more than 9.6Kbps.

Emerging 2.5G and 3G cellular networks all fit under the umbrella of International MobileTelecommunications-2000 (IMT-2000), an initiative of the ITU-T intended as an international digitalwireless network architecture for the twenty-first century. The various 3G specifications include thefollowing speeds and intended applications:

• 128 Kbps for high mobility applications

• 384 Kbps for mobile applications at pedestrian speeds

• 2 Mbps for fixed Wireless Local Loop (WLL) and in-building applications such as Wireless Local AreaNetworks (WLANs)

Note: The above data rates all are theoretical best case rates. Actual data rates usually are much lowerdue to factors such as ElectroMagnetic Interference (EMI), Radio Frequency Interference (RFI), signalattenuation and line-of-sight issues.

A shortlist of next-generation 2.5G and 3G wireless standards, and their theoretical maximum data rates,includes the following:

• High-Speed Circuit Switched Data (HCSD): 2+G interim step toward 2.5G and 3G networks. Runsover the GSM network through the linking of up to four GSM time slots at 14.4 Kbps for a totaltransmission rate of 57.6 Kbps.

• General Packet Radio Service (GPRS): 2.5G enhancement to GSM. A packet-switched service thatsupports TCP/IP and X.25 protocols, with Quality of Service (QOS) differentiation. GPRS has beendemonstrated to run at speeds up to 115 Kbps and has a theoretical transmission rate as high as171.2 Kbps, although the actual data rate is generally much less.

• Enhanced Data rates for Global Evolution (EDGE): 2.5G standard characterized as the final stage inthe GSM evolution in Europe; also capable of running over D-AMPS networks in the U.S. Anintermediate step in the evolution to 3G Wideband CDMA (WCDMA), EDGE is planned to supportdata transmission at rates up to 384 Kbps.

• Universal Mobile Telecommunications System (UMTS), also known as WCDMA: 3G technologyintended to support data transmission rates of 128 Kbps for high-mobility applications, 384 Kbps atpedestrian mobility speeds and 2 Mbps for fixed wireless applications.

• CDMA2000, also known as IS-856: 3G technology based on earlier versions of CDMA. The initialversion, known as 1xRTT (1 times Radio Transmission Technology), is a 2.5G technology initiallyoffering data speeds up to 153 Kbps, with throughput in the range of as much as 90 Kbps. Theenhanced version, known as 1xEV-DO (1 times EVolution-Data Optimized), is an asymmetrictechnology offering peak data rates of up to 2.4 Mbps on the forward link and 153 Kbps on thereverse link. GSM1x is a version intended as a transition specification for GSM operators.



Clearly, the confusion over cellular standards is likely to continue for some time. Multimode terminalequipment resolves these issues of incompatibility in some cases. There are a number of other voice anddata mobile wireless solutions, some of which are standard and others of which are proprietary in nature.

Wide-Area Data Networks

Broadband ISDN (B-ISDN)

Early in the development of ISDN, the necessity was recognized for an advanced form capable of carryingmultimedia information at rates of hundreds of millions of bits per second. In the ITU, two main servicecategories have been defined: interactive services and distributive services.

Interactive services are subdivided into conversational services, message services and retrieval services.Conversational services are usually bidirectional, although in some circumstances they can beunidirectional and in real time between users, or between a user and a host. Examples includevideoconferencing and high-speed data transmission. Message services will offer communication viastorage units, such as a mailbox or as message-handling functions, which include not only speech butalso moving pictures and high-resolution images. Retrieval services offer user access to informationstored centrally and accessed on demand. Examples of these services include film and high-resolutionimages, together with audio.

Distributive services are differentiated between those services with user presentation control—such asbroadcast services for TV and radio—and those with individual user control. The availability of highbandwidth enables a number of different types of information to be supported by one service, resulting inthe development of multimedia services. For example, video telephony includes audio and video andpossibly text and graphics. Many of the broadband services—such as video signal transmission—requirevariable bandwidth, which is best met by a packet-based technology. For this reason, ITU chose ATM asthe target transfer mode for B-ISDN.

ATM

It is important to note that ATM is a transfer mechanism and as such is, in principle, independent oftransmission technology. It is a fast-packet switching technique that uses short, fixed-length packetscalled “cells.” As such, ATM is also referred to as a “cell-switching technology.”

In principle, it is quite similar to other packet-switched techniques; however, the detail of its operation issomewhat different. Each ATM cell is made up of 53 octets, of which 48 octets generally make up theuser information field (payload) and five octets generally make up the header (overhead). The headeridentifies the “virtual path” to be used in routing a cell through the network. The virtual path defines theconnections through which the cell is routed to reach its destination. ATM is a form of TDM. It differs fromsynchronous multiplexing in that channel separation is not dependent on reference to a clock.

Frame Relay

Frame relay also is a form of fast packet switching. Actually an ISDN framing convention intended for LANinterconnection, frame relay is a technique for gaining access to a packet data network, such as TCP/IPor ATM. It is used primarily in data communications environments, although standards-based voice overframe relay (VoFR) implementations are not uncommon.

As an ISDN spin-off and an interim technology designed primarily to serve both local-area network (LAN)interconnection and host computer environments, frame relay achieves about 10 times the packetthroughput of X.25 packet-switching networks by letting information move across a network guided andchecked by the following seven core functions of Link Access Procedure on the D-channel (LAPD):



• Flag recognition

• Address translation

• Transparency

• Frame check sequence/generation

• Recognition of invalid frames

• Discard incorrect frames

• Fill interframe time

A frame relay limitation is its lack of a mechanism for error detection and correction. These functions areleft to the end nodes in the network. This is a major QOS issue when considering frame relay for thetransport of voice communications. This lack of error control is not an issue in most data communicationsapplications, as there generally is time to recover from data loss or error. As a point of fact, the fully digitalnature of the transmission facilities, which generally are largely optical, and switching elements results inrelatively little loss or error.

Frame Relay, X.25 and TCP/IP Compared

Unlike its predecessor X.25, frame relay does not store and forward data frames, but rather simplyswitches to the destination part way through the frame, thereby reducing transmission delay considerably.Because storage requirements are minimal, frame relay is more cost-effective than X.25.

X.25 and TCP/IP are similar in that they are both packet-switched protocols. However, they differ in anumber of areas:

• Transmission Control Protocol (TCP) operates at layer 4 of the OSI model; Internet Protocol (IP)operates at layer 3. The X.25 protocol operates at layers 2 and 3. Frame relay is a layer 2 protocol.

• TCP/IP offers only end-to-end error checking, while X.25 is error checked from node to node (layer2). Frame relay does not support error detection and correction.

• TCP/IP has a much more complicated flow control and window mechanism than X.25. Frame relaydoes have a flow control mechanism, although the networks generally do not assume anyresponsibility for flow control.

• The electrical and link layers (layers 1 and 2) are tightly specified in X.25, while TCP/IP is designed totravel over many different kinds of media with many different types of link service (for example,Ethernet, frame relay, X.25, ATM, Fiber Distributed Data Interface [FDDI]).

DSL

There are a number of specific technologies in the generic Digital Subscriber Line (xDSL) family.Asymmetrical Digital Subscriber Line (ADSL), the consumer-level service with which most users arefamiliar, is a broadband service currently being offered by local exchange carriers (LECs) and ISPs inmany areas where the local loops can support the demands of the technology.

ADSL delivers broadband Internet access over UTP local loops up to 18,000 feet in length, as long as theloops meet fairly stringent requirements. For example, load coils are not acceptable, and mixed gaugesand bridge taps are undesirable. For technical reasons that are beyond the scope of this article, ADSLruns in asymmetric mode, with considerably more bandwidth provided downstream (that is, from the



network edge to the customer premises) than upstream. Currently, the maximum rates offered by mostcarriers are 1.544 Mbps downstream and 256 Kbps upstream.

The standards describe considerably higher rates, although the higher frequency signals demand that thedistances be shorter and that the overall quality of the local loops be fairly exceptional. A great advantageof ADSL is that the digital packet data channel is “always on,” meaning that it is not necessary to dialthrough the PSTN, as is the case with traditional dial-up modems. This not only eliminates a step, but alsoeliminates both dial-up delays and busy signals. Additionally, ADSL offers the advantage ofsimultaneously supporting an analog channel for voice and fax communications.

In addition to ADSL, the xDSL family comprises Very High Data Rate DSL (VDSL), High Bit Rate DSL(HDSL), Single Line DSL (SDSL) and ISDN DSL (IDSL), among others.

Cable Modems

Cable modems recently have emerged to compete with ADSL in the broadband access market. Cablemodems are customer premises equipment (CPE) access devices that enable computer equipment toconnect to the Internet over a digital cable television (CATV) network that simultaneously supports a greatnumber of entertainment TV channels. In the most prevalent scenario, a PC is connected to an externalstand-alone cable modem via an RJ45 Ethernet network interface. The cable modem is attached to aCATV network via an F connector, which is a 75-ohm coaxial cable connector commonly found ontelevisions and videocassette recorders (VCRs).

PCs are not the only equipment that can be connected to cable modems. As small office/home office(SOHO) environments expand and cable access extends into the business environment, cable modemscan be connected to hubs, switches and routers to allow networks access to the Internet. Some cablemodems are now including routing and four-port hub capabilities into a single cable modem device.Newer versions of cable modems include USB connections for PC connectivity and peripheral componentinterconnect (PCI) cable modem cards.

While the bandwidth that cable modems can support is impressive, it is shared among all users on thesame CATV network segment. Performance will vary depending on the number of users who are onlinesimultaneously and the type of work each person is doing. Access speeds can and generally will besignificantly less than what is theoretically possible. While bandwidth can be supported in the 10-Mbpsrange and higher, more realistic bandwidth numbers are in the 1-Mbps to 3-Mbps range for downstreamtraffic, and 250 Kbps to 2.5 Mbps for upstream traffic.

Synchronous Optical Network (SONET) and Synchronous Digital Hierarchy (SDH)

Digital telecommunications networks in North America (T-carrier), Europe (E-carrier) and Japan (J-carrier)were not based on a common standard, a fact which makes their internetworking somewhat cumbersome.Issues of compatibility aside, these conventional digital transmission networks simply didn’t have thecapacity to handle increasing traffic demands. Fiber optics certainly had the bandwidth to meet thosedemands, but all of the established systems were proprietary in nature, which meant that they could notinterconnect, and therefore the carriers were not able to enjoy the benefits of deploying multivendornetworks.

At the request of its client/owners, Bellcore (now Telcordia Technologies) proposed to American NationalStandards Institute (ANSI) the development of the. SONET standards for optical transmission in 1985. In1988, the first phase of SONET standards were released. The Comite Consultatif InternationalTelegraphique et Telephonique (CCITT), which is now the ITU-T, was following SONET progress andbegan work on an international variation in 1986. The World CCITT SDH standards in its 1988 Blue Book.



Although SONET and SDH differ, primarily at the lower multiplexing levels, those issues are fairly readilyresolved in situations where interconnection is desirable. SONET/SDH and ATM are now the coretransport and switching technologies, offering network operators and end users several advantages,including:

• Interconnectivity and interoperability.

• Increased available bandwidth.

• Reduction in network equipment.

• Increased network flexibility.

• Backward compatibility with legacy digital carrier systems.

• Improved availability and faster provisioning of services.

• Savings in maintenance and operations costs.

• Improved network restoration and reconfiguration times.

SONET/SDH is built on T-carrier standards. Essentially, SONET is a high-speed optical version of T3,which is a T-carrier multiplexing level supporting the equivalent of 28 T1s and running at a signaling rateof 44.736 Mbps. SONET adds some overhead for network management purposes, which brings thesignaling rate up to 51.84 Mbps for Optical Carrier-level 1 (OC-1), which is the foundation level. Like T-carrier, SONET is TDM-based, with frames transmitted 8,000 times per second at the precise pace of 125microseconds. The frame structure is nine rows by 90 columns, each of which columns is a byte wide. Ofthe total 810 bytes, 765 bytes are payload, organized as nine rows by 85 columns.

Should OC-1 not be satisfactory, the next step in sequence typically is OC-3, which comprises three OC-1frames and which runs at a signaling speed of 155.52 Mbps. OC-3 is the foundation level for SDH. At thatrate, all three OC-1 frames are transmitted in sequence every 125 microseconds. The SONET/SDHhierarchy currently includes OC-768, which is the equivalent of 768 OC-1 frames multiplexed and runningat a signaling speed of 39.813 Gbps.

Transmission Media

On analog public telephone networks, transmission facilities can be two-wire or four-wire circuits. A two-wire circuit consists of two copper wires, each in a color-coded, insulated covering; the two wires areloosely twisted around one another to improve the performance characteristics of the circuit. At the mostfundamental level, a four-wire circuit is a pair of two-wire circuits, with each pair supporting transmissionin one direction. Twisted-pair wire comes in many forms: some cables are waterproof, some have fireretardant coverings and some are shielded for extra protection against electrical interference.

In a local environment, in which terminals (for example, client workstations and printers) are attached to aserver in proximity, data transfer generally occurs over data grade unshielded twisted pair (UTP) wiring.Over distances of 100 meters or so, data-grade UTP performs well and at much lower cost than coaxialcable.

Coaxial cable, which was heavily used prior to the introduction of UTP in LAN applications, consists of twoconductors. A solid core center conductor supports data transmission, while an outer conductor of solidmetal foil or metal mesh acts as a shield against ambient sources of electromagnetic interference. A“dielectric” (that is, a material that does not conduct direct electric current) separates the two conductors.A cable that consists of two central conductors in the same mesh tube is called “twinaxial cable.” As the



gauge of the center conductor in a coaxial cable is relatively thick, it offers relatively little resistance to anelectrical signal and therefore supports much higher frequency signals than does twisted pair. Thattranslates into greater bandwidth over longer distances, which explains its traditional use in both CATVnetworks and data centers. However, coaxial cable is expensive, bulky and inflexible.

The increase in availability and lower prices of fiber optic cabling and equipment has resulted in itsincorporation into a large proportion of new networks. This is especially the case where there is a need tofuture-proof installations against the rising demands for bandwidth. As discussed in some detail above,SONET/SDH resolved concerns about the lack of standards in the fiber optic domain. More recently,optical multiplexing techniques have developed in the form of Wavelength Division Multiplexing (WDM),which allows multiple wavelengths to share the same optical fiber, much as multiple electrical signals canshare the same copper cable through Frequency Division Multiplexing (FDM).

Figure 13: Cabling Types

The three basic types of communications cabling: twisted pair (in this example, shown in a 25-pair bundle), coaxialcable (coax) and fiber optic cable.

Microwave communications describes point-to-point terrestrial radio systems operating within thefrequency range of 1GHz to 30GHz. Once exclusively the domain of the common carriers, microwavecommunications has become a major competitor of standard wireline telephone communications.Generally speaking, microwave communications involves sending information via high-frequency radiobetween a transmitter and a receiver. As such, high-frequency signals require a clear “line of sight”; theremust be no obstructions such as buildings or trees in the path between transmitter and receiver.Contemporary microwave systems are digital in nature and offer signal quality to copper-based conductedtransmission media.

Satellite networks are essentially nonterrestrial microwave systems with signal repeaters placed in space.Satellite transponders, which are the operational elements of satellites, receive radio signals originatingfrom an earth station (that is, terrestrial antenna), filter the noise out of the signal and amplify it, shift thefrequency to avoid interference between incoming and outgoing signals, and transmit them back across



the assigned footprint (that is, coverage area), where any number of terrestrial antennas can receive itsimultaneously.

Most contemporary communications satellites are Geosynchronous Earth Orbiting (GEOs) operating inequatorial orbital slots at altitudes of approximately 22,300 miles. Once positioned properly, the satellitesrotate around the earth at the same speed as the earth rotates on its axis. In other words, they aresynchronized with the earth (geo). Also known as geostationary, the satellites are therefore always in thesame position relative to any point on the earth’s surface.

Data Integrity: Error Detection and Correction

Any communications circuit or system can suffer from interference. Electrical storms, electromagneticfields from electric motors, cross-talk from other circuits in proximity and other phenomena can impact thecircuit or system and introduce errors into the transmission. Twisted-pair copper cable systems and radiosystems are particularly susceptible, but they all can be affected. Whether it’s one bit or several thousandbits that are errored or lost in transit, the integrity of the information is compromised. Every errored ormissing bit is a potential catastrophe that must be identified, isolated, diagnosed and corrected in someway, or not. Some applications just don’t have the time to deal with error and loss. Rather, they justaccept it. Real-time uncompressed voice and video are examples.

Those exceptions aside, a typical file transfer or program download might involve thousands or millions ofbytes. Many thousands of such files might transit the network at any given time at speeds of thousands ormillions or even billions of bits per second, and the loss of even one bit could alter a character or controlcode. Therefore, data communications generally requires stringent accuracy controls. These controlsconsist of bits added to characters and blocks of characters at the sending end of the line. These bits arethen checked and verified at the receiving end of the line to determine whether data was lost or erroredduring transmission. Two basic data communications controls include:

• Parity checking

• Cyclic redundancy checking (CRC)

In data communications, parity denotes a relatively simple error-checking technique in which the 1 bits ina character are summed and a 1 or 0 parity bit is appended to create either an odd value (if odd parity isdesired) or an even value (if even parity is chosen) prior to transmission. The receiving device examineseach received character for parity. If the devices are set for odd parity and a received character has aneven parity, it is assumed that one or more bits were corrupted and that the character is in error. Thesituation is similar in an even parity system if the receiver calculates an odd bit value.

The fundamental parity checking technique is known as vertical redundancy checking (VRC), in referenceto the fact that human beings add numbers in vertical columns and the process is redundant, with thereceiving device repeating the mathematical process executed by the transmitting device. VRC is highlyunreliable, as an even number of errored bits creates an undetectable character error. The reliability ofthe technique is improved considerably if longitudinal redundancy check (LRC) is also used. Thiscombined approach involves the process of checking for errors in a block of characters by examiningparity values both vertically and horizontally as though the bits were in a matrix format. In fact, it wouldtake an odd set of compensating bit errors to create an undetectable error in a block. Parity checking afairly primitive technique used in asynchronous communications.

CRC, a more powerful error-detection technique used in synchronous transmissions, uses calculatedpolynomial check characters. CRC views an entire block of data as one long binary number that is dividedby another fixed binary number. The result of the calculation is a summary description of the data, which



is truncated, generally as a 16- or 32-bit value. This is expressed as eight-bit block-check characters(BCCs) that are appended to the data block prior to its transmission. BCCs appear in the Frame CheckSequence field of HDLC frames (see the previous figure, “HDLC Framing”), for example. At the receivingend, the process is repeated as the BCCs are recomputed from the received data and compared to thoseappended to the block by the transmitting device. If the BCCs agree, the block is declared error-free. Ifthey disagree, it means that one or more errors exist in the block. The entire block is suspect and must beretransmitted.

Since an error detected in a message must be corrected fairly immediately, a transmitting device mustreceive an acknowledgment on a near real-time basis of the received data’s accuracy. Because data mustbe retransmitted if there is an error during transport, transmitting equipment must store all information thathas not been acknowledged by the receiving station.

There are a number of error-correction techniques, and more are being developed all the time. The twobasic methods are “Stop and Wait” Automatic Repeat Request (ARQ) and “Go back N” ARQ:

• The Stop-and-Wait technique involves sending a block of data and stopping transmission at the endof the block. The receiving equipment verifies the accuracy of the data and sends back anacknowledgment (ACK) or negative acknowledgment (NAK) if an error is detected. If an ACK isreceived, the next block of data is transmitted; if a NAK is received, the original block is retransmitted.Idle time is significant in a stop-and-wait environment. This technique is used in Binary SynchronousCommunications (BSC), for example.

• The Go-back-N technique requires a line with a return path (that is, one with a full-duplex capability)so that the ACK or NAK for one block can be received while the equipment is sending a subsequentblock. There is no pause in transmission at the end of a block, and the equipment proceeds totransmit the next block. If a NAK is received, the sending unit backs up the number of (“N”) blocks tothe last acknowledged block and retransmits all subsequent data. The maximum number of blocksthat can be transmitted before an acknowledgment is received is specified by the protocol. Thistechnique is used in SDLC, for example.

Yet another alternative is that of forward error correction (FEC). This technique involves the embedding ofredundant characters in the data block. This information allows the receiving device to identify, diagnose,isolate and correct errors without the necessity for a retransmission. FEC is widely used in networkswhere bandwidth is limited and expensive, with cellular data communications being but one example.

Local Area Networks

LAN Protocols

A LAN is a communications network that is usually owned and operated by the enterprise customer. ALAN operates over a limited geographical area and enables many independent peripheral devices, suchas PCs and terminals, to be linked to a network through which they can share centralized hosts, files,applications, printers and various other resources. Some organizations (for example, banks or financialinstitutions) have enough data traffic within a city to make intracity networking viable. In this case,individual LANs may be interconnected to form a metropolitan area network (MAN).

LANs are generally fault-tolerant, incorporating a simple architecture with control distributed amongparticipating stations. Since the entire network does not depend on a single polling or switching device,the failure of one component does not bring down the entire LAN. Standard LAN protocols largelyemanate from the IEEE 802 working group. Two major LAN standards are:

• Ethernet (802.3)



• Token Ring (802.5)

Ethernet was developed by Xerox and subsequently modified and standardized by the IEEE as 802.3.That original Ethernet ran at 10 Mbps over copper coaxial cable. In contemporary usage, the preferredmedium is data grade UTP wiring, although STP and coaxial cable sometimes are used.

As packet networks sharing a common medium, LANs must make use of some sort of access mechanismin order to manage contention for these limited resources. The access technique use by Ethernet iscarrier-sense multiple access with collision detection (CSMA/CD). This technique requires that eachworkstation listen to the network (that is, sense the carrier frequency) to determine if it is available beforetransmitting. If, however, two or more workstations are transmitting data over the same segmentsimultaneously, their signals will collide and the data will be destroyed. CSMA/CD allows the workstationsto detect this collision, at which point each backs off and calculates a random time interval, after which itattempts to access the network and retransmit the data.

CSMA with collision avoidance (CSMA/CA) offers an alternative technique that is particularly useful inwireless LANs (WLANs). With CSMA/CA, a device senses the carrier and issues a request to send (RTS).If the channel is available, the destination device issues a clear to send (CTS), which essentially advisesthe other devices to back off and allow the communication to take place without contention.

As the volume of network traffic increased, however, the bandwidth offered by traditional 10-MbpsEthernet LANs became inadequate in many applications. The first step in alleviating these bottleneckswas the introduction of 100Base-T, which makes use of relatively simple hubs to which terminal connectvia Category 5 (Cat 5) data grade UTP at speeds up to 100 Mbps. While essentially a brute force attackon congestion, 100Base-T is effective. The next step was the development of switched Ethernet, whichprovides multiple 10/100-Mbps paths for a limited number of devices through the switching matrix, ratherthan a single 10/100-Mbps “pool” of bandwidth for which all devices must contend.

More recently, Gigabit Ethernet (GbE) surfaced. Generally implemented as a switched Ethernet solution,1000Base-T originally required the use of fiber optic cabling, although high performance Cat 5 UTP canperform acceptably well over distances up to 100 meters if installed correctly. Cat 6 cables, which wereintroduced in 2002, are designed specifically to support 1000Base-T.

Developed by IBM, Token Ring typically runs on a ring topology, although star networks based onswitches are also possible. There are never any collisions in Token Ring. The first station to switch on in anetwork owns the “token,” a unique sequence of control bits. While it has this token, it is capable oftransmitting data, and no other workstation can take the token on that revolution. After the current ownercompletes transmitting its data, it passes the token to the next workstation on the network, and so on.

There are three basic LAN topologies: linear bus, ring and star. The topology can be defined as thephysical layout of the network.

Figure 14: LAN Topologies

The Bus topology is typical of Ethernet networks; rings are associated with token-passing (Token Ring) networks.



In linear bus topology, stations are arranged along a single length of cable that can be extended at one ofthe ends. A tree is a complex linear bus in which the cable branches at both ends, but which offers onlyone transmission path between any two stations. All broadband networks and many baseband networksuse a bus or tree topology.

In a ring topology, stations are arranged along the transmission path so that a signal passes through onestation at a time before returning to its originating station; the stations form a closed circle.

A star topology has a central node that connects to each station by a single point-to-point link. Anycommunication between one station and another must pass through the central node.

In bus and ring networks, all transmissions are broadcast. Any signal transmitted on the network passesall the network’s stations. In star networks, signals sent through the central node are switched to theproper receiving station over a dedicated physical path.

The LAN market can be divided into two distinct segments: large-scale LANs and server-based LANs:

• Large-scale LANs are those that interconnect a variety of end-user devices, including terminals,microcomputers, minicomputers, mainframes, computer-aided design/computer-aided manufacturing(CAD/CAM) equipment and various other machines.

• Server-based LANs are specifically designed for interconnecting PCs.

A LAN centralizes the control of an organization’s distributed computing resources and ensures that eachdepartment’s PCs are compatible with the network and with machines from other departments. Ideally,through a LAN the manager can make sure that all company decisions are based on the same data.



When used properly, the LAN provides a common interface for a diversity of otherwise incompatibleequipment.

In the office, the LAN can give users fast and efficient access to a common pool of information, such ascustomer lists, schedules and document formats. It also allows an entire office to share expensiveresources, such as printers and duplicators, thereby streamlining the production and distribution of paperdocuments. Ultimately, a LAN can eliminate the need to circulate paper documents by electronicaldistribution (e-mail, instant messaging) of memos and other textual materials. In an automated factory orlaboratory, the LAN can simplify the process of “retooling” by allowing the user to download data from anumber of programmable devices simultaneously from a central site. It can also isolate failures andbottlenecks in plant operations.

At one point, interest in fiber optic LANs was increasing rapidly, with several fiber optic systems availablefor LAN applications based on a mature optoelectronic technology. More recently, however, interest in thistechnology waned as copper cabling became more sophisticated, with Category 5 (Cat 5) easilysupporting data rates of 100 Mbps, and even 1 Gbps. Optical fiber remains highly viable over distancesgreater than 100 meters, or where electromagnetic interference or security is an issue.

Internetworking

The Internet now has millions of hosts connecting many millions of people all over the world. Educators,consumers, telecommuters, librarians, hobbyists, researchers, government officials and businesses areamong the groups today that use the Internet for a variety of purposes—from communicating andcollaborating with one another to accessing valuable information and resources.

The Internet provides connectivity for a wide range of application processes called “network services.” Forexample, users can exchange electronic mail, instantly message each other in groups, access andparticipate in discussion forums, search databases, browse indexes and transfer files. Also, use of theInternet for multimedia applications, including voice, is still in the early stage.

Internetworking refers to the connecting together of two or more networks, which may be LANs, WANs ora mixture of the two. As the Internet continues to grow in size, popularity and efficiency, as LANsproliferate in business environments and as enterprises rely on several communications networkssimultaneously, managers seek better ways to move information from one network to another.Internetworking devices take a LAN signal and send it further than the original LAN specification allows.

Devices generally fall into three types, though—as so often happens in communications—a single productcan incorporate more than one function:

• Bridges link networks that are fundamentally compatible. They operate at layer 2 (Data Link) of theOSI reference model, with no involvement at higher layers. Bridges can be intelligent, selectivelyforwarding or filtering frames across networks according to the layer 2 address of devices onrespective LANs.

• Routers are switching devices that connect two LANs where multiple paths exist. Operating at layer3 and above, depending on the specifics, they inspect addresses and route packets of data betweennetworks. Very often today the routing function is added to a switch to give layer 3 functionality to alayer 2 device.

• Gateways interconnect dissimilar networks, such as the voice-oriented PSTN and a data WAN forsupport of IP telephony. Gateways are highly sophisticated, incorporating protocol conversionfunctions. Gateways are also used to connect dissimilar LANs, such as Ethernet and Token Ring, or



to connect LANs to other types of networks or architectures, such as X.25 public networks orSystems Network Architecture (SNA) hosts.

Figure 15: Bridges, Routers and Gateways

Bridges operate at the data link layer of the OSI model, routers are the network layer devices that select pathways tosend data to destinations and gateways operate at the upper layers of the OSI model.

Communications Equipment

Private Branch Exchanges (PBXs)

A PBX is a telephone switch located on an enterprise’s premises that primarily establishes voice-gradecircuits over access lines between individual users and the PSTN. The transmission of wide area PBXcalls is still typically over the PSTN. However, enterprise telephony calls are increasingly being routedthrough gateway devices over Ethernet-based LANs and WANs, the IP-based Internet, and even ATMand frame relay networks.



A PBX is a private system in that it is typically used by one organization or one building complex withcapacity requirements ranging from less than 100 lines (but typically above 40 lines) up to thousands oflines. Branch suggests a remote subsystem, as the first PBXs were like small partitions of central officeswitches located on a customer’s premises. In telephony, an exchange is defined as a device thatcontrols the connection of incoming and outgoing calls—in short, a switch.

In addition to the solid reliability and performance that have been characteristics of traditional PBXtechnology over its life cycle, advanced functionality, such as IP telephony, call center technology,electronic messaging (voice, fax and even unified messaging), computer-telephony integration (CTI),broadband capabilities, PBX networking and in-building wireless communications, have all beenintroduced as PBX enhancements over the past decade or so. PBXs also have become less proprietaryas many now feature open application programming interfaces (APIs) that encourage third-party softwaredevelopment.

In traditional digital PBX systems today, time division multiplexing (TDM) is still the most commonlydeployed switching matrix design. However, the packet-switching designs in newer IP-based telephonysystems, in particular, are rapidly pushing TDM and circuit switching to the background. Traditional circuit-switched PBXs are now supporting interfaces to IP, ATM or frame relay infrastructures and equipment viaan assortment of new software, gateways, trunk cards and line cards. For example, most PBXs todaysupport VoIP functionality in various ways, ranging from offering VoIP gateway interfaces to IP networks,with supporting IP line/trunk cards, and IP telephones, or by adding VoIP functionality to its core switchingmatrix.

Key Telephone Systems (KTS)

Smaller organizations (typically with less than 40 stations) use a KTS rather than a PBX. Other than size,a major difference between a PBX and KTS is that a key system does not utilize an operator console. Theheart of a KTS is a key service unit (KSU) that is the common control cabinet for all major systemoperations and functions. A KSU performs CO line connections, intercom functions, paging and stationconnections. Each KTS extension has a lamp indicator for all available outside lines, showing whether ornot they are busy as well as giving visual indication of an incoming call. A KTS does not require dialing acode to gain access to outgoing lines; PBXs invariably require dialing a number or code (such as “9”). Ahybrid system typically provides the combined features and benefits of both KTS and PBX systems.

Automatic Call Distribution (ACD)

ACD is an important application for businesses handling large volumes of incoming telephone calls, suchas in a call center environment. A PBX system equipped with integrated ACD software enables the switchto automatically route incoming customer calls to groups of call center agents with specific skill sets,agents who have been idle the longest or agents who have handled the fewest number of calls.

In many large call center applications, PBXs interface with stand-alone ACD systems. Leading PBX/ACDvendors also offer integrated ACD solutions for their PBX systems, giving customers a choice betweenimplementing a PBX/ACD solution with better integration capabilities or a third-party solution from astand-alone ACD vendor.

The agent capacities of PBX/ACD systems allow those vendors to compete for large call centerinstallations, traditionally the domain of stand-alone ACD vendors. In addition, the trend toward networkedcall centers has advantages for the PBX/ACD vendors, who see themselves as having deeper skill setsand more expertise in network communications than their stand-alone ACD counterparts.



Today’s client/server-based ACD solutions off-load many of the call-processing tasks that requireapplication-based intelligence to standard PC-server platforms. Utilizing CTI technology and API software,these solutions function as call/connection servers within the enterprise network. Call center businessesof all sizes have substantial investments in modern ACD call center technology. As such, ACD remains amajor revenue stream for PBX vendors.

With the shift toward supporting e-commerce and customer relationship management (CRM) solutionsand applications, along with the increasing emphasis on handling all contact media—voice calls, faxes, e-mails and Web-based inquiries—ACD vendors and the market, in general, now promote use of the term“contact center” instead of “call center.” Attributes of a contact center include:

• Multichannel contacts—the capability to combine the use of two or more media on a single contact.For example, a caller and agent can see the same Web page while speaking with each other on thetelephone or chatting via e-mail.

• Universal queue—the capability to route contacts to an agent regardless of channel. The result is thatagents are not separated by contact type, but can handle a voice call at one time and handle an e-mail or a Web contact at a separate time.

• Contact blending—allows agents to handle both inbound and outbound telephone calls via anycontact channel applicable to multichannel contacts and universal queue.

Voice Processing (Mail) Systems

A voice mail system records, stores and plays voice messages. It supports features that enable end usersto access, forward, reply to, schedule the delivery of and tag/edit messages, among other functions. Endusers, or subscribers, are the owners of personal voice mailboxes in a voice mail system. Subscribers canaccess messages in their voice mailboxes from telephones or PCs by entering account numbers andpersonal passwords. Telephone access allows information entry and all system commands to beperformed via the phone’s touch-tone keypad. In addition, more advanced systems deploy speechrecognition technology to manage messaging via simple voice commands over the phone.

Integrated voice mail systems typically have a message-waiting indicator (MWI), such as a light on atelephone or icon on an alphanumeric display. A ringing telephone can default to a voice mailbox thatdelivers an invitation to leave a message; the system then automatically records the message in memory.

The telephone user interface (TUI) of a voice mail system provides the subscriber with voice menuprompting for message management functions, including retrieval and playback of messages, messagedisposition (deleting, saving, replying to or forwarding messages to others), sending new messages toone or more subscribers and changing the setup of mailbox facilities—greetings, passwords, distributionlists and access to live assistance. In addition, the growth of e-mail usage has increased the popularity ofthe PC screen as a voice mail user interface. Point-and-click techniques, together with labeled icons,make desktop voice message management easier and more efficient.

The popularity of Internet-based e-mail has positioned it as a target for integration/consolidation with voicemail systems and services. These services are particularly well suited for small businesses and SOHOenvironments that do not have an internal enterprise e-mail server. In addition, the Internet offers a cost-effective means of voice mail networking between diverse voice mail systems using the Voice Profile forInternet Messaging (VPIM) industry standard. Popular Web browsers are also being employed by usersfor PC access to voice mail servers over the Internet and can be used by messaging systemadministrators to remotely manage user activities via the Web.



Unified messaging is an extension of voice mail that enables subscribers to access and managemessages such as voice mail, fax mail and e-mail from a single user interface. The goal is to simplify andspeed message handling by improving how subscribers receive, reply to and manage messages,regardless of communications medium.

Implementation can consist of a single unified messaging server or multiple servers behind a single userinterface. The user interface itself is typically a desktop PC driven by an application software module orintegrated software load. The PC-based interface also enables data files to be easily attached andretrieved with any form of message medium, not just e-mail. Several of today’s sophisticated unifiedmessaging packages allow subscribers to embed voice messages in fax and e-mail files, view faxesonscreen, be notified of e-mail (by PC or telephone) and redirect e-mail to a fax machine via telephonecommands. The telephone is another user interface (TUI) alternative for unified messaging. In addition tolistening to voice messages, a text-to-speech option is generally offered for the telephone retrieval of textmessages from e-mail or fax.

Terminal Adapters (TAs)

A terminal adapter, also known as an ISDN modem, has two main tasks. One is to adapt the format of thedata or voice signal at the R interface (the interface between a non-ISDN terminal and the TA) to the 64-Kbps B channel. The other task is to provide a means of setting up and clearing ISDN calls. Likemodems, terminal adapters can be packaged in a variety of ways. Individual basic-rate TAs can beobtained in stand-alone boxes, each with an ISDN port and one or two terminal ports. For central site use,a number of TAs can be mounted in a rack. Finally, a TA on a card can be plugged into a personalcomputer or workstation.

Communications Processors

Communications processors are multifunctional, program-controlled computers—typically called “servers”today—dedicated to communications and serving as control points, or nodes, in networks. In general, aprocessor performs one or more of three major functions:

• Front-end processing

• Intelligent switching

• Concentration

A front-end processor serves as a locally attached peripheral device to one or more larger computers,relieving them of the overhead involved in message handling and network control that is required in acommunications environment. An intelligent switch routes messages among the network’s various endpoints and participates in the network’s control and management, either under the control of a master(usually front end) processor or as a peer of other intelligent switches.

A concentrator controls a community of terminals, clusters of terminals or distributed applicationsprocessors. It gathers, queues and multiplexes their transmissions onto one or more high-speed networktrunks and participates in the network’s control and management, again either under the direction of amaster processor or as a peer of other concentrators and switches. Most high-end communicationsprocessors perform all three of these tasks.

Modems

Virtually all modern telephone networks use digital transmission to connect digital switching offices. At theedges, however, networks still use analog facilities to connect to customer equipment, especiallyresidence and office equipment, such as telephones, fax machines and computers. For the network to



carry data (e-mail, documents, spreadsheets or other data types) and digitized fax images, theinformation must be converted into a continuous (analog) line signal variable along one or more of theparameters of amplitude, frequency and phase.

A device called a modem, a contraction of the terms modulate/demodulate, performs the conversion fromdata format (1s and 0s) to analog format. Modems are always used in matching pairs and generallyconform to ITU-T standards. The unit at the sending end converts information coming from a host, PC orterminal. At the receiving end, another modem converts the analog signals back into data format beforeacceptance by the receiving devices. In the case of a fax, the receiving modem’s output is digitallyprocessed by a DSP in the receiving fax machine to create a nearly identical rendering (that is, facsimile)of the original image.

Modern modems can transmit and receive simultaneously (full duplex) on the public network at speeds upto 56 Kbps. V.34 is a popular international standard for dial-up modems, supporting speeds of 28.8 Kbps.The more recent V.90 standard supports downstream operation at 53.3 Kbps and upstream at up to 33.6Kbps. The even more recent V.92 standard improves on performance even further, increasing upstreamspeeds to a maximum of 44 Kbps, under optimum conditions.

Multiplexers

Multiplexers combine streams of data from many individual low-speed channels and transmit a combinedstream over one high-speed communications link. Multiplexers maximize the efficient use ofcommunications links in a network because users can lease one high-speed line for much less than itwould cost to lease many low-speed lines. Multiplexers generally fall into one of two very broadcategories:

• FDMs

• TDMs

FDMs are the earliest and least sophisticated form of multiplexing equipment. FDMs divide the allocatedbandwidth of a conditioned analog line into independent, permanently assigned lower-speed subchannelsthat operate on a particular frequency within the spectrum.

TDMs are digital devices that accept multiple digital inputs and convert them to one composite digitaloutput. Rather than divide a communications link into separate channels as FDMs do, a TDM dividesbandwidth into time slots while maintaining the integrity of the single channel. Within the TDMclassification, there are several different types of devices:

• Simple-fixed format

• Statistical (STDMs)

• Drop and insert

In a simple fixed-format multiplexer, the relationship between the input and the output is fixed. In a moresophisticated multiplexer, it may be varied.

Statistical time division multiplexers (STDMs) contain an added microprocessor that provides intelligentdata flow control and enhanced functionality, such as error control and sophisticated user diagnostics.The major difference between TDMs and STDMs is that STDMs dynamically allocate time slots on the linkto inputting devices on an as-needed basis, rather than in a fixed-dedicated basis. Therefore, rather thanwasting bandwidth when the inputting device is idle, the bandwidth is utilized to serve active devices (thatis, devices with data ready to go). STDMs work best when data flow is intermittent; if data from multiple



devices occurs simultaneously, one or more devices will have to wait. Unlike TDMs, STDMs have buffersfor holding data from attached devices, which enable them to handle a combined input speed (aggregatespeed) that exceeds the speed of the communications link.

Drop-and-insert (D/I) multiplexers are commonly used in private networks and in the dedicated facilitiesportion of public networks. D/I muxes are used to remove one or more of the channels from a multiplexedtransmission system or to add more channels to vacant slots in a multiplexed system. On a small scale,they perform a function similar to DCSs, described previously. Add/Drop Multiplexers (ADMs) used inATM networks are one example.

Protocol Converters

As mentioned earlier, not all data communications devices speak the same language. Network designerscan achieve flexibility and economic rewards by using products from more than one vendor, andequipment manufacturers have responded by developing products that overcome languageincompatibilities. Protocol converters and emulators can be hardware-based, software-based or acombination of both and can range from a microprocessor-based circuit board to a front-end processorwith the capability to handle conversion functions. Available conversion devices might handle one ormany types of conversions. For example, some devices handle only code or physical interfaceconversions, while others handle protocol conversion, device emulation, and code and interfaceconversions.

A protocol converter actually changes one protocol to another by stripping down the data from one deviceand reformatting it according to the rules of a new set of specifications. During the conversion sequence,the protocol converter accepts blocks of data in one protocol, adds or deletes the necessary controlcharacters, reformats the block and calculates the required check characters so that the receiving devicegets characters formatted according to its requirements.

For example, in an asynchronous-to-SDLC conversion, the converter accepts a string of characters,eliminates start and stop bits, changes the coding scheme from ACSII to EBCDIC, assembles thecharacters into a block, and adds appropriate headers and trailers to create complete frames.

PCs

The major end-user device in the network is the PC, and it has gained nearly universal acceptance as atool to perform local data analysis. It is only natural that a user would want to communicate with a networkhost via such an intuitive device.

PCs and data terminals differ in one major respect: the PC is an intelligent device, capable ofmanipulating and analyzing data and handling a variety of applications, but the data terminal is not. It was,therefore, often referred to as a “dumb” terminal because its basic function was to serve merely as aninterface between a human operator and a host computer. Even “intelligent” terminals—those that canperform some operations on collected data—do not offer the sophisticated capabilities of a PC.

Management and Control

As the voice and data worlds merge, the line between telemanagement and network managementsoftware systems blur. Computer manufacturers are incorporating telemanagement functionality intonetwork management systems. Telemanagement software vendors, on the other hand, are incorporatingnetwork management capabilities, which are generally voice-oriented and designed to provide PBXmanagement interfaces. The PBX market is the focal point for the convergence of the computing andtelecommunications approaches to network management. Most major PBX vendors now have a datanetworking strategy as well as a voice strategy.



Telemanagement Systems and Software

The growing importance of telecommunications networks within companies means that managementtools are essential. Telecommunications management networks are increasingly recognized as strategicassets that can increase customer satisfaction, build customer loyalty and develop new business.Telecommunications management systems not only monitor staff activity and customer response, butalso help the telecommunications manager optimize the use of the network and identify problems as theyarise.

Telecommunications management systems originated in service bureau-based call accounting software,which provided detailed station reports to large firms as a supplement to telephone company bills. Anatural evolution of service bureau software was the development of licensed software systems for aclient’s computer system. Changes in computer and software technology added significant options tocomputing alternatives during the mid-1980s, when PCs became increasingly popular. During the late1980s, LANs became even more popular. Today, software developers offer telemanagement solutions formany of these platforms.

Generally speaking, telemanagement system applications are organized into several categories:

• Call accounting and management

• Cost allocation and management

• Asset management

• Process management

Asset management facilitates the management of physical assets, such as equipment inventories andcable and wire resources. Asset management also may be used to manage logical assets, such assoftware. Process management automates a number of processes (for example, traffic analysis, networkdesign and optimization, directory management, and work order and trouble ticket management), whichenable the effective management of a telecommunications network.

Communications Software

An efficient method of controlling data communications networks utilizes combinations of hardware andsoftware for control purposes. Repetitive tasks that rarely change are best implemented as hardwaremodules, while dynamic tasks (such as the maintenance of terminal specifications) are best implementedthrough easily changed configuration software that can be altered without disrupting network operations.

Data communications software—often transparent to the user, particularly in large-scale data networks—can be implemented in several layers, requiring a support staff of specialized programmers for itsmaintenance and design. Communications software resides in PCs, servers, terminal controllers, front-end systems and mainframes. It is almost always required for establishing some phase of long-distancecomputer operations.

Communications software is used for the allocation, control and management of the following:

• Data communications links, which are the actual facilities used for data transmission.

• Central-site resource requirements, including mass storage, memory and CPU time.

• Terminal networks and remote computing resources.

• The relationship between local and remote applications software and their databases.



Communications software focuses on a number of objectives: communications between various devices,communications between end users (or between end users and applications), communications betweendatabases, communications between applications, and management and control of communicationsactivities. While hardware generally supports line, device and presentation functions, software supportsthe following:

• Executive functions that control the sequencing of tasks.

• Directive functions that make decisions about how, when, which and how much of the systemresources should be committed to a given task.

• Quality-assurance functions that monitor what is happening in the network and provide fallbackresources in the event of system failure.

• Intersystem functions that act as interfaces between the various layers of data communicationssoftware and hardware.

Recent development efforts have advanced communications software toward the interaction of end-userprograms on a peer-to-peer basis. This bilateral interaction makes the diverse connection schemes andhost servers typically found in large corporations completely transparent to end users. It alsodemonstrates the shift away from rigid, hierarchical network control toward distributed control.

Technology Analysis

Business Use

Network Management

In a complex multivendor environment, end users must assume more responsibility for the network’songoing functionality. They must seek out appropriate solutions for network redundancy, fortroubleshooting and verification, and for network management. Choosing the appropriate equipment fornetwork testing, monitoring and control allows users to carry out that responsibility. As the complexity ofthe network increases, however, individual test devices can prove inadequate. For such environments,the network management system is essential.

In the past, network management was not really management, it was crisis intervention. Nowadays,network managers are more responsible for the control and monitoring of their own systems. Networkmanagement systems have developed from many sources: the desire of interconnect vendors for a value-added selling point, the proliferation of easy-to-use management software and the users’ needs to gettheir communications under their own control.

Figure 16: Network Management Functions

From a technology standpoint, network management can be depicted as the intersection of seven differentfunctional areas.



A minimal network management system consists of:

• A CPU

• A hard disk or diskette storage device

• An operator console

• A set of local and remote monitoring devices

All network management systems include mechanisms for monitoring network components. When thenetwork management system vendor also manufactures modems, the vendor usually designs themonitoring device as a built-in modem feature, eliminating the need for the user to acquire separatemonitoring devices. On other systems, stand-alone monitoring devices must be attached to modems ormultiplexers at each remote site.

In most network management systems, monitoring devices examine only the status of the modem ormultiplexer, its interface with the equipment, its interface with the transmission facility and the condition ofthe transmission facility. Information on the modem or multiplexer and its interfaces comes from thepresence or absence of signals on various interface leads. Information on the transmission facility comesfrom the measurement of various parameters, such as signal level, noise, distortion, phase jitter and linehits. If a given interface signal or analog characteristic falls out of specification, the system’s monitors willset off an alarm to notify the operator of a failure.

Some network management systems can switch automatically from a failing component to a “hot” standbyunit either on receipt of an alarm or on command from the operator. Some systems can also bypass afailed communications line by a call placed automatically over the switched-voice network. Suchautomatic dial backup procedures require two switched-network calls for full-duplex operation.

As networks become increasingly sophisticated, network management systems grow in complexity. Oneof the more interesting recent developments is the incorporation of expert systems techniques intonetwork management. An expert system—software that contains rules for making logical inferences—canadd a degree of intelligence to a network management system. Such an intelligent system can do morethan isolate faults; it can also suggest to an operator the possible causes of those faults, test hypothesesabout them and propose courses of action to remedy them.

Multiple network management systems can coexist within a corporate network. Communications networkstoday may be built from several different types of dissimilar equipment and may incorporate severalsmaller networks of different types. Full network management may require interconnecting managementinformation from all of these networks into a usable form. One of the frontiers of network management isthe construction of systems that can bring together this dissimilar information and integrate it throughdecision support or executive support systems.



Technology Leaders

Listed below are some of the leading vendors in both the data and voice communications industries forequipment and services:

• 3Com

• BT

• Alcatel

• AT&T

• Avaya

• Cisco Systems

• Ericsson

• Hewlett-Packard

• IBM

• Lucent Technologies

• MCI WorldCom

• Motorola

• NEC

• Nokia

• Nortel Networks

• Siemens

• Sprint

• Verizon

Insight

Movement continues today toward the integration of enterprise voice and data communications networks,spurred-on largely by the interest in voice over data networks, with the emphasis clearly on VoIP.Justifications for converging onto a single infrastructure have progressed beyond toll bypass applicationsto projections of lower total cost of ownership (TCO), more efficient system management andadministration, and enhanced applications use. But QOS issues and lack of consensus on standardscontinue to be inhibitors, along with a perceived shortage of personnel with expertise in both areas. Inaddition, with shaky economic conditions still existing globally, enterprises are more carefully scrutinizingany modifications to current infrastructures.

Ray Horak, president of The Context Corp., Mt. Vernon, Washington, developed this report exclusively forGartner. Context is an independent consultancy that works closely with manufacturers, developers,distributors, carriers and end users across a wide range of technologies and applications—at both thestrategic and tactical levels.



Mr. Horak is an internationally recognized author, technical writer, seminar leader and lecturer. Hehas written the best-selling Communications Systems and Networks, published by John Wiley and Sonsand currently in its third edition, and is senior contributing editor for Newton’s Telecom Dictionary. He is amember of the faculties of Network World Technical Seminars and Terrapinn and is a regular speaker atleading industry conferences. Mr. Horak serves on the editorial boards of a number of periodical industrypublications and on the advisory boards of several colleges and universities.

The report draws heavily on Communications Systems and Networks, authored by Ray Horak andpublished by John Wiley and Sons.

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