Basic Concepts of Communications-An Introduction

DPRO-89990 Technology Overview
Richard Costello 24 October 2002
Basic Concepts of Communications: An Introduction
Summary
Between the existing voice and data communications technologies and the emerging communications
technologies, there is a broad range of basic awareness required by the knowledgeable end user.
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
Table 2: ADSL Service
List Of Figures
Figure 1: The OSI Reference Model
Figure 2: High-Level Data Link Control (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
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Figure 8: Pulse Code Modulation (PCM)

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: Depiction of ATM and Synchronous Transfer Mode (STM) Cells
Figure 14: Cabling Types
Figure 15: LAN Topologies
Figure 16: Bridges, Routers and Gateways
Figure 17: Data and Voice Network Convergence
Figure 18: Network Management Functions
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Copyright 2002 24 October 2002 2
Technology Basics
Telecommunications systems consist of two main elements—transmission and switching—and may be

analog or digital. A telecom network consists of a number of switching nodes joined by transmission links.
The switching mechanism enables transmission facilities to be shared, thus eliminating the need for every
user to have a dedicated line to every other user. A purely analog telecom system can convey a
continuous range of signals through the network, while a digital system quantifies the information before
sending it. A technique called pulse-code modulation (PCM) is used to convey speech digitally.
Communications suggests a system of routes or paths through which information travels from party to
party. In a voice communications network, these paths can connect local devices such as private branch
exchanges (PBXs) and terminals in one building or geographically dispersed equipment.
Data communications involves the exchange of data over short-haul connections—local-area distances
up to several miles—and long-haul connections over virtually unlimited distances. The term “data” refers
to a broad range of information, including documents (text and drawings) and spreadsheets, e-mail,
customer records, daily tallies, and other information that is digitally coded and intelligible to a variety of
machines—such as servers, PCs, mainframes, terminals and other machines. With converged networks,
data takes on the additional meaning of voice, video and facsimile—in short, anything that can be
converted to bits and bytes and packaged according to agreed standards (protocols).
Prerequisites for Communications
Underlying the development of any network are certain standards and prerequisites for communications.
Specific conditions must be met before communications can occur. This principle holds true in any
information exchange: two people attempting to exchange ideas must speak the same language for
communications to take place. Otherwise, although words are spoken by one and heard by another, no
communication occurs.
In data communications networks, devices must also speak the same language and follow the same
rules, and mechanisms must be in place to ensure that data 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, causing real confusion among data communications users.
However, various organizations including the International Telecommunication Union-Telecommunication
(ITU-T), the International Standards Organization (ISO) and the Electronics Industries Association (EIA)
have interface standards that are widely recognized and used throughout the industry.
Open Systems Interconnection
To communicate, devices must be compatible on various levels. The ISO’s Open Systems
Interconnection (OSI) reference model for data communications consists of a seven-layer hierarchy that
defines physical interface characteristics, as well as protocol details at respective levels so that
applications can exchange data reliably. The OSI model does not describe a specification for any
particular communications system, but serves as a reference point for the establishment of a standard
data communications system.
Each layer of the OSI model defines a particular function that involves not only the transfer of data from
one machine to another, but also the integrity of the information transmitted. If received, data is garbled
and unintelligible; it is useless and must be retransmitted. The receiving device must be able to let the
sending device know whether a transmission has been effectively completed. These aspects of data
exchange, along with others, are defined in the OSI model, which is structured in an upwardly compatible
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manner. If there is compatibility on one level, it is assumed that compatibility exists on the levels beneath.
These levels are briefly defined in 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 involve
compatibility on only the first two layers of the model. In complex networks, however, hundreds of
conditions must be met before communications can occur, and compatibility must be established on all
seven levels. Since this report deals primarily with basic data communications, the prerequisites for a
simple data exchange include compatibility on the interface, data transfer code, protocol and
synchronization levels.
Interfaces
An interface is the physical connection between two communications devices, comparable 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 assigned signal functions such
as carriage return, line feed and request to send (information). The Electronic Industries Association (EIA)
RS-232-C is the industry-standard interface for connecting data terminal equipment (DTE) and data
communications equipment (DCE).
Each pin in a 25-pin connector represents a standard specification. Pin assignments are explicit and
unalterable, except for those that are unassigned. Unassigned pins can be used to handle special
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functions 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-C and
RS-449. It specifies standards for expanded transmission speeds, longer cable lengths and additional
functions.
Universal Serial Bus (USB) is a 12-Mbps serial bus developed by several leading PC and
communications vendors to facilitate a “plug and play” capability for devices (printers, modems, mice,
telephones, joysticks and other devices.) 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 are
connected to its ports. USB also includes a “hot attach” capability, which enables devices to be
connected, detected and configured (and disconnected) even when the power is on.
Codes
As previously noted, information travels between machines in patterns of 1s and 0s:

• Each 1 or 0 is called a bit.
• Eight bits make up a byte (or character or octet) of data, which is handled as a logical unit.
A communications code encompasses a standard message format and a set of rules for use and
placement of control characters in a message. Codes consist of separate bit patterns for each text,
graphics or control character used in an information exchange. For example, the widely used American
Standard Code for Information Interchange (ASCII) is a seven-bit code. Seven bits of 1s and 0s represent
each character in the code set; an eighth bit is used for error checking. The ASCII code consists of 128
characters—95 graphics characters and 33 control characters.
Other commonly used codes are Baudot and Extended Binary-Coded Decimal Interchange Code
(EBCDIC), consisting of 256 characters. Baudot, named after Emil Baudot and first established in 1874, is
a five-bit code used on vintage teleprinter terminals such as those made by Telex. It does not, however,
have an error-checking capability or means of checking that the information is received. EBCDIC, the
code used on synchronous IBM equipment, consists of eight-bit coding (256 characters). An end-user
device that operates with one type of code (for example, ASCII) cannot accept data from a device using a
different code (for example, EBCDIC), unless a conversion is performed to make one code compatible
with the other. Devices called code converters handle this function.
Unix- and DOS-based operating systems, except for Microsoft Windows NT, use ASCII for text files.
Windows NT (now also called Windows 2000) uses a newer code, Unicode, consisting of 65,536 unique
character definitions (16 bits or two bytes per character). IBM’s System 390 servers use EBCDIC.
Conversion programs allow different operating systems to change a file from one code to another.
Protocols
Communications protocols cover a wide spectrum and range from single character-by-character
transmissions with no error checking to complex rules for moving large amounts of data among many
devices. 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, and
initiate corrective action
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Prior to being sent, the payload is chunked and successively packaged together with overhead
information. Through hardware or software, the sending device automatically formats the data and adds
the required overhead bits before transmitting each block. The receiving device automatically checks the
overhead bits before signaling that data has been received. If necessary, due to errors picked up along
the way and detected at the receiving end, the receiving end can request retransmission of one or more
blocks.
High-Level Data Link Control (HDLC) is one of the most commonly used protocols in what is Level 2 of
the OSI. Level 1 is the detailed physical level that involves actually generating and receiving electrical
signals. Level 3 is the higher level that has knowledge about the network, including access to router
tables that indicate where to forward or send data. On sending, programming in Level 3 creates a frame
that usually contains source and destination network addresses. HDLC (Level 2) encapsulates the Level 3
frame, adding data link control information to a new, larger frame. Variations of HDLC are used in X.25
(protocol) public networks and frame relay (protocol) public and private networks.
Figure 2: High-Level Data Link Control (HDLC) Framing
A frame of data is created by HDLC. The beginning eight-bit flag alerts the receiving device that a frame of data is
about to be received. The end flag signifies that the transmission is complete. The address field names the station to
which the information is being sent (or in some cases the sending station), and control information provides counts
of frames transmitted and received. The frame check sequence provides error control information.
Beginning and end flags are actually the same 8-bit sequence (1000 0001). In a typical sequence of
frames, there is only one flag between frames, which serves to end one frame and begin the next. If one
or more bytes of data anywhere in the frame look like a flag, the receiver will get out of sequence and will
be unable to rebuild the payload. To avoid this, the sending end looks at every byte being coded and
converts it to a defined “escape” sequence if it detects a flag character in the payload. At the receiver, flag
characters perform their task and are stripped away; the receiver then restores any detected escape
sequences to the original coding.
Communications protocols are either bit-oriented or byte-oriented:
• Byte-oriented protocols transmit data in eight-bit blocks, and they require an acknowledgment after
each transmitted block before the next block can be sent. Bisynchronous (BISYNC) is a byte-oriented
protocol that defines specific characters for specific functions.
• Bit-oriented protocols transmit data in blocks of any length up to a specified maximum; an
acknowledgment can take place after one or several blocks have been sent, depending on the
protocol. Data (payload) is normally sent in blocks that range from 80 to 512 characters.
Synchronous Data Link Control (SDLC) is a bit-oriented protocol from IBM that has a unique pattern
of bits for specific functions.
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Since no single protocol standard exists, terminals performing the same functions under different
protocols cannot be used on the same system. Devices called protocol converters/emulators overcome
these incompatibilities, allowing terminals from different manufacturers to operate on the same network.
However, an important data communications protocol that has become a de facto standard protocol is
IBM’s Binary Synchronous Communications (BSC), which processes data in blocks of up to 512
characters.
Data Transfer
There are two types of data transfer—parallel and serial. Parallel data transfer employs a communications
interface with a sequence of dedicated wires, each serving one purpose. A typical parallel
communications cable has a number of control wires and eight data wires that carry the signals for
alphanumeric characters. Because there are eight wires, an entire eight-bit character is transmitted in one
strobe, in parallel. This form of transmission is, however, limited by distance. If delays occur, the data
from the eight separate channels will not arrive together and cannot be matched exactly. This
phenomenon is referred to as “skew.”
Most data communications devices transmit in a serial fashion. Used for long-haul communications, serial
data transfer operates bit-by-bit rather than in parallel. The datastream is therefore subject to certain
prerequisites:
• Data characters (parallel codes) must be converted to a serial bitstream in order to travel on the
communications line.
• The serial bitstream must be broken up into individual characters from five to eight bits long.
Figure 3: Types of Data Transmission
In a parallel transmission, each bit in a character is transmitted simultaneously on a separate line. In a serial
transmission, bits are transmitted in sequence over one line.
Synchronization
Data transmission can be asynchronous or synchronous. Asynchronous transmission, often called start-
stop transmission, transmits or receives one character (seven or eight bits) at a time. Each character has
its own “timing” device (that is, a start bit and a stop bit), letting the receiver know where a character
begins and ends. The time between the transmission of each character is referred to as idle time.
Asynchronous transmission is inefficient, due to the overhead required for start and stop bits and the idle
time between transmissions. With asynchronous communications, transmission speeds as high as 33.6
Kbps can be supported in most equipment today.
Figure 4: Start and Stop Bits
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During transmission, a data character consisting of seven or eight bits is preceded by a start bit and ended with a
stop bit that lets the receiving device know where a character begins and ends.
Synchronous transmission sends a block of characters over a communications link in a continuous

bitstream. Data transfer is controlled by a timing device called a clock. Initiated at the sending device (for
example, a terminal, modem, multiplexer or front-end processor), the clock runs at a frequency equal to
the transmission rate. Each block is preceded by sync bits or a unique character pattern to allow
synchronization, and special idle characters are transmitted if no data is being sent. In the case of HDLC,
flag characters are transmitted in the absence of data.
To accommodate the transmission of a large number of blocks, terminals involved in synchronous data
transfer must have a buffer for storing the character blocks, as they wait to be transmitted or processed at
the receiving end. Synchronous transmission is generally used for higher-speed data transfers. In many
cases, a device can operate both asynchronously and synchronously, and users can select the
appropriate operation by activating a switch. Equipment that transmits in both ways can be used for both
interactive (two-way) and batch (one-way) applications.
Simplex, Half-Duplex and Full-Duplex
Three modes of data transmission are available:

• Simplex transmission occurs in one direction only.
• Half-duplex transmission occurs in both directions, but only in one direction at a time.
• Full-duplex transmission occurs in both directions simultaneously.
Half-duplex operation is possible on both two-wire and four-wire circuits, but on a two-wire line the user
must deal with turnaround time. Turnaround refers to the halt in transmission that occurs when travel is
reversed from one direction to another on the line. In full-duplex transmission there is no turnaround
time—data travels in both directions on the circuit simultaneously. Full-duplex operation generally requires
a four-wire line, although some sophisticated modems can handle this mode on a two-wire circuit.
Figure 5: Transmission Methods
Simplex transmission goes in one direction only. Half-duplex transmission goes in two directions, but only one at a
time. Full-duplex transmission can go in both directions at the same time.
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Transmission Signal Attributes
For the purpose of telecommunications, the signal that is transmitted has three fundamental attributes—
frequency, amplitude and phase. Frequency is the number of cycles in a given time, usually per second
(measured in hertz [Hz]). Amplitude is the height of the peaks of the waveform and, in the case of sound,
determines the volume or “loudness.” 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 is of
particular importance in modulation theory and is the basis for several commonly used modulation
techniques discussed later in this report.
Figure 6: Transmission Signal Attributes
For the purpose of telecommunications, a transmitted signal has three fundamental attributes—frequency, amplitude
and phase.
Bandwidth
The bandwidth of a communications channel determines a channel’s information-carrying capacity.

Bandwidth is defined as “the range of frequencies that the channel is capable of transmitting without
interference or signal loss” and is measured in hertz. The greater the range of frequencies a medium can
handle, the greater its information-carrying capacity. In data communications, bandwidth is generally
specified in bits per second (bps). A channel that supports a 2-Mbps bandwidth can support a
transmission rate of two million bits per second.
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Data Transmission Rates
The rate of transmission can be expressed as the modulation rate, the data signaling rate and the data
transfer rate:
• The modulation rate is expressed in baud units and is used to describe the rate at which changes in
the condition of the circuit can be made in a given time.
• The data signaling rate is used to express the rate at which data is transmitted and is expressed in
bits per second (bps).
• The data transfer rate describes the rate at which data arrives after transmission and is usually
expressed in terms of bits per second (bps).
Transmission Quality
The quality of the transmission can be thought of as the conveying of information without the loss of
information or the addition of any unwanted information (noise). The quality of an analog communication
system is usually measured in terms of the signal-to-noise ratio, which is usually expressed
logarithmically in terms of decibels (dB) and bandwidth. The equivalent measures of the quality of a digital
system are the bit error rate and the bit rate.
Multiplexing Techniques
Frequency Division Multiplexing (FDM)
Multiplexing allows communications media to be shared between users. The two main types of
multiplexing—time division multiplexing (TDM) and frequency division multiplexing (FDM)—both aim to
maximize the number of message signals that can be transmitted over a transmission link.
FDM was the earliest and least sophisticated method of multiplexing. It divides the allocated bandwidth of
a conditioned analog line into independent, permanently assigned, lower-speed subchannels that operate
on particular frequencies within the spectrum. The speed (bits per second) at which the channel operates
depends on the amount of bandwidth (Hz) assigned to each channel; the required bandwidth increases or
decreases in proportion to the operating speed. Therefore, the slower the transmission rates, the more
subchannels can be assigned within the bandwidth; the faster the rates, the fewer subchannels can be
assigned.
FDM was the basis for early voice networks but has been replaced in large measure by time-division
techniques. FDM continues to find important applications though, such as in the deployment of Digital
Subscriber Line (DSL) technology and in optical fiber media, where the technique is known as wave
division multiplexing (WDM).
Figure 7: Multiplexing Techniques
Multiplexing maximizes the efficient use of communications links, allowing a business to lease a single high-speed
line for much less than it costs to lease many low-speed lines.
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Time Division Multiplexing (TDM)
Rather than divide a communications link into frequency-separated channels as FDM does, TDM divides
time into slices called time slots. With TDM, each inputting terminal takes its turn at transmitting and
receiving data in a continuous fashion; the order in which the multiplexer serves the terminals is fixed.
Depending on the multiplexer type, the device accepts only one bit, byte or packet of data from each input
line; puts it into a specifically allocated time slot on the high-speed transmission line; and then moves on
to the next terminal in the sequence. If the inputting device has no data to send, the TDM fills out the
assigned slot with some type of information. The process of accepting data from many 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 and enables
the receiver to work out which bits belong to which channel. Input signals are sampled one after the 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) combined with TDM is still the most widely used method of transmitting
analog signals over digital transmission links. PCM is the process of converting information into the digital
information required for transmission. There are three steps to PCM:
• Sampling
• Quantizing
• Encoding
Figure 8: Pulse Code Modulation (PCM)
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The analog waveform is sampled every T seconds. The samples are then converted to integer numbers using eight
bits per sample. Finally, the integer numbers representing the samples of a given channel are sent in sequence as a
binary 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 the
information. The sampling rate needs to be at least twice the highest frequency of the signal in order to
extract the information, known as the Nyquist rate. Nyquist’s theorem is sometimes referred to as the
sampling theorem and has an important application to the digitization of speech using PCM.
If we take the bandwidth of speech to be 3kHz, then the signal must be sampled at least 6,000 times per
second (2 × 3,000Hz) to preserve the original information for transmission in digital mode. Practical
implementations use 8,000 samples per second (8kHz), which yield a sample every 125 microseconds
(the interval between samples T is 1/8,000 second = 0.125 milliseconds = 125 microseconds). At the
receiving end, the discrete values are converted to speech signals. This forms the basis for ITU-T PCM.
Each sample is adjusted to the nearest of ±128 signal (voltage) levels, a process known as quantization.
The signal is then encoded by converting each analog sample into a code word consisting of eight binary
digits (a sign bit and seven speech bits). In the case of speech, this generates an eight-bit sample 8,000
times a second, which produces a 64-Kbps signal. It turns out that 125 microseconds is a very long time
to wait for samples, long enough in fact that many speech signals can be sampled and coded in the space
of 125 microseconds.
Digital Facilities (T1 and E1)
Repeating the process in turn for 24 speech signals (24 time slots) yields the popular T1 format of 1.544
Mbps (1,544,000 bits per second). This consists of the aggregate rate due to speech (24 speech channels
× 64 Kbps per channel = 1,536,000 bps) plus a flag bit inserted every cycle of the 8,000Hz clock. The flag
bit is sometimes “1” and sometimes “0” to create a unique pattern for the receiver to lock on (synchronize)
for proper channel identification. A collection of 12 frames is called a superframe (SF), corresponding to
D4 format. A collection of 24 frames is used in the extended superframe (ESF) format. The T1
transmission medium degrades frames the farther they travel down the line to unacceptable levels of
signal weakness and distortion. Devices called repeaters are inserted every 5,000 to 6,000 feet to detect
and regenerate the 1s and 0s.
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In Europe, the sample rate of 8,000Hz is applied to 32 channels to yield a 2.048-Mbps multiplexed
datastream (32 channels × 64 Kbps) known as E1. Two of the channels are used for synchronization and
signaling, respectively, leaving 30 speech channels. The European technique is more efficient (yielding 30
channels rather than 24) but was not usable in the U.S. due to constraints imposed by the legacy (that is,
analog) public network. As in the U.S., repeaters are used in Europe to prevent the buildup of attenuation
and distortion.
Digital facilities—using a combination of TDM and PCM techniques (described above) and carried by fiber
optic cables—dominate long-distance networks. They offer high-transmission speeds, but do not require
modems to convert digital data into analog form before transmission. Optical facilities are less prone to
error and can transmit data for long distances without regeneration. The public telephone networks
throughout the world now are primarily composed of digital facilities supporting digital devices used for a
range of digital communications—voice, data, video and other networks.
Voice Compression
Adaptive differential PCM (ADPCM) is a form of waveform encoding—one of the main techniques used in
speech compression. With ADPCM, only the difference between one sample and the previous one is
coded using four different values; thus for each eight-bit PCM sample, a four-bit ADPCM sample is
generated, resulting in a 32-Kbps constant bitstream. ADPCM can also be used in video compression,
where, for example, it can usefully code high-contrast edges. The use of ADPCM enables 48 speech
channels, rather than 24, to be carried on the 1.544-Mbps T1 facility.
Other compression standards exist; for example, the ITU-T G.728 standard defines 16-Kbps audio. Even
lower bit rates can be found in the wireless world, where radio channel bandwidth is precious and must be
conserved 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, the sophisticated coders used in cell phones and VoIP gateways do a remarkably good job.
Digital Signal Processors (DSPs)
Special-purpose microprocessors with instruction sets designed to manipulate digital signals—termed

digital signal processors (DSPs)—are now widely used. DSP technology was originally developed to
provide encryption for military applications. Today, a wide range of applications depends on DSPs for
transmission, including modem signal processing, video compression, TV enhancement and
straightforward speech compression.
Transmission, Switching and Internet Telephony
Transmission facilities are the paths that connect local and geographically dispersed equipment. Most
frequently, they are telephone cables made of copper wire, but increasingly fiber-optic cables, orbiting
satellites or microwave-radio beams also serve as transmission facilities. The major types of long-
distance communications facilities are circuit-switched (dial-up), packet-switched and nonswitched
(leased).
Public Switched Telephone Network
The public switched telephone network (PSTN) comprises analog and digital facilities that go through a
central switch. From the residential/home-office point of view, analog lines connect devices such as
telephones, computers and modems to the central office.
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For business customers, telephone companies connect to key telephone systems, PBXs and other
customer premises equipment using analog or digital facilities (or both) as the case may be. Digital
facilities are used exclusively between central offices. See 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 entire
interexchange carrier (IXC) networks. End users pre-subscribe to exactly one IXC, although it can be bypassed in
favor of a different IXC on a call-by-call basis (“dial around”). Tie trunks (6) bypass the switched network by
connecting PBXs in different locations over private facilities.
A PBX enables people within the enterprise to talk to each other (phone to phone), to gain access to the
public network (outgoing calls), and to receive calls from the public network (incoming calls). The latter
two are accomplished through the previously mentioned analog or digital facilities. Is the facility that
connects a central office to the PBX a line or a trunk? If we accept that a line is a dedicated resource and
a trunk is a shared resource, then the answer is that it is both a line and a trunk. From the central office’s
point of view, it is a line because it is dedicated to a single customer. From the end user’s point of view, it
is a trunk because many users (employees) share it, in the sense that they take turns using it, usually by
dialing “9” or some other access code. Thus, the phone (that is, station) side of the PBX is referred to as
the line side; the central office side of the PBX is referred to as the trunk side. Figure “Telephone Network
Architecture” provides a pictorial of this terminology.
Analogously, the PBX side of the central office is known as the line side; the interoffice side of the central
office is known as the trunk side. To round out the story, the switched network has another type of switch
known as a tandem switch (in Europe, a transit switch). Its purpose is to switch trunks to other trunks. A
typical long-distance call thus traverses the customer’s serving (that is, originating) central office, one or
more tandem switches and the destination central office. Add to this the local lines at the originating and
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terminating central offices, interoffice trunks between central offices and perhaps PBXs at respective
ends, and a “switch train” is created, whose ease of use belies its complexity.
The two basic technologies comprising switched networks today are circuit switching and packet
switching. IP telephony derives its attraction and efficiency from packet switching technology.
Circuit Switching
Circuit-switched networks establish a circuit between the calling and called parties for the duration of the
call. During this time, the full traffic capacity of the circuit is available to the connected parties only.
Individual transmission paths (time slots) in the switch train remain dedicated to the call in progress, even
during silent periods (pauses between words and sentences, thinking time, staying quiet while the other
person talks or other lapses in talking.). Time slots up and down the switch train remain assigned and are
released only when the parties disconnect.
Switching enables transmission facilities to be shared, eliminating the need for every user to have a
dedicated line to every other user. The design of networks takes into account the volume of traffic
generated by the users of the network and is usually designed to give a specified grade of service (GOS).
GOS is a measure of probability of a call not being able to be set up because one or more links of the
network is already fully used.
Traffic is measured in Erlangs—the average number of calls in progress at any one time on a particular
set of paths (that is, trunks) connecting a particular pair of switches or average calls in progress through a
switch. Two factors determine the number of trunks: traffic (stated in Erlangs) and GOS (stated as the
fraction of blocked calls out of all calls offered to the set of trunks).
Packet Switching
Packet switching is a networking technology used in public and private data networks. Packet-switched
networks can support a variety of special or customized services and functions, which when combined,
form value-added networks (VANs). A popular way of linking host computers to packet-switched networks
has typically been via an X.25 line. X.25 is an older ITU-T protocol for interfacing DTE and DCE operating
in a packet-switched mode in a private data network. Frame Relay is a higher-speed, packet-switching
technique utilized now as a popular form of data transport. But IP is currently the primary protocol for
transmitting packetized data among network systems.
On a packet-switched network, data travels in groups of characters with control information appended to
the beginning and end of each group. The character group and control information are collectively called a
packet, comparable to an “envelope” of data that is not “opened” until it reaches its destination. Packets
are dynamically routed through a network; that is, if a line fails or is overloaded, the system automatically
reroutes packets over the most efficient path. This allows multiple packets of one message to be sent via
different routes. Therefore, packets require sequence numbering so that they can be rebuilt
(reassembled) at their destination in the correct order.
Nonswitched (Leased) Facilities
A nonswitched facility is an analog or digital line that does not go through a central switch and can,
therefore, be rented or leased exclusively to one customer for voice and data communications. One
example is a facility known as a PBX tie trunk (see Figure “Telephone Network Architecture” above),
which connects PBXs on nonswitched, leased circuits. Companies with high levels of traffic prefer this
arrangement, as it is more economical than the pay-per-call structure of the public network. Users dial a
special code, for example “8” to access a tie trunk (more commonly known as a tie line).
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Although a leased line does not go through a central switch, it nonetheless does go through a central
office—namely, the central office’s distribution equipment consisting of wire and fiber optic distributing
frames, digital cross-connect equipment, and multiplexing equipment such as channel banks and
add/drop multiplexers.
Digital cross-connect systems (also known as DCS equipment) should not be confused with the central
switch. Whereas the latter switches are lines to lines (intraoffice calls), lines to trunks (outgoing interoffice
calls) and trunks to lines (incoming interoffice calls), a DCS is much like an electronic patch panel that
allows individual digital voice channels, digital data channels or entire digital facilities (for example, T1) to
be connected for purposes of private (nonswitched) networking.
Internet Telephony
Internet telephony refers to communications services—voice, facsimile and voice-messaging

applications—that are transported on the Internet, a packet-based network. In general, use of the Internet
in this manner bypasses significant portions, but not all, of the public switched telephone network (PSTN).
See 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 internal
network (intranet) carrying voice calls and data across a wide-area IP network, an example of voice/data
convergence.
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Phone to Phone on the Internet
The five steps involved in an Internet telephony call, performed by the originating network and reversed at
the destination, are listed in Table “Steps in an Internet Telephony Call.” Also listed are the entities at the
respective ends of the transmission facilities as the call progresses. For simplicity, calling and called
parties are assumed to be using analog telephones.
Table 1: Steps in an Internet Telephony Call
Description Entity at One End Entity at Other End
[1] Call origination in the normal manner from an Calling party PSTN originating end
analog telephone office, which routes the call
to the originating Internet
Service Provider (ISP)
[2] Conversion of analog voice signal to digital PSTN originating end office —
(PCM) format
[3] Local transmission of PCM voice signal on PSTN originating end office Originating ISP’s gateway
interoffice trunk (T1 or E1) equipment
[4] Compression/translation of PCM voice signal into Originating ISP’s gateway —
Internet Protocol (IP) packets equipment
[5] Transmission of IP voice signal on the Internet Originating ISP’s IP packet- Internet backbone
switching infrastructure
(packet switch, router)
[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 Destination ISP’s gateway —
digital (PCM) format equipment
[3] Local transmission of PCM voice signal on Destination ISP’s gateway PSTN terminating end
interoffice trunk (T1 or E1) equipment 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 PSTN terminating end Called party
telephone office
PBX to PBX on an Intranet
In a traditional PBX environment (lower half of Figure “Voice on IP Networks”), employees place IP
telephony calls using the preceding steps (in Table “Steps in an Internet Telephony Call”), but are
preceded by an access code (usually “9”) to reach the end office. Alternatively, at locations where IP-
enabled PBXs are in place, the PBXs perform gateway functions so they can connect to legacy end-office
switches. This enables the business to receive calls from the PSTN using traditional facilities (analog
ground-start or loop-start trunks, or T1 digital trunks) and to carry voice call originations (and PC-
originated data) on the company’s local-/wide-area intranet. This avoids the costs of ISPs and tie trunks
and the per-call charges incurred on the public network. In case of a problem on the intranet, callers can
dial a code (“9”) to access the PSTN.
Terminology
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Terms such as Internet telephony, IP telephony and voice over IP (VoIP) are often used, or misused,
interchangeably to denote similar types of calls and services. But Internet telephony refers to calls that are
transported specifically over the Internet, while IP telephony typically refers to calls that are carried on a
private network running the IP protocol. And VoIP is more properly a technology banner, implying defined
(open) implementation standards. By convention, the public IP network (the Internet) is capitalized; private
IP networks (intranets) are not.
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 and sent into a network designed to carry data
packets—not digitized speech packets—and switched onto different paths to or toward the destination.
Time slots are not fixed for the duration of the call; indeed, they may be switched as often as every packet
in some cases. If a time slot is not available at the instant the packet is ready for forwarding, there will be
a gap of some duration that will impair listening.
Gaps are common on packet-based telephony networks, especially when the Internet is used as the
transport medium. If gaps persist during a call, users tend to hang up in hope of getting a better “circuit”
on retry. Pure data packets on the other hand (not digitized speech packets) can experience long delays
without being impaired. The serving packet switch simply waits until it has a logical chunk and sends it to
the user’s computer at will. Data is delayed but not impaired and eventually shows itself as a completed
screen display, file transfer, e-mail download or other data chunk.
Satellite Communications
Satellites provide communications by receiving radio signals at one earth station and transmitting them
back at a different frequency to another earth station. Communications satellites usually operate in
geostationary orbits above the earth: an equatorial orbit within a period of 24 hours such that the satellite
is always in the same position relative to a point on the earth’s surface.
The satellite itself serves as an active relay, and its communications relay systems consisting of
transponders and antennas are its most important components. These systems amplify the end-user
transmissions received from sending earth stations before retransmitting them to receiving earth stations.
The signal at the transmitter is much stronger than the signal at the receiver. Since simultaneous signals
at the same frequency interfere with one another, the satellite converts the signal it receives to another
frequency before transmission.
Very Small Aperture Terminals (VSATs)
Very Small Aperture Terminals (VSATs), also known as micro earth stations or personal earth stations,
allow for the reliable transmission of data via satellite using small diameter antennas of typically 0.9 to 1.8
meters. With their great reliability, versatility and flexibility, VSAT technology offers a cost-effective
alternative to other communication options.
Some VSAT networks have a single host computer, but there is no restriction on the number of hosts that
can be connected to the hub. Thus, a number of applications and closed user groups can be served by a
single hub earth station. For companies with 10 to 200 remote sites, a shared-hub facility achieves a very
economical means of accessing VSAT technology. Most interactive VSAT equipment suppliers run data
rates between 56 Kbps and 512 Kbps outbound and 56 Kbps to 256 Kbps inbound.
Microwave Communications
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Microwave communications describes point-to-point radio operating across the frequencies of 1GHz to
30GHz of the electromagnetic spectrum. Once exclusively the territory of the common carriers, microwave
communications has become a major competitor of standard, wireline telephone communications.
Generally speaking, microwave communications involves sending waves of information between a radio
transmitter and a radio receiver, each mounted on a tower. As microwave communications requires a
clear “line of sight,” there must be no obstructions in the path followed by the waves, so the towers must
be high enough to avoid interference from buildings, trees or other structures.
Intelligent Networks (INs)
The concept of intelligent networks (INs)—based on digital switching, database intelligence and ITU-T
SS7 (Signaling System 7) signaling—provides customers with a host of advanced network features and
services such as distributed call processing, one-number services, class of service (COS), private
network services and network billing enhancements, to name a few. As shown in 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) are
omitted for clarity.
SSPs are typically end offices, but can be tandem switches in sparsely populated areas with capabilities
that go beyond straightforward switching—enhanced software and hardware enable them to communicate
with application databases. The SSP formats and sends a request (an SS7 message) to an SCP, where
an application database is stored, and suspends call progress until a response is received that provides
the 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 and
routing request messages—between SS7 nodes. As a cost-effective alternative to interconnecting all
nodes directly to one another, STPs serve as centralized hubs in the SS7 network. Many nodes are linked
to a single STP, and in turn, all STPs are interconnected. Messages sent to an STP are routed to the
correct destination node. Because an STP connects to many SS7 nodes, it must be capable of handling
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high-message throughput and must be equipped to support many signaling links. Redundancy and
reliability are also key attributes.
SCPs are the centers of intelligence in the network. The main function of an SCP is to accept a query for
information, retrieve the relevant information from the appropriate database and send a response to the
originator (typically an SSP) of the request. Adding or updating databases, without affecting any other
node in the network, can increase the functionality of an SCP.
Integrated Services Digital Network (ISDN)
The Integrated Services Digital Network (ISDN)—defined and described in ITU-T Recommendations—
extends to the customer premises the capabilities and benefits of the IN to support a wide range of voice
and nonvoice applications by the same network. Built on top of standard, unshielded twisted-pair (UTP)
telephone wire, ISDN is an end-to-end (customer premises-to-customer premises) digital network that
integrates enhanced voice and image features with high-speed data and text transfer.
ISDN provides two interfaces—basic rate interface (BRI) and primary rate interface (PRI). Both interfaces
consist of B channels and D channels. The B channels provide transparent digital channels for voice or
high-speed data transmission at 64 Kbps per channel. The D channel (one D channel per interface)
provides a nontransparent channel for signaling, telemetry and low-speed packet switching at 16 Kbps or
64 Kbps:
• A basic rate service provides two 64-Kbps B channels and one 16-Kbps D channel (that is, 2B + D).
The D channel provides call control and an option to carry X.25 data packets end to end.
• A primary rate service provides thirty 64-Kbps channels and one D channel (30B + D) in Europe and
twenty-three 64-Kbps B channels and one 64-Kbps D channel (23B + D) in North America and
Japan.
Mobile Communications
Modern mobile communications revolve around two main ideas. The first, “smaller is better,” breaks a
relatively large geographic area into contiguous hexagonal cells of equal size. Radio towers situated at
the center of cells communicate with mobile telephones (cell phones) and other devices (such as mobile
personal digital assistants).
The second concept, “less is better,” reduces the power transmitted from respective towers. Taken
together, reduced transmitted power and small cellular areas enable radio frequencies to be reused. This
has a multiplication effect, enabling many more users to be served compared to transmitting over a larger
area at a higher power.
Cellular/Wireless Basics
The centerpiece of cellular/wireless networks is the mobile switching center (MSC), which interconnects
small radio coverage areas into a larger system. See 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 stations
to a Mobile Switching Center (MSC), which tracks the location of devices that have been turned on.
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Call Origination
When a device such as a mobile phone is turned on, it scans for an unused command channel, locks on
and 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 base
stations can simultaneously transmit on several different radio channel frequencies and thus can be
in contact with many mobile devices simultaneously.
• The MSC connects the call to the PSTN, which then connects the call to its destination (home phone,
PBX or another mobile phone).
• The MSC commands the mobile phone to switch to a talk channel (radio channel selected by the
MSC) 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 command
channel. Assuming the call originates from a residence:
• The PSTN switches the call to the MSC, which has been tracking the whereabouts of the mobile
phone from the moment it was turned on. (If the mobile phone is engaged in a call [status = busy], the
MSC announces the status to the caller. If the mobile phone is turned off, the MSC provides a
different announcement.)
• The MSC sends a page (bearing the phone’s mobile telephone number) to the cell site, which
broadcasts it to all phones in the cell site.
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• When the target phone responds to the page command, it is commanded to switch to the talk
channel (radio channel selected by the MSC) for the call. The MSC then commands the phone to
alert 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”), it
may traverse a cell site and find itself in a different cell site. At some point, the received signal strength of
the 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 with
a stronger signal. A change in frequency occurs, coordinated by the MSC, as adjacent cell sites operate
on mutually exclusive frequency “lists.”
Handoff to a Different MSC
Conceptually, the situation is similar when the new cell site is served by a different MSC. The details are
complex, though, as the MSCs must exchange messages (IS-41 or SS7 messages depending on the
network) 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 MSC
and the first MSC’s switch, a trunk to the second MSC and the second MSC’s switch, a trunk to the new
base station, a radio channel to the mobile phone and the mobile phone itself. Theoretically, the process
can continue indefinitely, the talk path accumulating additional MSCs (and trunks) during the mobile
phone’s apparent odyssey. Practically, calls are finite, and even the longest ones terminate before the
switch 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 the
distant location communicates with the phone’s home MSC to find out whether or not to provide service to
the visiting phone (credit worthiness), as well as the service options and phone features that it should (or
should not) honor. In this way, the home MSC knows the whereabouts of the mobile phone, specifically, it
knows the identity of the visited MSC, and the visited MSC knows how to provide service as if the
subscriber resided there. Call origination is handled by the visited MSC and is the same as described
above (that is, the visited MSC treats the roaming phone as if it were 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 the
home MSC (say Dallas), which will notice that the mobile phone is out of the home service area. The
following steps now occur:
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• The home MSC sends a message to the visited MSC (say New Orleans) requesting that the 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 mobile
phone’s telephone number (as sent in the original request from the home MSC) and sends the TLDN
to 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 phone
currently 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 (ringing
tone) and commands the phone to switch to the channel that will carry the call. The subscriber
answers and converses with the originating party.
• When the call finishes (one or both parties hang up), the visited MSC releases the TLDN to the pool
of available numbers, for use on another call.
TLDNs are not published, as they are reserved for internal network routing only. During a random call
origination, if a TLDN is dialed in error, the PSTN will route the call to the MSC (assumed to be in New
Orleans), 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 digital
cellular radio system known as Global System for Mobile Communications (GSM). There also exists an
array of mobile communications technologies that focus on a variety of subsegments of the mobile
communications marketplace. That marketplace can be further subdivided into segments that address
different aspects, characteristics or needs of its customer base.
Cellular communications began in forms like advanced mobile phone service (AMPS) and digital-
advanced mobile phone service (D-AMPS) and variations thereof, followed by upgrade systems like IS-54
and second-generation systems like CDMA/IS-95 and GSM/Digital Cellular System (DCS) 1800. Private
mobile radio (PMR) services now also include Specialized Mobile Radio (SMR) services. Some of the
major types of mobile systems in use today include:
• Wireless Application Protocol (WAP) supporting wireless Internet communications
• Paging
• Private Mobile Radio (PMR) and Specialized Mobile Radio (SMR)

• Time Division Multiple Access (TDMA) digital cellular phone service
• Code Division Multiple Access (CDMA) digital cellular phone service (also known as spread
spectrum)
• Digital European Cordless Telecommunications (DECT)
• Analog/digital cellular phone systems (AMPS/D-AMPS)

• Global System for Mobile Communications (GSM)
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• Personal Handy-Phone System (PHS)
• Digital Short Range Radio (DSSR)

• Personal Communications Services (PCS) and Personal Communications Networks (PCNs)
• Universal Mobile Telephony System (UMTS)—a wideband CDMA wireless technology in Europe
The number of subscribers who can share a cell site is much greater than the number of radio channels.
This is because not all subscribers require service at the same time. An analog cell site with 50 radio
channels has the capacity to support approximately 1,000 subscribers. On the other hand, digital
standards support approximately 3 to 20 times as many subscribers as an analog system. In most areas,
mobile phones are able to switch between analog and digital modes of operation. In this way, if a digital
service provider is not present in a geographic area, users switch to the analog mode for service.
Wireless Data Communications
Second-generation mobile data speeds were around 14.4 Kbps. Prior to third-generation (3G) mobile
system networks, the highest practical speed available was the General Packet Radio Service (GPRS)
bandwidth of 30-40 Kbps (on the downlink) to 40-50 Kbps (on the uplink). 3G networks can transmit data
at speeds of 144 Kbps.
Wide-Area Data Networks
Broadband ISDN (B-ISDN)
Eventually, the need was identified for the development of an advanced form of ISDN capable of carrying
multimedia information at rates of hundreds of millions of bits per second (bps). In the ITU, two main
service categories 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 be
unidirectional and in real time between users, or between a user and a host. Examples include
videoconferencing and high-speed data transmission. Message services will offer communication via
storage units such as a mailbox or as message-handling functions, which not only include speech, but
also moving pictures and high-resolution images. Retrieval services offer user access to information
stored centrally and accessed on demand. Examples of these services include film and high-resolution
images, together with audio.
Distributive services are differentiated between those services with user presentation control—such as
broadcast services for TV and radio—and those with individual user control. The availability of high
bandwidth enables a number of different types of information to be supported by one service, resulting in
the development of multimedia services. For example, video telephony includes audio and video and
possibly text and graphics. Many of the broadband services—such as video signal transmission—require
variable bandwidth, which is best met by a packet-based technology. For this reason, ITU chose
Asynchronous Transfer Mode (ATM) as the target transfer mode for B-ISDN.
Asynchronous Transfer Mode (ATM)
It is important to note that ATM is a transfer mechanism and as such is, in principle, independent of
transmission technology. It is a fast-packet, switching technique that uses short, fixed-length packets
called cells. As such, ATM is also referred to as a cell-switching technology.
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In principle, it is quite similar to other packet-switched techniques; however, the detail of its operation is
somewhat different. Each ATM cell is made up of 53 octets or bytes. Of these, 48 octets make up the user
information field (payload), and five octets make up the header (overhead). The header identifies the
“virtual path” to be used in routing a cell through the network. The virtual path defines the connections
through which the cell is routed to reach its destination.
ATM is a form of TDM. It differs from synchronous multiplexing in that channel separation is not
dependent on reference to a clock. Figure “Depiction of ATM and Synchronous Transfer Mode (STM)
Cells” gives an example of how ATM and STM cell arrangement varies.
Figure 13: Depiction of ATM and Synchronous Transfer Mode (STM) Cells
Source: Nortel Networks.
Frame Relay
Frame relay, also called fast packet switching, is a high-speed, packet-switching technology that can be
used in conjunction with ATM and, in fact, was optimized to provide access to cell-relay networks. It is
used primarily in data communications environments and is also considered an alternative method for
transporting voice communications (such as voice over frame relay).
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 packet
throughput of X.25 packet-switching networks by letting information move across a network guided and
checked 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
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• Discard incorrect frames
• Fill interframe time

One of frame relay’s major limitations is that it does not provide any error correction inside the frame relay
network. Frames that are in error are detected and discarded by network nodes. Recovery by the
retransmission of missing frames is left to the end nodes in the network. This is a major quality of service
(QOS) issue when considering frame relay for the transport of voice communications.
Frame Relay, X.25 and TCP/IP Compared
Unlike X.25, frame relay does not store and forward, but simply switches to the destination part way
through the frame, 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 a
number of areas:
• Transmission Control Protocol (TCP) operates at Level 4 of the ISO stack; Internet Protocol (IP)
operates at Level 3. The X.25 protocol operates at Levels 2 and 3; Levels 4 through 6 are not
implemented.
• TCP/IP has only end-to-end error checking and flow control (Level 4), while X.25 is error checked
from node to node (Level 2).
• TCP/IP has a much more complicated flow control and window mechanism than X.25 to compensate
for the fact that a TCP/IP network is completely passive.
• The electrical and link levels are tightly specified in the X.25 specifications, while TCP/IP is designed
to travel 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]).
Digital Subscriber Line (DSL)
Asymmetrical Digital Subscriber Line (ADSL) is a broadband service currently being offered by telcos
(where available). ADSL’s most popular applications so far seem to be located in home offices,
consumers and small businesses. ADSL previously allowed users to dial into an Internet service provider
(ISP) at 16 Kbps and then download files at 1.544 Mbps. Upgrades to the service currently allow users to
connect to their ISPs at up to 1 Mbps (or higher speeds) upstream while maintaining download times at
T1 (or higher) speeds. This would support Web pages with more complicated graphics, videos and sound
clips. The ADSL speed ranges are shown in Table “ADSL Service” (CO stands for central office.)
Table 2: ADSL Service
Service Name Download Upload Speeds (Subscriber to Distance Limit (CO to
Speeds (CO to CO) Subscriber)
Subscriber)
ADSL 1.5 Mbps to 16 Kbps to 640 Kbps 12,000 to 18,000 feet maximum
8.192 Mbps on one pair of wire
Today’s analog modems routinely offer up to 56 Kbps. However, 56 Kbps is probably the practical limit for
analog modems. ISDN can increase this to 128 Kbps, but this is still slow compared to ADSL speeds of
between 1.5 Mbps and 8 Mbps. ADSL offers customers the instantaneous downloading of massive
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graphics and even video applications over the Internet. A significant benefit of ADSL is that it is “always
on,” allowing continuous usage.
In addition to ADSL, the generic digital subscriber line (xDSL) family comprises other variations of DSL
including 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, a technology that competes with DSL in the broadband access area, have also emerged
as a solution for providing significantly faster Internet access speeds, compared to analog and digital
modems, at relatively inexpensive rates.
Cable modems are customer premises equipment (CPE) access devices that enable computer equipment
to connect to the Internet and other online services over a cable television (CATV) network. High-speed
connections over the cable network allow users to stay connected to the Internet and other online
services 24 hours a day without interference to telephone service or cable TV service.
In the most prevalent scenario, a PC is connected to an external stand-alone cable modem via an RJ45
Ethernet network interface. The cable modem is attached to a CATV network via an F connector. An F
connector is a 75-ohm coaxial cable connector commonly found on televisions 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 modems
can be connected to hubs, switches and routers to allow networks access to the Internet. Some cable
modems are now including routing and four-port hub capabilities into a single cable modem device.
Newer versions of cable modems include Universal Serial Bus (USB) connections for PC connectivity and
peripheral component interconnect (PCI) cable modem cards.
While the bandwidth that cable modems can support is impressive, it is shared among users, either those
on a neighborhood network node or those on a corporate remote LAN network. Performance will vary
depending on the number of users who are online simultaneously and the type of work each person is
doing. Access speeds can and will be significantly less than what is theoretically possible. While
bandwidth can be supported in the 10-Mbps range and higher, more realistic bandwidth numbers are in
the 1-Mbps to 3-Mbps range for downstream traffic and 250 Kbps to 2.5 Mbps for upstream traffic.
Synchronous Digital Hierarchy (SDH)
Digital telecommunications networks in North America, Europe and Japan, along with their
plesiochronous (almost synchronous) transmission equipment, were not based on a common standard,
making any interworking of these systems extremely cumbersome. The plesiochronous digital hierarchy
(PDH) was not suited to the efficient delivery and management of high-bandwidth connections, nor was it
flexible enough to meet the demands being placed on it.
To overcome the problems associated with the PDH, synchronous transmission was developed. In 1985
ANSI began work on Synchronous Optical Network (SONET) based on a proposal by Bellcore (now
Telcordia Technologies). In 1988 the Comite Consultatif International Telegraphique et Telephonique
(CCITT) (now the ITU-T) published the World CCITT Synchronous Digital Hierarchy (SDH) standards in
its Blue Book. SONET/SDH and ATM are now the core transport and switching technologies for B-ISDN
and offer network operators and end users several advantages, including:
• Increased available bandwidth
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• Reduction in network equipment
• Increased network flexibility

• Capability to work with existing plesiochronous systems (for migration purposes)
• Cost savings in hardware
• Savings from reduction in maintenance and operations costs

• Improved network restoration and reconfiguration
• Better availability and faster provisioning of services
• Future-proof network supporting metropolitan area networks (MANs), PCNs, B-ISDNs and other
networks.
• Network software controllable

• Standardization such that transmission equipment from different manufacturers can interwork on the
same link
The standardization of SDH involved the formation of transmission signals formed on the basis of
synchronous transport module level 1 (STM-1) frames. For each STM-1 frame there are nine rows by 270
columns of standard bytes, producing a total of 2,430 bytes. With a standard frame duration of 125
microseconds, the basic synchronous transmission level would be 155.52 Mbps. Multiples of STM-1, such
as STM-4 (622 Mbps) and STM-16 (2.5 Gbps) are produced by interleaving and multiplexing several
STM-1 frames.
Aside from broadband ISDN, synchronous transmission systems play an all-important role in services
networks: such as personal communications where individuals are assigned a personal number
irrespective of location, and the network automatically routes their calls to them. Personal
communications obviously need extremely flexible, intelligent network and sophisticated signaling
systems. SDH provides the infrastructure for such a network.
Fiber Distributed Data Interface (FDDI)
FDDI is a networking protocol that works as a token-passing ring. The nominal speed is 100 Mbps, and it
achieves almost that in an ideal configuration. It is also possible to get 200 Mbps by using duplex FDDI, if
there is duo-ring cabling. In case of failure, FDDI is autowrapping. It is known for its speed, resilience,
length of cabling between nodes (up to two kilometers) and total size of network. FDDI is also renowned
for the high prices of its cards compared with most other LAN types. For these reasons, it is usually found
in large LAN or campus backbones.
Despite the name, FDDI does not have to run on fiber. Older standards for running on copper were
Shielded DDI on shielded cable (SDDI) and Copper DDI on unshielded cable (CDDI), which were
subsumed into the Twisted Pair-Physical Medium Dependent (TP-PMD) standard. All FDDI standards are
set by ANSI.
A recently standardized enhancement to FDDI, called FDDI-II, supports the same functionality over fiber
and twisted copper pair wiring as FDDI, but also uses 64 Kbps channels of the 100 Mbps bandwidth to
carry voice and video calls.
Transmission Media
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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 are
loosely twisted around one another to minimize electrical interference. A four-wire circuit is a pair of two-
wire circuits. Twisted-pair wire, often called standard telephone wire, comes in many forms: some cables
are waterproof, some have flame-resistant coverings and some are shielded for extra protection against
electrical interference.
In a local environment, in which terminals are attached to a host in close proximity, data transfer generally
occurs over standard twisted-pair wiring or coaxial cable. Coaxial cable consists of two conductors—a
copper mesh tube and a wire that runs down the cable’s central axis. A “dielectric” separates the two
conductors and provides insulation so that they cannot contact one another. A cable that consists of two
central conductors in the same mesh tube is called “twinaxial cable.”
Coaxial cable carries signals with much higher bandwidths than twisted-pair wire, and it is often used to
carry high-speed data traffic (for example, cable modems), as well as television broadcast signals, thus,
the term cable TV. However, coaxial cable has certain limitations: expense, bulkiness and inflexibility.
Coaxial cable can cost from 10 to 20 times more than twisted-pair copper wire, and it is very difficult to
rearrange, causing problems in office environments where equipment moves and changes occur often.
Products called “coax eliminators” were developed to replace coax with twisted-pair wire. Coaxial cable is
still largely used in IBM environments and is the basic medium for LANs, cable modems and cable TV.
The increase in availability and lower prices of fiber optic cabling and equipment has resulted in its
incorporation into a large proportion of new networks. This is especially the case where there is a need to
future-proof installations against the rising demand of bandwidth.
Figure 14: Cabling Types
The three basic types of communications cabling: twisted pair (in this example, shown in a 25-pair bundle), coaxial
cable (coax) and fiber optic cable.
Accuracy Controls/Error Correction
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A communications line carrying both data and control information in a serial datastream is subject to
interference by storms, cross-talk from other lines and other phenomena that introduce errors into the
transmission. As previously noted, a communications line is essentially two wires in a multiwire cable that
runs within walls and floors, out and under streets, or along telephone poles to a service provider CO. The
wire can be handled by repair personnel, run over by vehicles, dropped or subjected to power surges
while data is being transmitted over it. These events can cause anywhere from one or two bits to several
thousand bits to be dropped from the datastream or otherwise corrupted, thus compromising the integrity
of the information.
Thousands of bits travel over the line every second, and the loss of even one bit could alter a character or
control code; therefore, data transmission requires accuracy controls. These controls consist of bits added
to characters and blocks of characters at the sending end of the line. These bits are then checked and
verified at the receiving end of the line to determine whether bits were lost during transmission. Two basic
data communications controls include:
• Vertical and horizontal parity
• Cyclic redundancy checking (CRC)

In data communications, parity denotes a relatively simple error-checking technique in which all
characters are specified as having either an odd number of bits (if odd parity is desired) or an even
number of bits (if even parity is chosen). Parity actually refers to adding a bit (a parity bit) to make the
number of bits in individual characters consistently odd or consistently even. If the receiving equipment in,
say, an odd-parity system gets a character consisting of an even number of bits, it is assumed that a bit
was corrupted and that the character is in error. The situation is similar in an even parity system when an
odd number of bits are received. In a vertical parity system (vertical redundancy check), a bit can be
appended to each character and verified by the receiving device. In a horizontal parity system
(longitudinal redundancy check), checking is performed on a block of characters.
CRC, a more powerful error-detection technique used in synchronous transmissions, uses polynomially
generated check characters. CRC views an entire block of data as one long binary number that is divided
by another fixed binary number. After the quotient is discarded, the 16-bit remainder is transmitted as two
block-check characters (BCCs). BCCs appear in the Frame Check Sequence field of HDLC frames (see
the previous figure “HDLC Framing”). At the receiving end, BCCs are computed from the received data
and compared to the BCC embedded (by the sending end) in the block. If the BCCs agree, the block is
declared error-free. If the BCCs disagree, it means that one or more errors exist in the block. The entire
block is suspect and must be retransmitted.
Since an error detected in a message must be corrected immediately, a transmitting device must receive
an acknowledgment on a real-time basis of the received data’s accuracy. Because data must be
retransmitted if there is an error during transport, transmitting equipment must store all information that
has not been acknowledged by the receiving station.
There are a number of error-correction techniques, and more are being developed all the time. The two
basic 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 end
of the block. The receiving equipment verifies the accuracy of the data and sends back an
acknowledgment (ACK) or negative acknowledgment (NAK) if an error is detected. If an ACK is
received, 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.
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• 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 subsequent
block. There is no pause in transmission at the end of a block, and the equipment proceeds to
transmit the next block. If a NAK is received, the sending unit backs up the number of (“N”) blocks to
the last acknowledged block and retransmits all subsequent data. The maximum number of blocks
that can be transmitted before an acknowledgment is received is specified by the protocol.
The CRC technique detects received errors with high probability but has no capability to correct them. To
do so, redundant characters must be added to the data and sent as part of the block. The benefit of this
additional complexity is that errors can be corrected without impacting the transmitter.
Local Area Networks
Local Area Network (LAN) Protocols
A local area network (LAN) is a communications network that is usually owned and operated by the
enterprise customer. A LAN operates over a limited geographical area and enables many independent
peripheral devices, such as PCs and terminals, to be linked to a network through which they can share
central processing units, memory banks and a variety of other resources. Some organizations (for
example, banks or financial institutions) have enough data traffic within a city to make intracity networking
viable. In this case, individual LANs are interconnected to form a metropolitan area network (MAN).
LANs are generally fault-tolerant, incorporating a simple architecture with control distributed among
participating 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. The U.S. protocol standards for LAN
connections are the Institute of Electrical and Electronic Engineers (IEEE) 802 standards. Two major LAN
standards are:
• Ethernet (802.3)
• Token Ring (802.4)
Ethernet was developed by Xerox and utilizes a 10-Mbps system operating over copper cable. Today, this
is mainly Unshielded Twisted Pair (UTP) wiring, but shielded and coaxial cable are also possible. In
theory, it allows an unlimited number of devices to be connected, but it is only effective for up to 50
workstations over a maximum distance of 500 meters on a single segment. Today, most LANs have less
than 10 stations on a segment, and many have a single station per hub or switch port. It can be built on a
multidrop bus or on a star topology.
Ethernet transmits data through carrier-sense multiple access with collision detection (CSMA/CD). This
means that one workstation transmits data as required. If, however, two or more workstations are
transmitting data, their signals will crash and destroy each other. CSMA/CD allows the workstations to
detect this collision and ensures the data is retransmitted after a random amount of time. CSMA with
collision avoidance (CSMA/CA) offers an alternative technique. Each workstation then “listens” to the
cable. If it “hears” something on the line (such as another workstation transmitting), it will not transmit its
own data, thus avoiding a collision.
As the volume of network traffic increases, however, the bandwidth offered by traditional 10-Mbps
Ethernet LANs becomes inadequate for a growing number of desktop/server computing environments.
The first step in alleviating these bottlenecks was the introduction of switched Ethernet, which provides a
10-Mbps path for a limited number of devices rather than a “pool” of bandwidth (such as 10 Mbps) shared
by all devices. For some environments, this is still inadequate, as many organizations seek higher-speed
solutions, particularly for the local backbone supporting the “server farm.” Among the high-speed LAN
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technologies available today, 100Base-T (Fast Ethernet) has become the choice in a large number of
organizations. Building on the broad acceptance of 10Base-T Ethernet, Fast Ethernet technology has
provided a smooth, nondisruptive evolution to 100-Mbps performance.
More recently, 1000Base-T (Gigabit Ethernet) and its associated products became available for
implementation and use in high-speed desktop LAN technology. Gigabit Ethernet provides speeds (1,000
Mbps) comparable with and faster than ATM, a competing technology, and requires the use of fiber optic
cabling.
Developed by IBM, Token Ring typically runs on a ring topology, although star networks based on
switches are also possible. There are never any collisions in Token Ring. The first station to switch on in a
network owns the “token,” a unique sequence of control bits. While it has this token, it is capable of
transmitting data, and no other workstation can take the token on that revolution. After the current owner
completes transmitting its data, it passes the token to the next workstation on the network, and so on.
As mentioned earlier, a LAN is limited in geographical scope (generally up to 10 kilometers), with devices
physically separated but not mobile. Devices can be on different floors of a building, on the same
industrial or university campus, or in several buildings in the same city. LAN devices are usually
intracompany, privately owned, user administered and not subject to regulation. LANs are structured or
integrated into a discrete physical entity with devices interconnected by a continuous medium, such as
copper cable, and supportive of full connectivity, allowing every user device on the network to
communicate with every other user device. A LAN also interconnects two or more communicating devices
that are high speed, typically supporting rates from 1 Mbps to more than 10 Mbps.
There are three basic LAN topologies: linear bus, ring and star. The topology can be defined as the
physical layout of the network.
Figure 15: LAN Topologies
The Bus topology is typical of Ethernet networks; rings are associated with token-passing (Token Ring) networks.
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In linear bus topology, stations are arranged along a single length of cable that can be extended at one of
the ends. A tree is a complex linear bus in which the cable branches at both ends, but which offers only
one transmission path between any two stations. All broadband networks and many baseband networks
use a bus or tree topology.
In a ring topology, stations are arranged along the transmission path so that a signal passes through one
station 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. Any
communication 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 passes
all the network’s stations. In star networks, signals sent through the central node are circuit-switched to
the proper 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 each
department’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.
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When used properly, the LAN provides a common interface for a diversity of otherwise incompatible
equipment.
In the office, the LAN can give users fast and efficient access to a common pool of information such as
customer lists, schedules and document formats. It also allows an entire office to share expensive
resources such as printers and duplicators, thereby streamlining the production and distribution of paper
documents. Ultimately, a LAN can eliminate the need to circulate paper documents by electronically
distribution (e-mail, instant messaging) of memos and other textual materials to each worker. In an
automated factory or laboratory, the LAN can simplify the process of “retooling” by allowing the user to
download a number of programmable devices simultaneously from a central site. It can also isolate
failures and bottlenecks in plant operations.
At one point, interest in fiber optic LANs was increasing, with several fiber optic systems available for LAN
applications based on a mature optoelectronic technology. More recently, however, interest in this
technology waned as copper cabling became more sophisticated, with Category 5 structured cabling (IBM
standard) able to carry all the LAN types of traffic, including FDDI and ATM, at a far cheaper cost and with
less difficult installation.
Internetworking
The Internet now has millions of hosts connecting billions of people all over the world. Educators,
consumers, telecommuters, librarians, hobbyists, researchers, government officials and businesses are
among the groups today that use the Internet for a variety of purposes—from communicating and
collaborating with one another to accessing valuable information and resources. The Internet provides
connectivity for a wide range of application processes called network services. For example, users can
exchange electronic mail, instantly message each other in groups, access and participate in discussion
forums, search databases, browse indexes and transfer files. Also, use of the Internet 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 or
a mixture of the two. As the Internet continues to grow in size, popularity and efficiency, as LANs
proliferate in business environments and as enterprises rely on several communications networks
simultaneously, 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 product
can incorporate more than one function:
• Bridges link networks that are fundamentally compatible. They operate at Layer 2 (Data Link) of the
OSI reference model without interpreting the protocols at Layer 3 and above. Bridges can be
intelligent, selectively forwarding or filtering frames across networks according to the Layer 2 address
of devices on respective LANs.
• Routers are switching devices that connect two LANs where multiple paths exist. Operating at Layer
3 of the OSI model, they inspect addresses and route packets of data between networks. Very often
today the routing function is added to a switch to give Layer 3 functionality to a Layer 2 device. Many
routers today can also handle the transport of voice communications for IP telephony or can be
upgraded as such.
• Gateways interconnect dissimilar networks, such as the PSTN (voice) to a WAN (data) for IP
telephony. For that reason, gateways are more sophisticated than bridges, and they incorporate
protocol conversion functions. Gateways are also used to connect dissimilar LANs, such as Ethernet
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and Token Ring, or to connect LANs to other types of networks or architectures, such as X.25 public
networks or Systems Network Architecture (SNA) hosts.
Figure 16: Bridges, Routers and Gateways
Bridges operate at the data link layer of the OSI model, routers are the network layer devices that select pathways to
send 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 premises that primarily establishes voice-grade
circuits over access lines between individual users and the public switched telephone network (PSTN).
The transmission of PBX calls is still typically over copper wiring and fiber access lines between end
users and the PSTN. However, enterprise telephony calls are increasingly being routed through gateway
devices over Ethernet-based LANs and WANs, the Internet, and even ATM and frame relay networks.
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A PBX is a private system in that it is typically used by one organization or one building complex with
capacity requirements ranging from less than 100 lines (but typically above 40 lines) up to several
thousand lines. Branch suggests a remote subsystem. The first PBXs were like small central office
switches located on a customer’s premises. In telephony, an exchange is defined as a group of
equipment controlling the connection of incoming and outgoing calls—in short, a switch.
In addition to the solid reliability and performance that have been characteristics of PBX technology over
its life cycle, advanced functionality such as IP telephony, CRM and call center technology, voice/unified
messaging, computer-telephony integration (CTI), broadband capabilities, PBX networking and inbuilding
wireless communications have all helped the PBX extend its mature life cycle. Digital signaling and new
interfaces between PBXs, computers and the Internet have more recently helped the PBX become less
proprietary and more compatible with a wide range of standards and application programming interfaces
(APIs).
In traditional digital PBX systems today, time division multiplexing (TDM) is still the most commonly
deployed switching matrix design. However, the packet-switching designs in newer IP-based telephony
systems, in particular, are rapidly pushing TDM and circuit-switching further along in their life cycles.
Traditional circuit-switched PBXs are now supporting interfaces to IP, ATM or frame relay infrastructures
and equipment via an assortment of new software, gateways, trunk cards and line cards. For example,
most PBXs today support 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 switching matrix.
In addition to the traditional PBX vendors doing VoIP, newer entrants into the PBX marketplace currently
include vendors such as Cisco Systems (CallManager), 3Com (NBX/SS3), Artisoft (TeleVantage),
Shoreline (IP Voice Communications System) and Vertical Networks (InstantOffice), among others. These
vender systems are targeting the IP telephony space by offering several variations on a theme for
transporting VoIP-based data networks for the enterprise.
Key Telephone Systems (KTS)
Smaller organizations (typically with less than 40 stations) use a key telephone system (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. The heart of a KTS is often called a key service unit (KSU). It is the common control
cabinet for all major system operations and functions. A KSU performs CO line connections, intercom
functions, paging and station connections. Each KTS extension has a lamp indicator for all available
outside lines, showing whether or not they are busy as well as giving visual indication of an incoming call.
A KTS does not require dialing a code to gain access to outgoing lines; PBXs invariably require dialing a
number or code (such as “9”). A hybrid 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, such
as in a call center environment. A PBX system equipped with integrated ACD software enables the switch
to 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 from major vendors
such as Aspect and Rockwell. Leading PBX/ACD vendors such as Avaya, Nortel Networks and Siemens
also offer integrated ACD solutions for their PBX systems, giving customers a choice between
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implementing a PBX/ACD solution with better integration capabilities or a third-party solution from a
stand-alone ACD vendor.
The agent capacities of PBX/ACD systems allow those vendors to compete for large call center
installations, traditionally the domain of stand-alone ACD vendors like Aspect and Rockwell. In addition,
the trend toward networked call centers has advantages for the PBX/ACD vendors, who see themselves
as having deeper skill sets and 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 require
application-based intelligence to standard PC-server platforms. Utilizing CTI technology and application
programming interface (API) software, these solutions function as call/connection servers within the
enterprise network. Call center businesses of all sizes have substantial investments in modern ACD call
center technology. As such, ACD remains a major revenue stream for PBX vendors.
With the shift toward supporting e-commerce and customer relationship management (CRM) solutions
and 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 the
telephone or chatting via e-mail.
• Universal queue—the capability to route contacts to an agent regardless of channel. The result is that
agents 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 any
contact 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 users
to access, forward, reply to, schedule the delivery of and tag/edit messages, among other functions. End
users, or subscribers, are the owners of personal voice mailboxes in a voice mail system. Subscribers can
access messages in their voice mailboxes from telephones or PCs by entering account numbers and
personal passwords. Telephone access, allows information entry and all system commands to be
performed via the phone’s touch-tone keypad. In addition, more advanced systems deploy speech
recognition technology to manage messaging via simple voice commands over the phone.
Integrated voice mail systems typically have a message-waiting indicator such as a light on a telephone or
icon on an alphanumeric display. A ringing telephone can default to a voice mailbox that delivers 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 menu
prompting for message management functions, including retrieval and playback of messages, message
disposition (deleting, saving, replying to, or forwarding messages to others), sending new messages to
one or more subscribers, and changing the setup of mailbox facilities—greetings, passwords, distribution
lists and access to live assistance. In addition, the growth of e-mail usage has increased the popularity of
the 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.
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The popularity of Internet-based e-mail has positioned it as a target for integration/consolidation with voice
mail systems and services. These services are particularly well suited for small businesses and SOHO
environments 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 for
Internet Messaging (VPIM) industry standard. Popular Web browsers are also being employed by users
for PC access to voice mail servers over the Internet and can be used by messaging system
administrators to remotely manage user activities via the Web.
Voice mail systems can be either stand-alone solutions or integrated software/hardware solutions
installed in the cabinet of a telephone system, such as a PBX. Major voice mail suppliers include the
traditional PBX vendors and their products:
• Avaya Intuity AUDIX Multimedia Message Server, Octel Messaging Systems
• Nortel Networks Meridian Mail

• Siemens PhoneMail
Major stand-alone voice mail systems include:
• Cisco Unity (formerly Active Voice)

• Active Voice Repartee CTI and Repartee VP (now part of NEC America)
• Captaris CallXpress
• Key Voice Interchange
Unified messaging is an extension of voice mail that enables subscribers to access and manage
messages such as voice mail, fax mail and e-mail from a single user interface. The goal is to simplify and
speed 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 user
interface. The user interface itself is typically a desktop PC driven by an application software module or
integrated software load. The PC-based interface also enables data files to be easily attached and
retrieved with any form of message medium, not just e-mail. Several of today’s sophisticated unified
messaging packages allow subscribers to embed voice messages in fax and e-mail files, view faxes
onscreen, be notified of e-mail (by PC or telephone) and redirect e-mail to a fax machine via telephone
commands. The telephone is another user interface (TUI) alternative for unified messaging. In addition to
listening to voice messages, a text-to-speech option is generally offered for the telephone retrieval of text
messages from e-mail or fax.
Some of the leading unified messaging vendors include Cisco (Unity), Active Voice/NEC (Repartee CTI)
and Captaris (CallXpress). Traditional PBX vendors offer the following unified messaging solutions:
Siemens (Xpressions), Nortel Networks (CallPilot) and Avaya (Intuity AUDIX Multimedia Message Server
and Unified Messenger).
Terminal Adapters (TAs)
A terminal adapter has two main tasks. One is to adapt the format of the data 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. Like modems, terminal adapters can be
packaged in a variety of ways. Individual basic-rate TAs can be obtained in stand-alone boxes, each with
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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 personal computer 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, a
processor 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 a
communications environment. An intelligent switch routes messages among the network’s various end
points 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 applications
processors. It gathers, queues and multiplexes their transmissions onto one or more high-speed network
trunks and participates in the network’s control and management, again either under the direction of a
master processor or as a peer of other concentrators and switches. Most high-end communications
processors perform all three of these tasks.
Modems
Virtually all modern telephone networks use digital transmission to connect digital switching offices. At the
edges, however, networks still use analog facilities to connect to customer equipment, especially
residence 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, the
information must be converted into a continuous (analog) line signal. A device called a modem, a
contraction of the terms modulate/demodulate, performs the conversion from data format (1s and 0s) to
analog format. Modems are always used in pairs. The unit at the sending end converts information
coming from a host, PC or terminal. At the receiving end, another modem converts the analog signals
back into data format before acceptance by the receiving devices. In the case of a fax, the receiving
modem’s output is digitally processed by a DSP in the receiving fax machine to create a nearly identical
rendering of the original image.
Modern modems can transmit and receive simultaneously (full duplex) on the public network at speeds up
to 56 Kbps. The V.90 standard supports operation at 56 Kbps. V.34 is a popular international standard
today for dial-up modems, supporting speeds of 28.8 Kbps with full-compression facilities.
Multiplexers
Multiplexers combine streams of data from many individual low-speed channels and transmit a combined
stream over one high-speed communications link. Multiplexers maximize the efficient use of
communications links in a network because users can lease one high-speed line for much less than it
would cost to lease many low-speed lines. Multiplexers generally fall into one of two very broad
categories:
• Frequency division multiplexers (FDMs)

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• Time division multiplexers (TDMs)

Frequency division multiplexers (FDMs) are the earliest and least sophisticated form of multiplexing
equipment. FDMs divide the allocated bandwidth of a conditioned analog line into independent,
permanently assigned, lower-speed subchannels that operate on a particular frequency within the
spectrum.
TDMs are digital devices that accept multiple digital inputs and convert them to one composite digital
output. Rather than divide a communications link into separate channels as FDMs do, a TDM divides
bandwidth into time slots while maintaining the integrity of the single channel. Within the TDM
classification, there are several different types of devices:
• Simple-fixed format
• Statistical (statistical time division multiplexers [STDMs] or statistical multiplexers)

• Drop and insert
In a simple fixed-format multiplexer, the relationship between the input and the output is fixed. In a more
sophisticated multiplexer, it may be varied.
Statistical time division multiplexers (STDMs) contain an added microprocessor that provides intelligent
data 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 link
to inputting devices on an as-needed basis, rather than in a fixed-dedicated basis. Therefore, rather than
wasting bandwidth when the inputting device is idle, the bandwidth is utilized to serve active devices (that
is, 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 buffers
for holding data from attached devices, which enable them to handle a combined input speed (aggregate
speed) that exceeds the speed of the communications link.
Drop-and-insert (D/I) multiplexers are commonly used in private networks and in the dedicated facilities
portion of public networks. D/I muxes are used to remove one or more of the channels from a multiplexed
transmission 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.
Protocol Converters
As mentioned earlier in the report, not all data communications devices speak the same language.
Network designers can achieve flexibility and economic rewards by using products from more than one
vendor, and equipment manufacturers have responded by developing products that overcome language
incompatibilities. Protocol converters and emulators can be hardware-based, software-based or a
combination of both, and can range from a microprocessor-based circuit board to a front-end processor
with the capability to handle conversion functions. Available conversion devices might handle one or
many types of conversions. For example, some devices handle only code or interface conversions (Layer
One of the ISO model), while others handle protocol conversion, device emulation, and code and
interface conversions.
A protocol converter actually changes one protocol to another by stripping down the data from one device
and re-wrapping 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 control
characters, reformats the block and calculates the required check characters so that the receiving device
gets characters formatted according to its requirements.
DPRO-89990
For example, in an ASCII-to-SDLC conversion, the converter accepts a string of characters, eliminates
start and stop bits, assembles the characters into a block, and adds appropriate headers and trailers to
create complete frames. All protocol converters have some intermediate storage area to hold characters
for conversion; this buffering extends the response time in the communications exchange.
PCs
The major end-user device in the network is the PC, and it has gained nearly universal acceptance as a
tool to perform local data analysis. It is only natural that a user would want to communicate with a network
host via such an intuitive device.
PCs and data terminals differ in one major respect: the PC is an intelligent device, capable of
manipulating 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 an
interface between a human operator and a host computer. Even “intelligent” terminals—those that can
perform 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 management
software systems blur. Computer manufacturers are incorporating telemanagement functionality into
network management systems. Telemanagement software vendors, on the other hand, are incorporating
network management capabilities, which are generally voice-oriented and designed to provide PBX
management interfaces. The PBX market is the focal point for the convergence of the computing and
telecommunications approaches to network management. Most major PBX vendors now have a data
networking strategy as well as a voice strategy. See Figure “Data and Voice Network Convergence.”
Figure 17: Data and Voice Network Convergence
The Siemens Hicom 150 H exemplifies the trend toward PBXs that can perform IP-packet switching, as well as
TDM-circuit switching. Local users (1) connect directly to the IP packet-switched infrastructure, while those at (2), (3)
and (4) connect through external networks. Users have on-demand voice connectivity with each other. IP phones
connect to the IP infrastructure and are controlled by the PBX.
DPRO-89990
Source: Siemens ICN.
Telemanagement Systems and Software
The growing importance of telecommunications networks within companies means that management
tools are essential. Telecommunications management networks are increasingly recognized as strategic
assets that can increase customer satisfaction, build customer loyalty and develop new business.
Telecommunications management systems not only monitor staff activity and customer response, but
also help the telecommunications manager optimize the use of the network and identify problems as they
arise.
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. A
natural evolution of service bureau software was the development of licensed software systems for a
client’s computer system. Changes in computer and software technology added significant options to
computing alternatives during the mid-1980s, when PCs became increasingly popular. During the late
1980s, LANs became even more popular. Today, software developers offer telemanagement solutions for
many 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 inventories and cable and wire resources. Process
management automates a number of processes—traffic analysis, network design and optimization,
directory management, work order and trouble management—which enable the effective management of
a telecommunications network.
Communications Software
An efficient method of controlling data communications networks utilizes combinations of hardware and
software for control purposes. Repetitive tasks that rarely change are best implemented as hardware
modules, while dynamic tasks such as the maintenance of terminal specifications are best implemented
through 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 its
maintenance 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-distance
computer 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.
DPRO-89990
Communications software focuses on a number of objectives: communications between various devices,

communications between end users (or between end users and applications), communications between
databases, communications between applications, and management and control of communications
activities. While hardware generally supports line, device and presentation functions, software supports
the following:
• Executive functions that control the sequencing of tasks.
• Directive functions that make decisions about how, when, which and how much of the system
resources should be committed to a given task.
• Quality-assurance functions that monitor what is happening in the network and provide fallback
resources in the event of system failure.
• Intersystem functions that act as interfaces between the various layers of data communications
software and hardware.
Recent development efforts have advanced communications software toward the interaction of end-user
programs on a peer-to-peer basis. This bilateral interaction makes the diverse connection schemes and
host servers typically found in large corporations completely transparent to end users. It also
demonstrates 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’s
ongoing functionality. They must seek out appropriate solutions for network redundancy, for
troubleshooting and verification, and for network management. Choosing the appropriate equipment for
network testing, monitoring and control allows users to carry out that responsibility. As the complexity of
the 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. Network
management 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 get
their communications under their own control.
Figure 18: Network Management Functions
From a technology standpoint, network management can be depicted as the intersection of seven different
functional areas.
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A minimal network management system consists of:

• A CPU
• A hard disk or diskette storage device
• An operator’s console
• A set of local and remote monitoring devices
All network management systems include mechanisms for monitoring the network’s components. When
the network management system vendor also manufactures modems, the vendor usually designs the
monitoring device as a built-in modem feature, eliminating the need for the user to acquire separate
monitoring devices. On other systems, stand-alone monitoring devices must be attached to modems or
multiplexers at each remote site.
In most network management systems, monitoring devices examine only the status of the modem or
multiplexer, its interface with the equipment, its interface with the transmission facility and the condition of
the transmission facility. Information on the modem or multiplexer and its interfaces comes from the
presence or absence of signals on various EIA interface leads. Information on the transmission facility
comes from the measurement of various analog parameters, such as signal level, noise, distortion, phase
jitter and line hits. If a given interface signal or analog characteristic falls out of specification, the system’s
monitors will set off an alarm to notify the operator of a failure.
Some network management systems can switch automatically from a failing component to a “hot” standby
unit either on receipt of an alarm or on command from the operator. Some systems can also bypass a
failed communications line by a call placed automatically over the switched-voice network. Such
automatic dial backup procedures require two switched-network calls for full-duplex operation.
As networks become increasingly sophisticated, network management systems grow in complexity. One
of the more interesting recent developments is the incorporation of expert systems techniques into
network management. An expert system—software that contains rules for making logical inferences—can
add a degree of intelligence to a network management system. Such an intelligent system can do more
than isolate faults; it can also suggest to an operator the possible causes of those faults, test hypotheses
about them and propose courses of action to remedy them.
Another recent development is the use of multiple network management systems within a complete
corporate network. Communications networks today may be built from several different types of dissimilar
equipment and may incorporate several smaller networks of different types. Full network management
may require interconnecting management information from all of these networks into a usable form. One
of the frontiers of network management is the construction of systems that can bring together this
dissimilar information and integrate it through decision support or executive support systems.
DPRO-89990
Technology Leaders
Listed below are some of the leading vendors in both the data and voice communications industries for
equipment and services:
• 3Com
• Alcatel
• AT&T
• Avaya
• Cisco Systems
• Ericsson
• Hewlett-Packard
• IBM
• Lucent Technologies
• MCI WorldCom
• Motorola
• NEC
• Nokia
• Nortel Networks
• Siemens
• Sprint
Insight
Movement continues today toward the integration of enterprise voice and data communications networks,
spurred-on largely by the interest in transporting VoIP-based data networks. Justifications for converging
onto a single infrastructure have progressed beyond toll bypass applications to projections of lower total
cost of ownership (TCO), more efficient system management and administration, and enhanced
applications use. But QOS issues and lack of consensus on standards continue to be major inhibitors,
along with a perceived shortage of personnel with expertise in both areas. In addition, with shaky
economic conditions still existing throughout the globe, enterprises are more carefully scrutinizing any
modifications to current infrastructures.
DPRO-89990

Basic Concepts of Communications-An Introduction

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DPRO-89990 Technology Overview

Richard Costello 24 October 2002

Basic Concepts of Communications: An Introduction

Figure 8: Pulse Code Modulation (PCM)

Telecommunications systems consist of two main elements—transmission and switching—and may be

Prerequisites for Communications

Open Systems Interconnection

Figure 1: The OSI Reference Model

As previously noted, information travels between machines in patterns of 1s and 0s:

• Data representation and coding

Figure 2: High-Level Data Link Control (HDLC) Framing

Figure 3: Types of Data Transmission

Figure 4: Start and Stop Bits

Synchronous transmission sends a block of characters over a communications link in a continuous

Simplex, Half-Duplex and Full-Duplex

Three modes of data transmission are available:

Figure 5: Transmission Methods

Transmission Signal Attributes

Figure 6: Transmission Signal Attributes

The bandwidth of a communications channel determines a channel’s information-carrying capacity.

Data Transmission Rates

Frequency Division Multiplexing (FDM)

Figure 7: Multiplexing Techniques

Time Division Multiplexing (TDM)

Pulse Code Modulation

Figure 8: Pulse Code Modulation (PCM)

Digital Facilities (T1 and E1)

Digital Signal Processors (DSPs)

Special-purpose microprocessors with instruction sets designed to manipulate digital signals—termed

Public Switched Telephone Network

Nonswitched (Leased) Facilities

Internet telephony refers to communications services—voice, facsimile and voice-messaging

Figure 10: Voice on IP Networks

Phone to Phone on the Internet

PBX to PBX on an Intranet

Packet Telephony Pitfall

Very Small Aperture Terminals (VSATs)

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

Integrated Services Digital Network (ISDN)

Figure 12: Mobile Communications Network

Handoff to the Same MSC

Handoff to a Different MSC

Wireless Communication Standards

• Private Mobile Radio (PMR) and Specialized Mobile Radio (SMR)

• Analog/digital cellular phone systems (AMPS/D-AMPS)

• Personal Handy-Phone System (PHS)

• Digital Short Range Radio (DSSR)

Wireless Data Communications

Broadband ISDN (B-ISDN)

Asynchronous Transfer Mode (ATM)

Source: Nortel Networks.

• Discard incorrect frames

• Fill interframe time

Frame Relay, X.25 and TCP/IP Compared

Digital Subscriber Line (DSL)

Synchronous Digital Hierarchy (SDH)

• Reduction in network equipment

• Increased network flexibility

• Savings from reduction in maintenance and operations costs

• Network software controllable

Fiber Distributed Data Interface (FDDI)

Figure 14: Cabling Types

Accuracy Controls/Error Correction

• Cyclic redundancy checking (CRC)