Introduction to Data Communications

In Data Communications, data generally are defined as information that is stored in digital
form. Data communications is the process of transferring digital information between two or
more points.
Information is defined as the knowledge or intelligence. Data communications can be
summarized as the transmission, reception, and processing of digital information. For data
communications to occur, the communicating devices must be part of a communication
systemmade up of a combination of hardware (physical equipment) and software (programs).
The effectiveness of a data communications system depends on four fundamental
characteristics: delivery, accuracy, timeliness, and jitter.

A data communications system has five components:

1. Message: The message is the information (data) to be communicated. Popular forms of
information include text, numbers, pictures, audio, and video.
2. Sender: The sender is the device that sends the data message. It can be a computer,
workstation, telephone handset, video camera, and so on.
3. Receiver: The receiver is the device that receives the message. It can be a computer,
workstation, telephone handset, television, and so on.
4. Transmission medium: The transmission medium is the physical path by which a message
travels from sender to receiver. Some examples of transmission media include twisted -pair
wire, coaxial cable, fiber-optic cable, and radio waves.
5. Protocol: A protocol is a set of rules that govern data communications. It represents an
agreement between the communicating devices.

Standards Organizations for Data Communications

An association of organizations, governments, manufacturers and users form the
standards organizations and are responsible for developing, coordinating and maintaining the
standards. The intent is that all data communications equipment manufacturers and users
comply with these standards. The primary standards organizations for data communication
are:
1. International Standard Organization (ISO)
ISO is the international organization for standardization on a wide range of subjects. It is
comprised mainly of members from the standards committee of various governments
throughout the world. It is even responsible for developing models which provides high
level of system compatibility, quality enhancement, improved productivity and reduced
costs. The ISO is also responsible for endorsing and coordinating the work of the other
standards organizations.
2. International Telecommunications Union-Telecommunication Sector
(ITU-T)
ITU-T is one of the four permanent parts of the International Telecommunications Union
based in Geneva, Switzerland. It has developed three sets of specifications: the V series for
modem interfacing and data transmission over telephone lines, the X series for data

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transmission over public digital networks, email and directory services; the I and Q series
for Integrated Services Digital Network (ISDN) and its extension Broadband ISDN. ITU-T
membership consists of government authorities and representatives from many
countries and it is the present standards organization for the United Nations.
3. Institute of Electrical and Electronics Engineers (IEEE)
IEEE is an international professional organization founded in United States and is
compromised of electronics, computer and communications engineers. It is currently the
world‘s largest professional society with over 200,000 members. It develops
communication and information processing standards with the underlying goal of
advancing theory, creativity, and product quality in any field related to electrical
engineering.
4. American National Standards Institute (ANSI)
ANSI is the official standards agency for the United States and is the U.S voting
representative for the ISO. ANSI is a completely private, non-profit organization
comprised of equipment manufacturers and users of data processing equipment and
services. ANSI membership is comprised of people form professional societies, industry
associations, governmental and regulatory bodies, and consumer goods.
5. Electronics Industry Association (EIA)
EIA is a non-profit U.S. trade association that establishes and recommends industrial
standards. EIA activities include standards development, increasing public awareness,
and lobbying and it is responsible for developing the RS (recommended standard) series of
standards for data and communications.
6. Telecommunications Industry Association (TIA)
TIA is the leading trade association in the communications and information technology
industry. It facilitates business development opportunities through market
development, trade promotion, trade shows, and standards development. It represents
manufacturers of communications and information technology products and also
facilitates the convergence of new communications networks.
7. Internet Architecture Board (IAB)
IAB earlier known as Internet Activities Board is a committee created by ARPA
(Advanced Research Projects Agency) so as to analyze the activities of ARPANET whose
purpose is to accelerate the advancement of technologies useful for U.S military. IAB is a
technical advisory group of the Internet Society and its responsibilities are:
I. Oversees the architecture protocols and procedures used by the Internet.
II. Manages the processes used to create Internet Standards and also serves as an
appeal board for complaints regarding improper execution of standardization
process.
III. Responsible for administration of the various Internet assigned numbers
IV. Acts as a representative for Internet Society interest in liaison relationships with
other organizations.
V. Acts as a source of advice and guidance to the board of trustees and officers of
Internet Society concerning various aspects of internet and its technologies.
8. Internet Engineering Task Force (IETF)
The IETF is a large international community of network designers, operators, vendors
and researchers concerned with the evolution of the Internet architecture and smooth
operation of the Internet.
9. Internet Research Task Force (IRTF)
The IRTF promotes research of importance to the evolution of the future Internet by
creating focused, long-term and small research groups working on topics related to
Internet protocols, applications, architecture and technology.

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1.2: A Communication Model
The underlying purpose of a digital communications circuit is to provide a transmission path
between locations and to transfer digital information from one station (node, where computers
or other digital equipment are located) to another using electronic circuits. Data
communications circuits utilize electronic communications equipment and facilities to
interconnect digital computer equipment. Communication facilities are physical means of
interconnecting stations and are provided to data communications users through public
telephone networks (PTN), public data networks (PDN), and a multitude of private data
communications systems.
The following figure shows a simple two-station data communications circuit. The
main components are:
Source: - This device generates the data to be transmitted; examples are mainframe
computer, personal computer, workstation etc. The source equipment provides a means for
humans to enter data into system.
Transmitter: - A transmitter transforms and encodes the information in such a way as to
produce electromagnetic signals that can be transmitted across some sort of transmission
system. For example, a modem takes a digital bit stream from an attached device such as a
personal computer and transforms that bit stream into an analog signal that can be handled by
the telephone network.
Transmission medium: - The transmission medium carries the encoded signals from the
transmitter to the receiver. Different types of transmission media include free-space radio
transmission (i.e. all forms of wireless transmission) and physical facilities such as metallic
and optical fiber cables.
Receiver: - The receiver accepts the signal from the transmission medium and converts it
into a form that can be handled by the destination device. For example, a modem will
accept an analog signal coming from a network or transmission line and convert it into a
digital bit stream.
Destination: - Takes the incoming data from the receiver and can be any kind of digital
equipment like the source

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Data Communication Circuit Arrangements
A data communications circuit can be described in terms of circuit configuration and
transmission mode.
Circuit Configurations
Data communications networks can be generally categorized as either two point or
multipoint. A two-point configuration involves only two locations or stations, whereas a
multipoint configuration involves three or more stations.

A two-point circuit involves the transfer of digital information between a mainframe

computer and a personal computer, two mainframe computers or two data
communications networks. A multi-point network is generally used to interconnect a single
mainframe computer (host) to many personal computers or to interconnect many personal
computers and capacity of the channel is either Spatially shared: Devices can use the link
simultaneously or Timeshare: Users take turns
Transmission Modes
There are four modes of transmission for data communications circuits:

In simplex mode(SX), the communication is unidirectional, as on a one-way street. Only one

of the two devices on a link can transmit; the other can only receive. Commercial radio
broadcasting is an example. Simplex lines are also called receive-only, transmit-only or
oneway- only lines.
In half-duplex(HDX) mode, each station can both transmit and receive, but not at
the same time. When one device is sending, the other can only receive, and vice versa. The

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half-duplex mode is used in cases where there is no need for communication in both
directions at the same time; the entire capacity of the channel can be utilized for each
direction. Citizens band (CB) radio is an example where push to talk (PTT) is to be pressed or
depressed while sending and transmitting.
In full-duplex mode(FDX) (called duplex), both stations can transmit and receive
simultaneously. One common example of full-duplex communication is the telephone
network. The full-duplex mode is used when communication in both directions is required all
the time. The capacity of the channel must be divided between the two directions.
In full/full duplex (F/FDX) mode, transmission is possible in both directions at the
same time but not between the same two stations (i.e. station 1 transmitting to station 2,
while receiving from station 3). F/FDX is possible only on multipoint circuits. Postal system
can be given as a person can be sending a letter to one address and receive a letter from
another address at the same time.
Data Communications Networks
Any group of computers connected together can be called a data communications network,
and the process of sharing resources between computers over a data communications
network is called networking. The most important considerations of a data communications
network are performance, transmission rate, reliability and security.
Network Components, Functions, and Features
The major components of a network are end stations, applications and a network that will
support traffic between the end stations. Computer networks all share common devices,
functions, and features, including servers, clients, transmission media, shared data, shared
printers and other peripherals, hardware and software resources, network interface card
(NIC), local operating system (LOS) and the network operating system (NOS).
Servers: Servers are computers that hold shared files, programs and the network operating
system. Servers provide access to network resources to all the users of the network and
different kinds of servers are present. Examples include file servers, print servers, mail
servers, communication servers etc.
Clients: Clients are computers that access and use the network and shared network
resources. Client computers are basically the customers (users) of the network, as they
request and receive service from the servers.
Shared Data: Shared data are data that file servers provide to clients, such as data files,
printer access programs, and e-mail.

Shared Printers and other peripherals: these are hardware resources provided to the users
of the network by servers. Resources provided include data files, printers, software, or any
other items used by the clients on the network.
Network interface card: Every computer in the network has a special expansion card called
network interface card (NIS), which prepares and sends data, receives data, and controls
data flow between the computer and the network. While transmitting, NIC passes frames of
data on to the physical layer and on the receiver side, the NIC processes bits received from the
physical layer and processes the message based on its contents.
Local operating system: A local operating system allows personal computers to access files,
print to a local printer, and have and use one or more disk and CD drives that are located on
the computer. Examples are MS-DOS, PC-DOS, UNIX, Macintosh, OS/2, Windows 95, 98,
XP and Linux.
Network operating system: the NOS is a program that runs on computers and servers that
allows the computers to communicate over a network. The NOS provides services to clients
such as log-in features, password authentication, printer access, network administration
functions and data file sharing.

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2.1: Signals, Noise, Modulation and Demodulation
Computers transmit data using digital signals, sequences of specified voltage levels.
Computers sometimes communicate over telephone lines using analog signals, which are
formed by continuously varying voltage levels. Electrical signals can be in analog or digital
form. With analog signals, the amplitude changes continuously with respect to time with no
breaks or discontinuities. A sine wave is the most basic analog signal.
Digital signals are described as discrete; their amplitude maintains a constant level for
a prescribed period of time and then it changes to another level. If only two levels are
possible, it is called a binary signal. All binary signals are digital, but all digital signals are not
necessarily binary. Converting information signals to a different form is called modulation
and the reverse process is called demodulation. The modulating signal is the information and
the signal being modulated is the carrier.
Two basic types of electronic communications systems are analog and digital. An
analog digital communications system is a communications system in which energy is
transmitted and received in analog form and are also propagated through the system in
analog form. Digital communications covers a broad range of communications techniques
including digital transmission and digital modulation.
Signal Analysis
Mathematical signal analysis is used to analyze and predict the performance of the circuit on
the basis of the voltage distribution and frequency composition of the information signal.
Amplitude, Frequency and Phase
A cycle is one complete variation in the signal, and the period is the time the waveform
takes to complete on cycle. One cycle constitutes 360 degrees (or 2π radians). Sine waves can
be described in terms of three parameters: amplitude, frequency and phase.
Amplitude (A): It is analogous to magnitude or displacement. The amplitude of a signal is the
magnitude of the signal at any point on the waveform. The amplitude of electrical signal is
generally measured in voltage. The maximum voltage of a signal in respect to its average
value is called its peak amplitude or peak voltage.

Frequency (f):
The time of one cycle of a waveform is its period, which is measured in
seconds. Frequency is the number of cycles completed per second. The measurement unit
for frequency is the hertz, Hz. 1 Hz = 1 cycle per second. The frequency of the signal can be
calculated fromT=1/f

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Phase (Ø): The phase of the signal is measured in degrees or radians with respect to a
reference point. A phase shift of 180 degrees corresponds to a shift of half a cycle.

A phase shift of 360 degrees corresponds to a shift of one complete cycle. If two sine
waves have the same frequency and occur at the same time, they are said to be in phase,
or they are said to out of phase. The difference in phase can be measured in degrees, and is
called the phase angle,Ɵ

Periodic Signals
A signal is periodic if it completes a pattern within a measurable time and is characterized by
amplitude, frequency and phase. Mathematically, a single frequency voltage waveform is
v(t) = V sin(2πft + θ), where
v(t) is time-varying voltage sine wave
V is peak amplitude in volts
f is frequency in hertz
t is time in seconds
θ is phase in degrees or radians

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It is called a periodic wave because, it repeats at a uniform rate. A series of sine, cosine or
square waves constitute an example of periodic waves, which can be analyzed in either the
time domain or the frequency domain.
Time domain: Time domain is a term used to describe the analysis of mathematical
functions, or physical signals, with respect to time. In the time domain, the signal or
function's value is known for all real numbers, for the case of continuous time, or at various
separate instants in the case of discrete time. An oscilloscope is a time-domain tool
commonly used to visualize real-world signals in the time domain. A time domain graph
shows how a signal changes over time.

Frequency Domain: frequency domain is a term used to describe the analysis of mathematical
functions or signals with respect to frequency, rather than time. A spectrum analyser is a
frequencydomain instrument which displays amplitude-versus frequency plot (called a frequency
spectrum). The horizontal axis represents frequency and the vertical axis amplitude showing a
vertical deflection for each frequency present in the waveform, which is proportional to the
amplitude of the frequency it represents.

Fourier series:
The Fourier series is used in signal analysis to represent the sinusoidal
components of nonsinusoidal periodic waveforms. A Fourier series decomposes a periodic
function or periodic signal into a sum of simple oscillating functions, namely sines and cosines. It can
be
expressed as:
f(t) = A0 + A1 cosα + A2 cos2α + A3 cos3α + ... .... .An cos nα
+ A0 + B1 sinβ + B2 sin2β + B3 sin3β +... .... ...Bn sin nβ
Where α=β

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Any periodic waveform is comprised of an average dc component and a series of
harmonically related sines or cosine waves. A harmonic is an integral multiple of the
fundamental frequency.
frequency (repetition rate) of the waveform. Second multiple is called second harmonic,
third multiple is called third harmonic and so forth.
Wave symmetry: It describes the symmetry of a waveform in the time domain, i.e., its
relative position with respect to the horizontal (time) and vertical (amplitude) axes.
Even symmetry:
If a periodic voltage waveform is symmetric about the vertical axis, it is said to
have axes, or mirror, symmetry and is called an even function. For all even functions, the β
coefficients are zero. Even function satisfy the condition f(t) = f(-t)

Odd symmetry: If a periodic voltage waveform is symmetric about a line midway between
the vertical axis and the negative horizontal axis and passing through the coordinate origin, it
is said to have to point or skew, symmetry and is called an odd function. For all odd
functions, the α coefficients are zero. Odd function satisfies f(t) = -f(-t)

Half-wave symmetry: If a periodic voltage waveform is such that the waveform for the first
half cycle repeats itself except with the opposite sign for the second half cycle, it is called to
have half-wave symmetry. Half-wave symmetry implies that the second half of the wave is
exactly opposite to the first half. A function with half-wave symmetry does not have to be
even or odd, as this property requires only that the shifted signal is opposite. Half-wave
functions satisfy the condition f(t) = -f(T+t)/2

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Frequency Spectrum and Bandwidth
The frequency spectrum of a waveform consists of all the frequencies contained in the
waveform and their respective amplitudes plotted in the frequency domain.
Bandwidth of an information signal is simply the difference between the highest and lowest
frequencies contained in the information and the bandwidth of a communication channel is
the difference between the highest and lowest frequencies that the channel will allow to
pass through it.
Electrical Noise and Signal-To-Noise Ratio
Noise is any disturbance or distortion that comes in the process of communication . Electrical
noise is defined as any undesirable electrical energy that falls within the passband of the
signal. A noise signal consists of a mixture of frequencies with random amplitudes. Noise
can originate in various ways. The most prevalent and most interfering to data
communication signals are man-made noise, thermal noise, correlated noise, and
impulse noise.
Man-made noise: It is the kind of noise produced by mankind. The main sources are
sparkproducing mechanisms like commutators in electric motors, automobile ignition
systems,
ac power-generating and switching equipment, and fluorescent lights. It is impulsive in
nature
and contains a wide range of frequencies propagated in the free space like the radio waves.
Man-made noise is most intense in more densely populated areas and sometimes is
referred to as industrial noise.
Thermal noise: This is the noise generated by thermal agitation of electrons in a
conductor. It is also referred to as white noise because of its uniform distribution across the
entire electromagnetic frequency spectrum. Noise power density is the thermal noise power
present in a 1-Hz bandwidth and is given by No = KT.
Thermal noise is independent of frequency and thus thermal noise present in any
bandwidth is N = KTB where N is thermal noise power in watts, K is Boltzmann's
constant in joules per Kelvin, T is the conductor temperature in kelvin (0K = -273oC), and B is
the bandwidth in hertz. Noise power is often measured in dBm. From the equation above,
noise power in a resistor at room temperature, in dBm, is: NdBm = -174 dBm + 10 log B
Correlated noise: this noise is correlated to the signal and cannot be present in a circuit
unless there is a signal. Correlated noise is produced by nonlinear amplification and includes
harmonic distortion and intermodulation distortion. Harmonic distortion occurs when
unwanted harmonics of a signal are produced through nonlinear amplification and is also
called amplitude distortion. Intermodulation distortion is the generation of unwanted sum
and difference frequencies produced when two or more signals are amplified in a nonlinear
device.
Impulse noise: This noise is characterized by high-amplitude peaks of short duration in the
total noise spectrum. It consists of sudden bursts of irregularly shaped pulses that generally
last between a few microseconds and several milliseconds, depending on their amplitude
and origin. In case of voice communications, impulse noise is very annoying as it generates a
sharp popping or crackling sound where as it is devastating in data circuits.
Signal-to-noise power ratio: Signal-to-noise ratio (often abbreviated SNR or S/N) is
defined as the ratio of signal power to the noise power corrupting the signal. A ratio higher
than 1:1 indicates more signal than noise. Signal-to-noise ratio is defined as the power ratio
between signal (meaningful information) and the background noise (unwanted signal)

10
where P is average power in watts. The ratio often expressed in
decibels as S/N (dBm) = 10 log(PS/PN)
Attenuation:
The strength of a signal falls off with diatance over any transmission medium.For
guided medium , this reduction in strength is exponential ; V2= V1-
where = loss per unit length , and x= distance
Delay Distortion :
Caused when velocity of propagation of a signal in a medium varies with frequency.
For a band limited signal, the velocity tends to be highest near the centre frequency
and falls off towards the edges of the band. Thus various frequency components of a
signal will arrive at the receiver at different times- resulting in phase shifts between
the different frequencies.

The Radio Spectrum and its Utilization.

Due to the difference in propagation characteristics, each band of frequencies has its
own scope of application. The most frequently used ranges of the radio frequency
spectrum are divided into the bands shown in Table 1.
Table 1.
Designation Abbreviation Frequency Range Services supported.
Very Low Frequency VLF Under 30KHZ Marine Radio
Low Frequency LF 30KHZ-300KHZ Trans-oceanic
communication
Medium Frequency MF 300KHZ-3MHZ AM Radio
broadcasts,
High Frequency HF 3MHZ-30MHZ International
broadcast (SW),
amateur radio, ship-
shore radio.
Very High Frequency VHF 30MHZ-300MHZ Air navigation, TV
broadcasts, FM
radio, Meteorology,
Telephone public
carriers.
Ultra High Frequency UHF 300MHZ-3GHZ TVm broadcasts,
Mobile
telephony,Satellite
Super High Frequency SHF 3GHZ-30GHZ Satellite
Extremely High EHF 30GHZ-300GHZ Experimental
Frequency Satellite.

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Analog Modulation Systems
A sine wave has three main components: amplitude, frequency and phase and can be
expressed as v(t) = V sin(2πft + θ). If the information signal is analog and the amplitude
(V) of the carrier is varied proportional to the informational signal, amplitude modulation
(AM) is produced. If the frequency (f) is varied proportional to the information signal,
frequency modulation (FM) is produced and if the phase (θ) is varied proportional to the
information signal, phase modulation (PM) is produced. Frequency and phase modulation
are similar and often combined and are simply called angle modulation.

The process of impressing relatively low-frequency information signals onto a highfrequency

carrier signal is called modulation and the reverse process is called demodulation.
Analog modulation is used for the transmission of conventional analog signals, such as
voice, music, and video and not particularly useful for data communication systems.

Amplitude Modulation
Amplitude modulation is the process of changing the amplitude of a relatively high
frequency carrier signal in proportion to the instantaneous value of the modulating signal
(information). AM modulators are two-input devices, one of them is a single, relatively high
frequency carrier signal of constant amplitude and the second is the relatively
lowfrequency information signal. The following figure shows generation of AM waveform
when a single-frequency modulating signal acts on a high frequency carrier signal.

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Advantages of AM are simple to implement, needs a circuit with very few components and
inexpensive. The disadvantages include inefficient power usage and use of bandwidth and
also prone to noise. The total bandwidth required for AM can be determined from the
bandwidth of the audio signal: B AM = 2B
Angle Modulation
Angle modulation results whenever the phase angle of a sinusoidal signal is varied with
respect to time and includes both FM and PM. Whenever the frequency of a carrier signal is
varied, the phase is also varied and vice versa. If the frequency of the carrier is varied
directly in accordance with the information signal, FM results, whereas if the phase is varied
directly, PM results.
The above figure shows the FM and PM of a sinusoidal carrier by a single-frequency
modulating signal. Both FM and PM waveforms are identical except for their time
relationship (phase). With FM, the maximum frequency deviation occurs during the
maximum positive and negative peaks of the modulating signal. With FM, the maximum
frequency deviation occurs during the zero crossings in the modulating signal.

An important feature of FM and PM is that they can provide much better protection to the
message against channel noise when compared to AM. Also because of their constant
amplitude nature, they can withstand nonlinear distortion and amplitude fading.
Information Capacity, Bits, Bit Rate, Baud, and M-ARY Encoding
Information capacity is a measure of how much information can be propagated
through a communication system and a function of bandwidth and transmission time. It
represents the number of independent symbols that can be carried through a system in a

13
given unit of time. The most basic digital symbol used to represent information is the binary
digit, or bit. Bit rate is simply the number of bits transmitted during 1 second and is
expressed as bits per second (bps).
R.Hartley developed a useful relationship among bandwidth, transmission time and
information capacity called Hartley’s law given by:
I α B × t Where, I is the information capacity in bps, B is bandwidth in hertz and t is
transmission time in sec’s
Relation between information capacity of a communication channel to a bandwidth
and signal-to-noise ratio is given by Claude E. Shannon. The higher the signal-to-noise ratio,
the
better the performance and also information capacity is higher. The Shannon limit of
information capacity is
I = B log2 (1 + S/N ) or I = 3.32 B log10 (1 + S/N )
Where I is information capacity in bps, B is bandwidth in hertz and S/N is signal to noise
ratio.
M-ary Encoding
M-ary is a term derived from the word binary. M simply represents a digit that
corresponds to the number of conditions, levels, or combinations possible for a given
number of binary variables. For example, a digital signal with four possible conditions is an
Mary
system where M= 4 and if there are eight possible conditions, then M= 8. The number of bits
necessary to produce a given number of conditions is expressed mathematically as:
N = log2 M or it can be written as M = 2N, where N is no of bits necessary and M is number
of
conditions, levels or combinations possible with N bits. From the equation, it can be said that
if there is one bit, only 21 or two conditions are possible. For two bits 22 or four
conditions are possible.
Baud and Minimum Bandwidth
Baud, like bit rate is a rate of change. Baud refers to the rate of change of the signal on
the transmission medium after encoding and modulation have occurred. Baud is the
reciprocal of the time of one output signalling element, and a signalling element may
represent several information bits. Baud is also transmitted one at a time and a baud may
represent more than one information bit. So, the baud of the data communications system
may be considerably less than the bit rate.
According to H.Nyquist, binary digital signals can be propagated through an ideal
noiseless medium at a rate equal to twice the bandwidth of the medium. The minimum
theoretical bandwidth necessary to propagate a signal is called the minimum Nyquist
bandwidth or sometimes the Nyquist bandwidth. Using multilevel signalling, the Nyquist
formulation for channel capacity is fb = B log2 M where, fb is channel capacity in bps, B is
minimum Nyquist bandwidth in hertz and M is no of discrete signal or voltage levels . If N is
substituted, we get
B = baud = fb/N, where N is number of bits encoded into each signalling element.
Digital Modulation
Digital modulation is the transmission of digitally modulated analog signals between
two or more points in a communications system. Analog and Digital modulation systems use
analog carriers to transport information through the system, but digital modulation uses
digital modulating (information) signal. Analog systems use analog signal only. In, v(t) = V

14
sin(2πft + θ), if the information signal is digital and amplitude (V) of the carrier is varied
proportional to the information signal, a digitally modulated signal called amplitude -shift
keying (ASK) is produced. If the frequency (f) is varied proportional to the information signal,
frequency-shift keying (FSK) is produced and if the phase is varied proportional to the
information signal, phase-shift keying (PSK) is produced. If both amplitude and phase are
varied proportional to the information signal, quadrature amplitude modulation (QAM)
results.

Bit rate is the number of bits per second.

•Baud rate is the number of signal units per
second.
•Baud rate is less than or equal to the bit rate.
Baud rate= Bit rate / nr of bits per signal unit
1. An analog signal carries 4 bits in each signal unit.
If 1000 signal units are sent per second, find the baud
rate and the bit rate
Baud rate = 1000 bauds per second (baud/s)
Bit rate = 1000 x 4 = 4000 bps
2. The bit rate of a signal is 3000.
If each signal unit carries 6 bits, what is the baud rate?
Baud rate = 3000 / 6 = 500 baud/s

Digital modulation is ideally suited to a multitude of communications applications including

both cable and wireless systems. Applications include relatively low-speed voice-band data
communications systems, high-speed data transmission systems, digital satellite
communication systems and personal communication systems (PCS).

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Amplitude-Shift Keying
It is the simplest digital modulation technique where a binary information signal directly
modulates the amplitude of an analog carrier. Only two output amplitudes are possible and
ASK is sometimes called as digital amplitude modulation (DAM). Amplitude shift keying is
given in mathematical terms as:
vask(t) = [ 1 + vm(t) ][ A/2 cos(ωct)]
Where vask(t) is amplitude-shift keying wave, vm(t) is digital modulation (modulating) signal
in volts, A/2 is unmodulated carrier amplitude in volts and ωc is analog carrier radian
frequency in radians per second.

In the above equation, for the modulating signal vm(t), logic 1 is represented by +1V
and logic 0 is represented by -1V. So the modulated wave vask(t) is either Acos(ωct) or 0
i.e., the carrier is either on or off. ASK is sometimes referred as on-off keying (OOK). The rate
of change of the ASK waveform (baud) is the same as the rate of change of the binary input
making bit rate equal to baud. With ASK, the bit rate is also equal to the minimum Nyquist
bandwidth.

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BW=(1+d ) x Nbaud
Where:
BW is the bandwidth;
Nbaud is the baud rate
fc=carrier frequency
d is a factor related to the modulation process

Example
Find the minimum bandwidth for an ASK signal
transmitting at 2000 bps.
The transmission mode is half-duplex.

17
Solution
In ASK the baud rate and bit rate are the same.
The baud rate is therefore 2000. An ASK signal requires
a minimum bandwidth equal to its baud rate.
Therefore, the minimum bandwidth is 2000 Hz.
Example
Given a bandwidth of 5000 Hz for an ASK signal, what
are the baud rate and bit rate?
Solution
In ASK the baud rate is the same as the bandwidth,
which means the baud rate is 5000.
But because the baud rate and the bit rate are also the
same for ASK, the bit rate is 5000 bps.
Example
Given a bandwidth of 10,000 Hz (1000 to 11,000 Hz), draw the full-duplex ASK
diagram of the system. Find the carriers and the bandwidths in each direction.
Assume there is no gap between the bands in the two directions.
For full-duplex ASK, the bandwidth for each direction is
BW = 10000 / 2 = 5000 Hz
The carrier frequencies can be chosen at the middle of each band
fc (forward) = 1000 + 5000/2 = 3500 Hz
fc (backward) = 11000 – 5000/2 = 8500 Hz

Frequency Shift Keying

FSK is another simple, low-performance type of digital modulation. It is similar to FM, except
the modulating signal is a binary signal varying between two discrete voltage levels. FSK is
sometimes called as binary FSK (BFSK). FSK is generally expressed as
vfsk(t) = Vc cos{ 2π[fc + vm(t)Δf]t }
Where vfsk(t) is binary FSK waveform, Vc is peak analog carrier amplitude in volts, fc is
analog carrier center frequency in hertz, Δf is peak change or shift in the analog carrier
frequency and vm(t) is binary input(modulating) signal in volts. For logic 1, vm(t) = +1 and
for logic 0, vm(t) = -1 reducing the equation to vfsk(t) = Vc cos{ 2π[fc + Δf]t } and vfsk(t) =
Vc cos{ 2π[fc - Δf]t }
As the binary signal changes from a logic 0 to a logic 1 and vice versa, the output frequency
shifts between two frequencies: a mark, or logic 1 frequency (fm) and a space or logic 0
frequency (fs). The mark and space frequencies are separated from the carrier frequency by
the peak frequency deviation (Δf ) and from each other by 2 Δf.
With FSK, frequency deviation is defined as the difference between either the mark or

18
space frequency and the center frequency or half the difference between the mark and
space frequencies. Frequency deviation can be expressed as Δf = |fm - fs| / 2
The baud for BFSK is determined by placing N = 1, i.e., baud = fb/1 = fb
The minimum bandwidth for FSK is determined from;
B = |(fs - fb) - (fm - fb)| = |fs - fm| + 2fb . But |fs - fm| = 2Δf,
Therefore, B= 2(Δf + fb), where B is minimum Nyquist bandwidth in hertz and Δf is
frequency deviation and fb is input bit rate.
Relationship between baud rate and bandwidth in FSK

The bandwidth for FSK is equal to the baud rate Nbaud of the signal plus
the frequency shift (difference between the two carrier frequencies)
BW=Nbaud + fc1-fc0

Example
Find the minimum bandwidth for an FSK signal
transmitting at 2000 bps. Transmission is in half-duplex
mode, and the carriers are separated by 3000 Hz.
Solution
For FSK
BW = baud rate + fc1 fc0
BW = bit rate + fc1 fc0 = 2000 + 3000 = 5000 Hz
Example
Find the maximum bit rates for an FSK signal if the
bandwidth of the medium is 12,000 Hz and the difference
between the two carriers is 2000 Hz. Transmission is in
full-duplex mode.
Solution
Because the transmission is full duplex, only 6000 Hz is
allocated for each direction.
BW = baud rate + fc1 fc0
Baud rate = BW (fc1 fc0 ) = 6000 2000 = 4000
But because the baud rate is the same as the bit rate, the
bit rate is 4000 bps.

Phase-Shift Keying

19
Phase-shift keying (PSK) is a digital modulation scheme that conveys data by changing,
or modulating, the phase of a reference signal (the carrier wave). PSK uses a finite number of
phases; each assigned a unique pattern of binary digits. Usually, each phase encodes an
equal number of bits. PSK is not susceptible to the noise degradation that affects ASK or to
the bandwidth limitations of FSK.

Binary phase-shift keying: The simplest PSK technique is called binary phase-shift keying
(BPSK), where N = 1 and M = 2. Therefore, with BPSK two phases are possible for the carrier.
It uses two opposite signal phases (0 and 180 degrees). The digital signal is broken up
timewise into individual bits (binary digits). The state of each bit is determined according to
the state of the preceding bit. If the phase of the wave does not change, then the signal
state stays the same (0 or 1). If the phase of the wave changes by 180 degrees -- that is, if
the phase reverses -- then the signal state changes (from 0 to 1 or from 1 to 0). Because
there are two possible wave phases, BPSK is sometimes called biphase modulation or
phasereversal keying (PRK).

More sophisticated forms of PSK exist. In M-ary or multiple phase-shift keying (MPSK), there
are more than two phases, usually four (0, +90, -90, and 180 degrees) or eight (0, +45, -45,
+90, -90, +135, -135, and 180 degrees). If there are four phases (m = 4), the MPSK mode is
called quadrature phase-shift keying or quaternary phase-shift keying (QPSK), and each
phase shift represents two signal elements. If there are eight phases (m = 8), the MPSK mode
is known as octal phase-shift keying (OPSK), and each phase shift represents three
signal elements. In MPSK, data can be transmitted at a faster rate, relative to the number of
phase changes per unit time, than is the case in BPSK.
QPSK is an M-ary encoding scheme where N = 2 and M = 4, which has four output phases are
possible for a single carrier frequency needing four different input conditions. With two bits,
there are four possible conditions: 00, 01, 10, and 11. With QPSK, the binary input data are
combined into groups of two bits called dibits.

The above figure shows the output phase-versus-time relationship, truth table, and

20
constellation diagram for QPSK. A phase of 00 now represents 00; 900 represents 01; 1800
represents 10; and 2700 represents 11. Data can be transmitted twice as efficiently using
4PSK than 2-PSK. With 8-PSK, three bits are encoded forming tribits and producing eight
different output phases. With 8-PSK, N = 3, M = 8, and the minimum bandwidth and baud
equal one third the bit rate (fb /3). 8-PSK is 3 times as efficient as 2-PSK.

Relationship between baud rate and bandwidth in PSK

21
Example
Find the bandwidth for a 4-PSK signal transmitting at 2000 bps. Transmission is half-duplex
mode.
Solution
For 4-PSK the baud rate is one-half of the bit rate. The baud rate is therefore 1000.
A PSK require a bandwidth equal to its baud rate.
Therefore , the bandwidth is 1000 Hz.
Example
Given a bandwidth of 5000 Hz for an 8-PSK signal, what
are the baud rate and bit rate?
Solution
For PSK the baud rate is the same as the bandwidth,
which means the baud rate is 5000.
But in 8-PSK the bit rate is 3 times the baud rate,
So the bit rate is 15,000 bps.

Quadrature amplitude modulation

Quadrature amplitude modulation is a combination of ASK and PSK so that a maximum
contrast between each signal unit (bit, dibit, tribit,and so on) is achieved.

QAM Modulator

22
QAM Demodulator

Time domain for an 8-QAM signal

23
16-QAM constellations

Bit and baud rate comparison

24
it

Example
A constellation diagram consists of eight equally spaced points on a circle. If the bit rate is
4800 bps, what is the baud rate?
Solution
The constellation indicates 8-PSK with the points 45 degrees apart.
Since 23 = 8, 3 bits are transmitted with each signal unit.
Therefore, the baud rate is: 4800 / 3 = 1600 baud

Example
Compute the bit rate for a 1000-baud 16-QAM signal.
Solution
A 16-QAM signal has 4 bits per signal unit since log216 = 4.
Thus,
(1000)*(4) = 4000 bps
Example 12
Compute the baud rate for a 72,000-bps 64-QAM signal.
Solution
A 64-QAM signal has 6 bits per signal unit since log2 64 = 6.
Thus,
72000 / 6 = 12,000 baud

25
 4. Analysis and Synthesis of Waveforms
Fourier series and alternative representation of periodic signals
Periodic functions can be represented in an alternative way other than plotting it as a function of
time. For example, for a signal

 2 
g (t )  cos t,
 T 

we can plot it as a function of time as usual.

Alternatively, we can plot the amplitude of this sinusoid as a function of frequency. This plot of
amplitude vs frequency is called the spectrum of g(t).

Amplitude
1


2
T

 2   2 
Now, if g (t )  cos t   0.5 cos 2  t  , the time domain plot and the spectrum are show n as
 T   T 
follows.

26
 Amplitude
1

0.5

2 2
2
T T

We are leading towards something much more general. Actually, a periodic function g(t) of period T
can be expressed as an infinite sum of sinusoidal waveforms. This summation is called Fourier series.
Fourier series can be written in several forms. One such form is the trigonometric Fourier series:

a0  2n t 
2n t
g (t )    an cos( )   bn sin( )
2 n 1 T n 1 T

where

2n t
T /2
2
an  
T T / 2
g (t ) cos(
T
)dt , n  0, 1, 2, 

2n t
T /2
2
bn  
T T / 2
g (t ) sin(
T
)dt , n 1, 2, 

We will show the proof in class.

 2   2 
If g(t) has period T, the term a1 cos t   b1 sin t  referred to as the fundamental frequency and the
 T   T 
other terms are known as the harmonics of g(t).

Example
g (t )

27
In this case, the Fourier coefficients of g(t) is given by

2 T /2  2nt  2  /2  2nt  2 sin(n / T )

an 
T T / 2 g (t ) cos T dt  T 
 /2
cos
 T 
dt 
n
and

2 T /2  2nt  2  /2  2nt 
bn 
T 
T / 2
g (t ) sin
 T 
dt 
T 
 /2
sin
 T 
dt  0

So therefore, the function g(t) can be expressed as

 
2 sin(n / T )  2 
g (t )   cos n t
T n 1 n  T 

Fourier series approximation to periodic functions

So far, we have only claimed that we can express any periodic function as a summation of sinusoids
2
with frequencies that are integer multiples of . However, is that really true? Do we actually get
T
the original function back by summing all these sinusoids?

In practice, we cannot sum an infinite number of sinusoids, so summing a finite number of sinusoids
will always be an approximation to the original function g(t). In other words,

a0 N  2 
g (t )    an cos n t
2 n 1  T 

Where N is finite.

28
The following figures show how g(t) compares with its Fourier representation as a function of N.

From the figure, we can see that as N   , the Fourier series summation approach the original function g(t)
The sinc function
2 sin(n / T )
If we write an  and define a function
T n / T

sin(x)
sin c( x)  ,
x
we can re-write the Fourier coefficients as

2
an  sin c(n / T ) .
T

The sinc function is shown in the following figure as a function of x. It has zero crossings at
x  1,2,3,... and we define sin c(0)  1 . It is an important function that we will use in this
course.

29
Exponential Fourier series
2
Defining  0  as the fundamental frequency of g(t), we can write
T

a0   2    2 
g (t )    an cos n t    bn sin n t
2 n 1  T  n 1  T 
 
 0   an cosn0t    bn sinn0t 
a
2 n 1 n 1

Now, from Euler’s formula, since

1 j0t
cos  0t  (e  e  j0t )
2
1 j0t  j0t
sin  0t  (e  e )
2j

We can re-write

a0  2n t 
2n t a  
g (t )    an cos( )   bn sin( )  0   an cos( n 0t )   bn sin( n 0t )
2 n 1 T n 1 T 2 n 1 n 1

a0  an jnot  jnot bn jnot  jnot a 

a  jbn jnot an  jbn  jnot
   [ (e e )  (e e )]  0   ( n e  e )
2 n 1 2 2j 2 n 1 2 2

 c0   (cn e jnot c n e  jnot )
n 1

where c0 = a0 / 2, cn = (an – jbn) / 2 and c-n = (an + jbn) / 2, n0 is the frequency of exponential Fourier
series. The formula can be written in the compact form

g (t )  ce
n 
n
jn 0t

where
T /2
1
T T/ 2
cn  g (t )e jn0t dt , n  0, 1, 2,

The exponential Fourier series find extensive applications in communication theory. It is equivalent
to resolving the function in terms of harmonically related frequency components of a fundamental
frequency. A weighting factor c n is assigned to each frequency component. cn is called spectral
amplitude and represents the amplitude of nth harmonic. Graphic representation of spectral

30
amplitude along with the spectral phase is called complex frequency spectrum. If cn is purely real or
purely imaginary, we can disregard the phase spectrum.

Concept of negative frequency

The exponential Fourier series represents an arbitrary function g(t) by a linear combination of
exponential functions of frequencies, 0, 0, 20, 30, …, Therefore, there will be negative
frequencies in the signals we are studying. However, it should be noted that negative frequencies
exist in our formulation only because we want to express cos(t) and sin(t) as exponentials. There is no
negative frequency signals in real life, rather they are only a mathematical concept required to give a
compact mathematical description of a signal. We will show that it is more convenient to use
exponential rather than trigonometric Fourier series representation.

Time domain and frequency domain representation of a signal

A signal can be specified in two equivalent ways:

1. Time domain representation;

where g(t) is represented as a function of time. Graphical time domain representation is termed
as waveform.

2. The frequency domain representation;

Where the signal is defined in terms of its spectrum. The signals are considered as the
superposition of many signals at different frequencies, thus their characteristics can be investigated
in the frequency domain.

Any of the above two representations uniquely specifies the function, i.e., if the signal is specified
in time domain, we can determine its spectrum. Conversely, if the spectrum is specified, we can
determine the corresponding time domain function of a signal.

Signals can be observed in the frequency domain by the use of a Spectrum Analyzer.

Fourier Coefficients dependence on period of g(t)

If we change the period of a signal g(t), what does it do to the Fourier coefficients?

31
 2
an  sin c(n / T )
T

There is an inverse relationship between the pulse width , and the frequency spread of the
spectrum. As the period becomes larger, the fundamental frequency 0 becomes smaller, generate
more frequency components in a given range of frequency; and therefore the spectrum becomes
denser. However, the amplitudes of the frequency components become smaller.

If the period T goes to infinity, we are left with a single pulse of width  in the time domain. The
fundamental frequency 0, of this waveform, approaches zero, i.e., no spacing is left between two
line-components. The spectrum becomes continuous and exists at all frequencies rather than only at
the discrete frequencies. However, there is no change in the shape of the envelope of the spectrum.

32
The continuous spectrum described above corresponds to a single non-repetitive pulse; i.e., a non-
periodic function existing over the entire interval (- < t < ). Thus, we have arrived at the spectrum
of a non-periodic function, taking it as a special case of periodic function with period T approaching
infinity. The transformation of an non-periodic function to its spectral or frequency domain
representation is known as the Fourier transform.

33
34
35
36
Pulse modulaton

Digital Transmission is the transmittal of digital signals between two or more points in a communications
system. The signals can be binary or any other form of discrete-level digital pulses. Digital pulses can not be
propagated through a wireless transmission system such as earth’s atmosphere or free space.

Alex H. Reeves developed the first digital transmission system in 1937 at the Paris Laboratories of AT & T for
the purpose of carrying digitally encoded analog signals, such as the human voice, over metallic wire cables between
telephone offices.

Advantages & disadvantages of Digital Transmission

Advantages

--Noise immunity

--Multiplexing(Time domain)

--Regeneration

--Simple to evaluate and measure

Disadvantages

--Requires more bandwidth

--Additional encoding (A/D) and decoding (D/A) circuitry

Pulse Modulation

-- Pulse modulation consists essentially of sampling analog information signals and then converting those
samples into discrete pulses and transporting the pulses from a source to a destination over a physical transmission
medium.

--The four predominant methods of pulse modulation:

1) pulse width modulation (PWM)

2) pulse position modulation (PPM)

3) pulse amplitude modulation (PAM)

4) pulse code modulation (PCM).

37
Pulse Width Modulation

--PWM is sometimes called pulse duration modulation (PDM) or pulse length modulation (PLM), as the width (active
portion of the duty cycle) of a constant amplitude pulse is varied proportional to the amplitude of the analog signal at
the time the signal is sampled.

--The maximum analog signal amplitude produces the widest pulse, and the minimum analog signal amplitude
produces the narrowest pulse. Note, however, that all pulses have the same amplitude.

Pulse Position Modulation

--With PPM, the position of a constant-width pulse within a prescribed time slot is varied according to the amplitude of
the sample of the analog signal.

--The higher the amplitude of the sample, the farther to the right the pulse is positioned within the prescribed time
slot. The highest amplitude sample produces a pulse to the far right, and the lowest amplitude sample produces a
pulse to the far left.

Pulse Amplitude Modulation

--With PAM, the amplitude of a constant width, constant-position pulse is varied according to the amplitude of the
sample of the analog signal.

-The amplitude of a pulse coincides with the amplitude of the analog signal.

38
--PAM waveforms resemble the original analog signal more than the waveforms for PWM or PPM.

Pulse Code Modulation

--With PCM, the analog signal is sampled and then converted to a serial n-bit binary code for transmission.

--Each code has the same number of bits and requires the same length of time for transmission

Pulse Modulation

--PAM is used as an intermediate form of modulation with PSK, QAM, and PCM, although it is seldom used by itself.

--PWM and PPM are used in special-purpose communications systems mainly for the military but are seldom used for
commercial digital transmission systems.

--PCM is by far the most prevalent form of pulse modulation and will be discussed in more detail.

Pulse Code Modulation

--PCM is the preferred method of communications within the public switched telephone network because with PCM it
is easy to combine digitized voice and digital data into a single, high-speed digital signal and propagate it over either
metallic or optical fiber cables.

--With PCM, the pulses are of fixed length and fixed amplitude.

--PCM is a binary system where a pulse or lack of a pulse within a prescribed time slot represents either a logic 1 or a
logic 0 condition.

-PWM, PPM, and PAM are digital but seldom binary, as a pulse does not represent a single binary digit (bit).

PCM system Block Diagram

--The band pass filter limits the frequency of the analog input signal to the standard voice-band frequency range of 300
Hz to 3000 Hz.

--The sample- and- hold circuit periodically samples the analog input signal and converts those samples to a multilevel
PAM signal.

--The analog-to-digital converter (ADC) converts the PAM samples to parallel PCM codes, which are converted to serial
binary data in the parallel-to-serial converter and then outputted onto the transmission linear serial digital pulses.

--The transmission line repeaters are placed at prescribed distances to regenerate the digital pulses.

--In the receiver, the serial-to-parallel converter converts serial pulses received from the transmission line to parallel
PCM codes.

--The digital-to-analog converter (DAC) converts the parallel PCM codes to multilevel PAM signals.

--The hold circuit is basically a low pass filter that converts the PAM signals back to its original analog form.

39
 The block diagram of a single-channel, simplex (one-way only) PCM system.

PCM Sampling:

--The function of a sampling circuit in a PCM transmitter is to periodically sample the continually changing analog input
voltage and convert those samples to a series of constant- amplitude pulses that can more easily be converted to
binary PCM code.

--A sample-and-hold circuit is a nonlinear device (mixer) with two inputs: the sampling pulse and the analog input
signal.

--For the ADC to accurately convert a voltage to a binary code, the voltage must be relatively constant so that the ADC
can complete the conversion before the voltage level changes. If not, the ADC would be continually attempting to
follow the changes and may never stabilize on any PCM code.

--Essentially, there are two basic techniques used to perform the sampling function

1) natural sampling

2) flat-top sampling

40
--Natural sampling is when tops of the sample pulses retain their natural shape during the sample interval, making it
difficult for an ADC to convert the sample to a PCM code.

--The most common method used for sampling voice signals in PCM systems is flat- top sampling, which is
accomplished in a sample-and-hold circuit.

-- The purpose of a sample-and-hold circuit is to periodically sample the continually changing analog input voltage and
convert those samples to a series of constant-amplitude PAM voltage levels.

Sampling Rate

--The Nyquist sampling theorem establishes the minimum Nyquist sampling rate (fs) that can be used for a given PCM
system.

--For a sample to be reproduced accurately in a PCM receiver, each cycle of the analog input signal (fa) must be
sampled at least twice.

--Consequently, the minimum sampling rate is equal to twice the highest audio input frequency.

--Mathematically, the minimum Nyquist sampling rate is:

fs ≥ 2fa

--If fs is less than two times fa an impairment called alias or foldover distortion occurs.

41
Quantization and the Folded Binary Code:

Quantization

--Quantization is the process of converting an infinite number of possibilities to a finite number of conditions.

--Analog signals contain an infinite number of amplitude possibilities.

--Converting an analog signal to a PCM code with a limited number of combinations requires quantization.

Folded Binary Code

--With quantization, the total voltage range is subdivided into a smaller number of sub-ranges.

--The PCM code shown in Table 10-2 is a three-bit sign- magnitude code with eight possible combinations (four positive
and four negative).

--The leftmost bit is the sign bit (1 = + and 0 = -), and the two rightmost bits represent magnitude.

-- This type of code is called a folded binary code because the codes on the bottom half of the table are a mirror image
of the codes on the top half, except for the sign bit.

Quantization

--With a folded binary code, each voltage level has one code assigned to it except zero volts, which has two codes, 100
(+0) and 000 (-0).

--The magnitude difference between adjacent steps is called the quantization interval or quantum.

--For the code shown in Table 10-2, the quantization interval is 1 V.

--If the magnitude of the sample exceeds the highest quantization interval, overload distortion (also called peak
limiting) occurs.

--Assigning PCM codes to absolute magnitudes is called quantizing.

--The magnitude of a quantum is also called the resolution.

42
--The resolution is equal to the voltage of the minimum step size, which is equal to the voltage of the least significant
bit (Vlsb) of the PCM code.

--The smaller the magnitude of a quantum, the better (smaller) the resolution and the more accurately the quantized
signal will resemble the original analog sample.

--For a sample, the voltage at t3 is approximately +2.6 V. The folded PCM code is

sample voltage = 2.6 = 2.6

resolution 1

--There is no PCM code for +2.6; therefore, the magnitude of the sample is rounded off to the nearest valid code,
which is 111, or +3 V.

--The rounding-off process results in a quantization error of 0.4 V.

--The likelihood of a sample voltage being equal to one of the eight quantization levels is remote.

--Therefore, as shown in the figure, each sample voltage is rounded off (quantized) to the closest available level and
then converted to its corresponding PCM code.

--The rounded off error is called the called the quantization error (Qe).

--To determine the PCM code for a particular sample voltage, simply divide the voltage by the resolution, convert the
quotient to an n-bit binary code, and then add the sign bit.

43
Linear input-versus-output transfer curve

resolution
Qe 
2

Linear transfer function

Quantization Quantization error (Qe)

1) For the PCM coding scheme shown in Figure 10-8, determine the quantized voltage, quantization error (Qe) and
PCM code for the analog sample voltage of + 1.07 V.

A) To determine the quantized level, simply divide the sample voltage by resolution and then round the answer off to
the nearest quantization level:

+1.07V = 1.07 = 1

1V

The quantization error is the difference between the original sample voltage and the quantized level, or Qe = 1.07 -1 =
0.07

44
From Table 10-2, the PCM code for + 1 is 101.

Dynamic Range (DR): It determines the number of PCM bits transmitted per sample.

-- Dynamic range is the ratio of the largest possible magnitude to the smallest possible magnitude (other than zero)
that can be decoded by the digital-to-analog converter in the receiver. Mathematically,

DR 
Vmax

Vmax
 2n  1 
DR dB   20log 2n 1 
Vmin resolution =20 log Vmax

Vmin

where DR = dynamic range (unitless)

Vmin = the quantum value

Vmax = the maximum voltage magnitude of the DACs

n = number of bits in a PCM code (excluding the sign bit)

For n > 4

DR  2n  1  2n
 
DR dB   20log 2n 1  20n log 2  6n

Signal-to-Quantization Noise Efficiency

--For linear codes, the magnitude change between any two successive codes is the same.

--Also, the magnitude of their quantization error is also same.

The maximum quantization noise is half the resolution. Therefore, the worst possible signal voltage- to-quantization
noise voltage ratio (SQR) occurs when the input signal is at its minimum amplitude (101 or 001). Mathematically, the
worst-case voltage SQR is

resolution
Qe 
SQR = resolution = Vlsb =2 2

Qe V lsb /2

45
For input signal minimum amplitude

SQR = minimum voltage / quantization noise

Vmin resolution
SQR min    2
Qe Qe

For input signal maximum amplitude

SQR = maximum voltage / quantization noise

Vmax
SQR max  
Qe

SQR is not constant where R = resistance (ohms)

SQR (dB) = 10 log v2 /R v = rms signal voltage (volts)

(q 2 /12)/R q = quantization interval (volts)

v2 /R = average signal power (watts)

(q 2 /12)/R = average quantization noise power (watts)

Linear vs. Nonlinear PCM codes

Linear Nonlinear

Companding

--Companding is the process of compressing and then expanding

--High amplitude analog signals are compressed prior to txn. and then expanded in the receiver

--Higher amplitude analog signals are compressed and Dynamic range is improved

46
--Early PCM systems used analog companding, where as modern systems use digital companding.

Basic companding process

Analog companding

PCM system with analog companding

--In the transmitter, the dynamic range of the analog signal is compressed, and then converted o a linear PCM code.

--In the receiver, the PCM code is converted to a PAM signal, filtered, and then expanded back to its original dynamic
range.

47
-- There are two methods of analog companding currently being used that closely approximate a logarithmic function
and are often called log-PCM codes.

The two methods are 1) -law and

2) A-law

-law companding

Vmax ln 1   in 
V

Vout   Vmax 
ln 1   

where Vmax = maximum uncompressed analog input amplitude(volts)

Vin = amplitude of the input signal at a particular instant of time (volts)

 = parameter used tio define the amount of compression (unitless)

Vout = compressed output amplitude (volts)

A-law companding

--A-law is superior to -law in terms of small-signal quality

--The compression characteristic is given by

 A| x| where
1 y = Vout
 1  log A , 0 | x |
A
y  x = Vin / Vmax
1  log( A | x |) 1
 , | x | 1
 1  log A A

48
Digital Companding: Block diagram refer in text book.

--With digital companding, the analog signal is first sampled and converted to a linear PCM code, and then the linear
code is digitally compressed.

-- In the receiver, the compressed PCM code is expanded and then decoded back to analog.

-- The most recent digitally compressed PCM systems use a 12- bit linear PCM code and an 8-bit compressed PCM
code.

Digital compression error

--To calculate the percentage error introduced by digital compression

%error=12-bit encoded voltage - 12-bit decoded voltage X 100

12-bit decoded voltage

PCM Line speed

--Line speed is the data rate at which serial PCM bits are clocked out of the PCM encoder onto the transmission line.
Mathematicaly,

Line speed= samples X bits

second sample

Delta Modulation

--Delta modulation uses a single-bit PCM code to achieve digital transmission of analog signals.

--With conventional PCM, each code is a binary representation of both the sign and the magnitude of a particular
sample. Therefore, multiple-bit codes are required to represent the many values that the sample can be.

--With delta modulation, rather than transmit a coded representation of the sample, only a single bit is transmitted,
which simply indicates whether that sample is larger or smaller than the previous sample.

--The algorithm for a delta modulation system is quite simple.

--If the current sample is smaller than the previous sample, a logic 0 is transmitted.

--If the current sample is larger than the previous sample, a logic 1 is transmitted.

49
Differential DM

--With Differential Pulse Code Modulation (DPCM), the difference in the amplitude of two successive samples is
transmitted rather than the actual sample. Because the range of sample differences is typically less than the range of
individual samples, fewer bits are required for DPCM than conventional PCM.

50
5.0 Transmission Media
5.1 Introduction
Transmission media can be defined as physical path between transmitter and receiver in a data
transmission system. And it may be classified into two types as shown in Fig. 5.1.
Guided: Transmission capacity depends critically on the medium, the length, and whether
the medium is point-to-point or multipoint (e.g. LAN). Examples are co-axial cable, twisted pair,
and optical fiber.
Unguided: provides a means for transmitting electro-magnetic signals but do not guide
them. Example wireless transmission.

Characteristics and quality of data transmission are determined by medium and signal
characteristics. For guided media, the medium is more important in determining the limitations of
transmission. While in case of unguided media, the bandwidth of the signal produced by the
transmitting antenna and the size of the antenna is more important than the medium. Signals at
lower frequencies are omni-directional (propagate in all directions). For higher frequencies,
focusing the signals into a directional beam is possible. These properties determine what kind of
media one should use in a particular application. In this lesson we shall discuss the characteristics
of various transmission media, both guided and unguied.

Figure 2.2.1 Classification of the transmission media

Figure 5.1 Classification of the transmission media

51
5.2 Guided transmission media
In this section we shall discuss about the most commonly used guided transmission media such as
twisted-pair of cable, coaxial cable and optical fiber.
5.2.1 Twisted Pair

Figure 5.2 CAT5 cable (twisted cable)

In twisted pair technology, two copper wires are strung between two points:
• The two wires are typically ``twisted'' together in a helix to reduce interference between the two
conductors as shown in Fig.5.2. Twisting decreases the cross-talk interference between adjacent
pairs in a cable. Typically, a number of pairs are bundled together into a cable by wrapping them
in a tough protective sheath.
 Can carry both analog and digital signals. Actually, they carry only analog signals.
However, the ``analog'' signals can very closely correspond to the square waves representing bits,
so we often think of them as carrying digital data.

 Data rates of several Mbps common.

 Spans distances of several kilometers.

 Data rate determined by wire thickness and length. In addition, shielding to eliminate
interference from other wires impacts signal-to-noise ratio, and ultimately, the data rate.

 Good, low-cost communication. Indeed, many sites already have twisted pair installed in
offices -- existing phone lines!

Typical characteristics: Twisted-pair can be used for both analog and digital communication.
The data rate that can be supported over a twisted-pair is inversely proportional to the square of
the line length. Maximum transmission distance of 1 Km can be achieved for data rates up to 1
Mb/s. For analog voice signals, amplifiers are required about every 6 Km and for digital signals,
repeaters are needed for about 2 Km. To reduce interference, the twisted pair can be shielded with
metallic braid. This type of wire is known as Shielded Twisted-Pair (STP) and the other form is
known as Unshielded Twisted-Pair (UTP).

Use: The oldest and the most popular use of twisted pair are in telephony. In LAN it is commonly
used for point-to-point short distance communication (say, 100m) within a building or a room.

5.2.2 Base band Coaxial

With ``coax'', the medium consists of a copper core surrounded by insulating material and a
braided outer conductor as shown in Fig. 5.3. The term base band indicates digital transmission
(as opposed to broadband analog).

52
Figure 5.3 Co-axial cable

Physical connection consists of metal pin touching the copper core. There are two common ways
to connect to a coaxial cable:
. With vampire taps, a metal pin is inserted into the copper core. A special tool drills a
hole into the cable, removing a small section of the insulation, and a special connector is screwed
into the hole. The tap makes contact with the copper core.

With a T-junction, the cable is cut in half, and both halves connect to the T-junction. A T-
connector is analogous to the signal splitters used to hook up multiple TVs to the same cable wire.

Characteristics: Co-axial cable has superior frequency characteristics compared to twisted-pair

and can be used for both analog and digital signaling. In baseband LAN, the data rates lies in the
range of 1 KHz to 20 MHz over a distance in the range of 1 Km. Co-axial cables typically have a
diameter of 3/8". Coaxial cables are used both for baseband and broadband communication. For
broadband CATV application coaxial cable of 1/2" diameter and 75 Ω impedance is used. This
cable offers bandwidths of 300 to 400 MHz facilitating high-speed data communication with low
bit-error rate. In broadband signaling, signal propagates only in one direction, in contrast to
propagation in both directions in baseband signaling. Broadband cabling uses either dual-cable
scheme or single-cable scheme with a head end to facilitate flow of signal in one direction.
Because of the shielded, concentric construction, co-axial cable is less susceptible to interference
and cross talk than the twisted-pair. For long distance communication, repeaters are needed for
every kilometer or so. Data rate depends on physical properties of cable, but 10 Mbps is typical.
Use: One of the most popular use of co-axial cable is in cable TV (CATV) for the distribution of
TV signals. Another importance use of co-axial cable is in LAN.

5.2.3 Broadband Coaxial

The term broadband refers to analog transmission over coaxial cable. (Note, however, that the
telephone folks use broadband to refer to any channel wider than 4 kHz). The technology:
 Typically bandwidth of 300 MHz, total data rate of about 150 Mbps.
 Operates at distances up to 100 km (metropolitan area!).
 Uses analog signaling.
 Technology used in cable television. Thus, it is already available at sites such as
universities that may have TV classes.
 Total available spectrum typically divided into smaller channels of 6 MHz each. That is,
to get more than 6MHz of bandwidth, you have to use two smaller channels and somehow
combine the signals.

53
 Requires amplifiers to boost signal strength; because amplifiers are one way, data flows in
only one direction.

Two types of systems have emerged:

1. Dual cable systems use two cables, one for transmission in each direction:
 One cable is used for receiving data.
 Second cable used to communicate with headend. When a node wishes to transmit data, it
sends the data to a special node called the headend. The headend then resends the data on
the first cable. Thus, the headend acts as a root of the tree, and all data must be sent to the
root for redistribution to the other nodes.
2. Midsplit systems divide the raw channel into two smaller channels, with each sub channel
having the same purpose as above.

Which is better, broadband or base band? There is rarely a simple answer to such questions. Base
band is simple to install, interfaces are inexpensive, but doesn't have the same range. Broadband is
more complicated, more expensive, and requires regular adjustment by a trained technician, but
offers more services (e.g., it carries audio and video too).

5.2.4 Fiber Optics

In fiber optic technology, the medium consists of a hair-width strand of silicon or glass, and the
signal consists of pulses of light. For instance, a pulse of light means ``1'', lack of pulse means
``0''. It has a cylindrical shape and consists of three concentric sections: the core, the cladding,
and the jacket as shown in Fig. 5.4.

Figure 5.4 Optical Fiber

The core, innermost section consists of a single solid dielectric cylinder of diameter d 1 and of
refractive index n1. The core is surrounded by a solid dielectric cladding of refractive index n2 that
is less than n1. As a consequence, the light is propagated through multiple total internal reflection.
The core material is usually made of ultra pure fused silica or glass and the cladding is either
made of glass or plastic. The cladding is surrounded by a jacket made of plastic. The jacket is
used to protect against moisture, abrasion, crushing and other environmental hazards.

54
 Three components are required:
1. Fiber medium: Current technology carries light pulses for tremendous distances (e.g., 100s of
kilometers) with virtually no signal loss.
2. Light source: typically a Light Emitting Diode (LED) or laser diode. Running current through
the material generates a pulse of light.
3. A photo diode light detector, which converts light pulses into electrical signals.

Advantages:
1. Very high data rate, low error rate. 1000 Mbps (1 Gbps) over distances of kilometers
common. Error rates are so low they are almost negligible.
2. Difficult to tap, which makes it hard for unauthorized taps as well. This is responsible for
higher reliability of this medium.
How difficult is it to prevent coax taps? Very difficult indeed, unless one can keep the
entire cable in a locked room!

1. Much thinner (per logical phone line) than existing copper circuits. Because of its thinness,
phone companies can replace thick copper wiring with fibers having much more capacity for
same volume. This is important because it means that aggregate phone capacity can be
upgraded without the need for finding more physical space to hire the new cables.
2. Not susceptible to electrical interference (lightning) or corrosion (rust).
3. Greater repeater distance than coax.

Disadvantages:
. Difficult to tap. It really is point-to-point technology. In contrast, tapping into coax is
trivial. No special training or expensive tools or parts are required.
. One-way channel. Two fibers needed to get full duplex (both ways) communication.

Optical Fiber works in three different types of modes (or we can say that we have 3 types of
communication using Optical fiber). Optical fibers are available in two varieties; Multi-Mode
Fiber (MMF) and Single-Mode Fiber (SMF). For multi-mode fiber the core and cladding diameter
lies in the range 50-200μm and 125-400μm, respectively. Whereas in single-mode fiber, the core
and cladding diameters lie in the range 8-12μm and 125μm, respectively. Single-mode fibers are
also known as Mono-Mode Fiber. Moreover, both single-mode and multi-mode fibers can have
two types; step index and graded index. In the former case the refractive index of the core is
uniform throughout and at the core cladding boundary there is an abrupt change in refractive
index. In the later case, the refractive index of the core varies radially from the centre to the core-
cladding boundary from n1 to n2 in a linear manner. Fig. 5.5 shows the optical fiber transmission
modes.

55
Figure 5.5 Schematics of three optical fiber types, (a) Single-mode step-index, (b) Multi-mode
step-index, and (c) Multi-mode graded-index

Characteristics: Optical fiber acts as a dielectric waveguide that operates at optical frequencies
(1014 to 1015 Hz). Three frequency bands centered around 850,1300 and 1500 nanometers are
used for best results. When light is applied at one end of the optical fiber core, it reaches the
other end by means of total internal reflection because of the choice of refractive index of core
and cladding material (n1 > n2). The light source can be either light emitting diode (LED) or
injection laser diode (ILD). These semiconductor devices emit a beam of light when a voltage is
applied across the device. At the receiving end, a photodiode can be used to detect the signal-
encoded light. Either PIN detector or APD (Avalanche photodiode) detector can be used as the
light detector.

In a multi-mode fiber, the quality of signal-encoded light deteriorates more rapidly than single-
mode fiber, because of interference of many light rays. As a consequence, single-mode fiber
allows longer distances without repeater. For multi-mode fiber, the typical maximum length of
the cable without a repeater is 2km, whereas for single-mode fiber it is 20km.

Fiber Uses: Because of greater bandwidth (2Gbps), smaller diameter, lighter weight, low
attenuation, immunity to electromagnetic interference and longer repeater spacing, optical fiber
cables are finding widespread use in long-distance telecommunications. Especially, the single
mode fiber is suitable for this purpose. Fiber optic cables are also used in high-speed LAN
applications. Multi-mode fiber is commonly used in LAN.

 Long-haul trunks-increasingly common in telephone network (Sprint ads)

 Metropolitan trunks-without repeaters (average 8 miles in length)
 Rural exchange trunks-link towns and villages
 Local loops-direct from central exchange to a subscriber (business or home)
 Local area networks-100Mbps ring networks.

56
5.2.5 Unguided Transmission
Unguided transmission is used when running a physical cable (either fiber or copper) between
two end points is not possible. For example, running wires between buildings is probably not
legal if the building is separated by a public street.

Infrared signals typically used for short distances (across the street or within same room),
Microwave signals commonly used for longer distances (10's of km). Sender and receiver use
some sort of dish antenna as shown in Fig. 5.6.

Figure 5.6 Communication using Terrestrial Microwave

Difficulties:
 . Weather interferes with signals. For instance, clouds, rain, lightning, etc. may adversely
affect communication.
 Radio transmissions easy to tap. A big concern for companies worried about competitors
stealing plans.
 Signals bouncing off of structures may lead to out-of-phase signals that the receiver must
filter out.

5.2.6 Satellite Communication

Satellite communication is based on ideas similar to those used for line-of-sight. A
communication satellite is essentially a big microwave repeater or relay station in the sky.
Microwave signals from a ground station is picked up by a transponder, amplifies the signal and
rebroadcasts it in another frequency, which can be received by ground stations at long distances as
shown in Fig. 5.7.
To keep the satellite stationary with respect to the ground based stations, the satellite is placed in a
geostationary orbit above the equator at an altitude of about 36,000 km. As the spacing between
two satellites on the equatorial plane should not be closer than 40, there can be 360/4 = 90
communication satellites in the sky at a time. A satellite can be used for point-to-point
communication between two ground-based stations or it can be used to broadcast a signal received
from one station to many ground-based stations as shown in Fig. 5.8. Number of geo-synchronous
satellites limited (about 90 total, to minimize interference). International agreements regulate how
satellites are used, and how frequencies are allocated. Weather affects certain frequencies.
Satellite transmission differs from terrestrial communication in another important way: One-way
propagation delay is roughly 270 ms. In interactive terms, propagation delay alone inserts a 1
second delay between typing a character and receiving its echo.

57
Figure 5.7 Satellite Microwave Communication: point –to- point

Figure 5.8 Satellite Microwave Communication: Broadcast links

Characteristics: Optimum frequency range for satellite communication is 1 to 10 GHz. The most
popular frequency band is referred to as 4/6 band, which uses 3.7 to 4.2 GHz for down link and
5.925 to 6.425 for uplink transmissions. The 500 MHz bandwidth is usually split over a dozen
transponders, each with 36 MHz bandwidth. Each 36 MHz bandwidth is shared by time division
multiplexing. As this preferred band is already saturated, the next highest band available is
referred to as 12/14 GHz. It uses 14 to 14.5GHz for upward transmission and 11.7 to 12.2 GHz for
downward transmissions. Communication satellites have several unique properties. The most
important is the long communication delay for the round trip (about 270 ms) because of the long
distance (about 72,000 km) the signal has to travel between two earth stations. This poses a
number of problems, which are to be tackled for successful and reliable communication.

58
Another interesting property of satellite communication is its broadcast capability. All stations
under the downward beam can receive the transmission. It may be necessary to send encrypted
data to protect against piracy.
Use: Now-a-days communication satellites are not only used to handle telephone, telex and
television traffic over long distances, but are used to support various internet based services such
as e-mail, FTP, World Wide Web (WWW), etc. New types of services, based on communication
satellites, are emerging.
Comparison/contrast with other technologies:
1. Propagation delay very high. On LANs, for example, propagation time is in nanoseconds --
essentially negligible.
2. One of few alternatives to phone companies for long distances.
3. Uses broadcast technology over a wide area - everyone on earth could receive a message at
the same time!
4. Easy to place unauthorized taps into signal.

Satellites have recently fallen out of favor relative to fiber.

However, fiber has one big disadvantage: no one has it coming into their house or building,
whereas anyone can place an antenna on a roof and lease a satellite channel.

59
8.0 MULTIPLEXING

Multiplexing is the name given to techniques, which allow more than one message to be transferred
via the same communication channel. The channel in this context could be a transmission line, e.g. a
twisted pair or co-axial cable, a radio system or a fibre optic system etc.

A channel will offer a specified bandwidth, which is available for a time t, where t may  . Thus,
with reference to the channel there are 2 ‘degrees of freedom’, i.e. bandwidth or frequency and
time.

CHANNEL
BL BH freq

BH

Multiplexing is a technique which allows k

users to occupy the
channel for the duration in time that the
channel is available.

BL
Frequency

Time t

Now consider a signal v s (t )  Amp cos(t   ) . The signal is characterised by amplitude, frequency,
phase and time.

Various multiplexing methods are possible in terms of the channel bandwidth and time, and the
signal, in particular the frequency, phase or time. The two basic methods are:

1) Frequency Division Multiplexing FDM

FDM is derived from AM techniques in which the signals occupy the same physical ‘line’ but
in different frequency bands. Each signal occupies its own specific band of frequencies all the
time, i.e. the messages share the channel bandwidth.

60
 2) Time Division Multiplexing TDM

TDM is derived from sampling techniques in which messages occupy all the channel
bandwidth but for short time intervals of time, i.e. the messages share the channel time.

 FDM – messages occupy narrow bandwidth – all the time.

 TDM – messages occupy wide bandwidth – for short intervals of time.
These two basic methods are illustrated below.

time

time
M1
M2 BL
BL M3 M5
M4 M3 M4
B M5 M1 M2

BH t
BH
freq freq

t
BH
BL
M1

M2

B M3 M1 M2 M3 M4 M5

M4

M5
BH BL
t t
FDM TDM

Frequency Division Multiplexing FDM

FDM is widely used in radio and television systems (e.g. broadcast radio and TV) and was widely used
in multichannel telephony (now being superseded by digital techniques and TDM). The multichannel
telephone system illustrates some important aspects and is considered below. For speech, a
bandwidth of  3kHz is satisfactory. The physical line, e.g. a co-axial cable will have a bandwidth
compared to speech as shown below.

61
 3kHz

freq

GHz

From AM we have noted:

m(t)

freq
m(t) DSBSC
B

carrier DSBSC
cos( c t ) freq

fc

In order to use bandwidth more effectively, SSB is used i.e.

SSB
m(t) Filter
SSBSC

carrier
cos( c t ) freq

fc

Note – the USB has been selected.

We have also noted that the message signal m(t) is usually band limited, i.e.

62
 Speech Band m(t) SSB SSBSC
Limiting Filter
Filter
300Hz – 3400Hz
cos( c t )

The Band Limiting Filter (BLF) is usually a band pass filter with a pass band 300Hz to 3400Hz for
speech. This is to allow guard bands between adjacent channels.

f f f
300Hz 3400Hz 300Hz 3400Hz
10kHz

Speech m(t) Convention

For telephony, the physical line is divided (notionally) into 4kHz bands or channels, i.e. the channel
spacing is 4kHz. Thus we now have:

Guard Bands

Bandlimited
Speech
f

4kHz

Note, the BLF does not have an ideal cut-off – the guard bands allow for filter ‘roll off’ in order to
reduce adjacent channel crosstalk.

Consider now a single channel SSB system.

63
 m(t) DSBSC SSB SSBSC
BLF Filter

fc

300Hz 3400Hz

The spectra will be

m(t)

freq

DSBSC

freq
fc

freq
fc

Consider now a system with 3 channels

m1(t)

SSB
f BLF
Filter
fc1 f1

FDM
m2(t) Signal
SSB 
BLF
Filter M(t)
f
fc2 f2

SSB
BLF
m3(t) Filter
fc3 f3
f
FDM Transmitter
Bandlimited
or Encoder

64
Each carrier frequency, fc1, fc2 and fc3 are separated by the channel spacing frequency, in this case 4
kHz, i.e. fc2 = fc1 + 4kHz, fc3 = fc2 + 4kHz.

The spectrum of the FDM signal, M(t) will be:

4kHz 4kHz 4kHz

M(t)

Shaded areas are to

show guard bands.

f1 f2 f3
freq
fc1 fc2 fc3

Note that the baseband signals m1(t), m2(t), m3(t) have been multiplexed into adjacent channels, the
channel spacing is 4kHz. Note also that the SSB filters are set to select the USB, tuned to f1, f2 and f3
respectively.

A receiver FDM decoder is illustrated below:

SSB
LPF m1(t)
Filter

f1
fc1 Band
Limited
SSB
M(t) LPF m2(t)
Filter
FDM Back to
Signal f2 baseband
fc2

SSB
LPF m3(t)
Filter

f3
fc3

The SSB filters are the same as in the encoder, i.e. each one centred on f1, f2 and f3 to select the
appropriate sideband and reject the others. These are then followed by a synchronous demodulator,
each fed with a synchronous LO, fc1, fc2 and fc3 respectively.

65
For the 3 channel system shown there is 1 design for the BLF (used 3 times), 3 designs for the SSB
filters (each used twice) and 1 design for the LPF (used 3 times).

A co-axial cable could accommodate several thousand 4 kHz channels, for example 3600 channels is
typical. The bandwidth used is thus 3600 x 4kHz = 14.4Mhz. Potentially therefore there are 3600
different SSB filter designs. Not only this, but the designs must range from kHz to MHz.

centre frequency
Consider also the ‘Q’ of the filter, where Q is defined as Q  .
bandwidth

60kHz
For ‘designs’ around say 60kHz, Q  = 15 which is reasonable. However, for designs to have a
4 kHz
10,000kHz
centre frequency at around say 10Mhz, Q  gives a Q = 2500 which is difficult to
4 kHz
achieve.

To overcome these problems, a hierarchical system for telephony used the FDM principle to form
groups, supergroups, master groups and supermaster groups.

Basic 12 Channel Group

The diagram below illustrates the FDM principle for 12 channels (similar to 3 channels) to a form a
basic group.

m1(t)

m2(t)

m3(t) Multiplexer
freq

12kHz 60kHz
m12(t)

i.e. 12 telephone channels are multiplexed in the frequency band 12kHz  60 kHz in 4kHz channels 
basic group. A design for a basic 12 channel group is shown below:

66
 Band Limiting Filters
SSB Filter
DSBSC
4kHz
CH1 8.6  15.4kHz 12.3  15.4kHz
m1(t)

300Hz 3400kHz
f1 = 12kHz

4kHz

12.6  19.4kHz 16.3  19.4kHz

CH2
m2(t)
300Hz 3400kHz

f1 = 16kHz


Increase in 4kHz steps

FDM OUT
12 – 60kHz
4kHz

CH12 52.6  59.4kHz

m12(t)
56.3  59.4kHz
300Hz 3400kHz

f12 = 56kHz

Super Group

These basic groups may now be multiplexed to form a super group.

BASIC
12 SSB
GROUP
Inputs FILTER
12 – 60kHz

420kHz

BASIC
12 SSB
GROUP
Inputs FILTER
12 – 60kHz

468kHz

BASIC
SSB

12
GROUP
Inputs FILTER
12 – 60kHz

516kHz

BASIC
SSB
12 GROUP
FILTER
Inputs 12 – 60kHz

564kHz

BASIC
SSB
12 GROUP
FILTER
Inputs 12 – 60kHz

612kHz

67
5 basic groups multiplexed to form a super group, i.e. 60 channels in one super group.

Note – the channel spacing in the super group in the above is 48kHz, i.e. each carrier frequency is
separated by 48kHz. There are 12 designs (low frequency) for one basic group and 5 designs for the
super group.

612 kHz
The Q for the super group SSB filters is Q   12 - which is reasonable. Hence, a total of 17
48kHz
designs are required for 60 channels. In a similar way, super groups may be multiplexed to form a
master group, and master groups to form super master groups…

Time Division Multiplexing TDM

TDM is widely used in digital communications, for example in the form of pulse code modulation in
digital telephony (TDM/PCM). In TDM, each message signal occupies the channel (e.g. a transmission
line) for a short period of time. The principle is illustrated below:

1
m1(t) 1
m1(t)
2
m2(t) 2
m2(t)
3
m3(t) 3
Tx Rx m3(t)
4 SW1 SW2
m4(t) 4
m4(t)
5 Transmission
m5(t) 5
Line m5(t)

Switches SW1 and SW2 rotate in synchronism, and in effect sample each message input in a
sequence m1(t), m2(t), m3(t), m4(t), m5(t), m1(t), m2(t),…

The sampled value (usually in digital form) is transmitted and recovered at the ‘far end’ to produce
output m1(t)…m5(t). For ease of illustration consider such a system with 3 messages, m1(t), m2(t) and
m3(t), each a different DC level as shown below.

68
 m1(t) V1

0 t

m2(t) V2

0 t
m3(t) V3

0 t

SW1
‗Sample‘

t
Position 1 2 3 1 2 3

V3
V2
V1
t
m1(t) m2(t) m3(t) m1(t) m2(t) m3(t) m1(t)

Channel
Time
Slots

1 2 3 1 2 3 1

Time slot

In this illustration the samples are shown as levels, i.e. V1, V2 or V3. Normally, these voltages would be
converted to a binary code before transmission as discussed below.

Note that the channel is divided into time slots and in this example, 3 messages are time-division
multiplexed on to the channel. The sampling process requires that the message signals are a sampled
at a rate fs  2B, where fs is the sample rate, samples per second, and B is the maximum frequency in
the message signal, m(t) (i.e. Sampling Theorem applies). This sampling process effectively produces
a pulse train, which requires a bandwidth much greater than B.

Thus in TDM, the message signals occupy a wide bandwidth for short intervals of time. In the
illustration above, the signals are shown as PAM (Pulse Amplitude Modulation) signals. In practice
these are normally converted to digital signals before time division multiplexing.

69
Data Rate Management
One problem with TDM is how to handle a disparity in the input data rates. If data rates are
not the same, three strategies, or a combination of them, can be used. These strategies are
multilevel multiplexing, multiple-slot allocation, and pulse stuffing.
Multilevel Multiplexing : - Multilevel multiplexing is a technique used when the data rate of an input
line is a multiple of others. For example, we have two inputs of 20 kbps and three inputs of 40
kbps. The first two input lines can be multiplexed together to provide a data rate equal to the
last three. A second level of multiplexing can create an output of 160 kbps.

Multiple-Slot Allocation :- Sometimes it is more efficient to allot more than one slot in a
frame to a single input line. For example, we might have an input line with a 50-kbps data rate
can be given two slots in the output. We insert a serial-to-parallel converter in the line to
make two inputs out of one.

Multiple slot for 50kbps

Pulse Stuffing :- Sometimes the bit rates of sources are not multiple integers of each other.
Therefore, neither of the above two techniques can be applied. One solution is to make the
highest input data rate the dominant data rate and then add dummy bits to the input lines with
lower rates. This will increase their rates. This technique is called pulse stuffing, bit padding,
or bit stuffing.
Example : The input with a data rate of 46 is pulse-stuffed to increase the rate to 50 kbps.
Now multiplexing can take place.

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Frame Synchronizing

The implementation of TDM is not as simple as that of FDM. Synchronization between the
multiplexer and demultiplexer is a major issue. If the multiplexer and the demultiplexer are
not synchronized, a bit belonging to one channel may be received by the wrong channel. For
this reason, one or more synchronization bits are usually added to the beginning of each
frame. These bits, called framing bits, follow a pattern, frame to frame, that allows the
demultiplexer to synchronize with the incoming stream so that it can separate the time slots
accurately. In most cases, this synchronization information consists of 1 bit per frame,
alternating between 0 and 1.
Statistical Time-Division Multiplexing
In synchronous TDM, each input has a reserved slot in the output frame. This can be
inefficient if some input lines have no data to send. In statistical time-division multiplexing,
slots are dynamically allocated to improve bandwidth efficiency. Only when an input line has
a slot's worth of data to send is it given a slot in the output frame. In statistical multiplexing,
the number of slots in each frame is less than the number of input lines. The multiplexer
checks each input line in round-robin fashion; it allocates a slot for an input line if the line has

71
data to send; otherwise, it skips the line and checks the next line.

Slot Size
Since a slot carries both data and an address in statistical TDM, the ratio of the data size to
address size must be reasonable to make transmission efficient. For example, it would be
inefficient to send 1 bit per slot as data when the address is 3 bits. This would mean an
overhead of 300 percent. In statistical TDM, a block of data is usually many bytes while the
address is just a few bytes.
No Synchronization Bit
There is another difference between synchronous and statistical TDM, but this time it is at the
frame level. The frames in statistical TDM need not be synchronized, so we do not need
synchronization bits.
Bandwidth
In statistical TDM, the capacity of the link is normally less than the sum of the capacities of
each channel.

72
9.0 Error Control Techniques.
9.1. INTRODUCTION
A transmission channel is the physical medium through which the information is transmitted, such as telephone
lines, or atmosphere in the case of wireless communication. Undesirable disturbances (noise) can occur across
the communication channel, causing the received information to be different from the original information sent.
Coding theory deals with detection and correction of the transmission errors caused by the noise in the channel.
The primary goal of coding theory is efficient encoding of information, easy transmission of encoded messages,
fast decoding of received information and correction of errors introduced in the channel. Coding Theory is used
all the time: in reading CDs, receiving transmissions from satellites, or in cell phones . The key challenge coding
theorists face is to construct "good" codes and efficient algorithms for encoding and decoding them. Encoding
introduces redundancy into a stream of data, and decoding uses the redundancy to correct errors and extract the
original data. This paper will focus on the analysis, design, and applications of the important class of codes.
Error-correcting codes are more sophisticated than error detecting codes and require more redundant bits.The
number of bits required correcting multiple bits or burst error is so high that in most of the cases it is inefficient
to do so. For this reason, most error correction is limited to one, two or at the most three-bit errors. Different
error coding schemes are chosen depending on the types of errors expected, the communication medium's
expected error rate, and whether or not data retransmission is possible. In section second and third two basic
strategies for dealing with errors are described. One way is to include enough redundant information along with
each block of data sent to enable the receiver to deduce what the transmitted character must have been. The other
way is to include only enough redundancy to allow the receiver to deduce that error has occurred, but not which
error has occurred and the receiver asks for a retransmission. The former strategy uses Error-Correcting Codes
and latter uses Error-detecting Codes. Further in section four error control technique ARQ is explained and in
section five some forward error correction codes are discussed along with their uses in day to day technology.
Finally, in section five we provide summary of various codes and discuss scope of future extensions.

9.2: ERROR DETECTION SCHEMES

In telecommunication, all error detection codes transmit more bits than were in the original data. Most codes are
"systematic" i.e. the transmitter sends a fixed number of original data bits, followed by fixed number of check
bits which are derived from the data bits by some deterministic algorithm. The receiver applies the same
algorithm to the received data bits and compares its output to the received check bits; if the values do not
match, an error has occurred at some point during the transmission. Now we will discuss some error detection
codes along with their uses and limitations.
A. Repetition Codes
One simplest possible solution to this problem is to repeat the transmission several times so as to increase
the probability of receiving the correct message. This is called a repetition code. Perhaps the most popular
repetition code is "triple modular redundancy", sending the same message 3 times. Let's say we have a repeated
message of 3 bits that is sent 3 times. When it is received, each message is different:
101 001 100
Because we didn't receive exactly the same message all 3 times, we have detected that some errors occurred.
We can see that the middle bit is certainly a 0, because it is the same in all messages and first and the last digit
are likely to be a 1 because 2 out of 3 of the messages say that these values are 1. The transmitted message then
was most likely 101.
The advantage of repetition codes is that they are extremely simple, and are in fact used in some transmissions of
numbers stations . Due to the simplicity of the channel encoding and decoding for repetition codes, they find
applications in fading channels and non-AWGN environments. Repetition codes are one of the few known codes
whose code rate can be automatically adjusted to varying channel capacity, by sending more or less parity
information as required to overcome the channel noise.
B. Parity Check Codes
Another error detection scheme is called the parity check or check bit. A parity check is the process of counting
the number of ones in a binary sequence. The parity check bit is a bit that is added to make the sequence even or
odd. It can be calculated via an XOR sum of the bits, yielding 0 for even parity and 1 for odd parity
If an odd number of bits (including the parity bit) are transmitted incorrectly, the parity bit will be incorrect and
thus indicates that an error occurred in transmission. The parity bit is only suitable for detecting errors; it cannot
correct any errors, as there is no way to determine which particular bit is corrupted. The data must be discarded
entirely, and re-transmitted from scratch . An even number of flipped bits will make the parity bit appear correct
even though the data is erroneous. However, parity has the advantage that it uses only a single bit and requires
only a number of XOR gates to generate. Extensions and variations on the parity bit mechanism are horizontal
redundancy checks, vertical redundancy checks, and "double," "dual," or "diagonal" parity (used in RAID-DP).

73
C. Checksum

A checksum or hash sum is a fixed size datum computed from an arbitrary block of digital data for the purpose
of detecting accidental errors that may have been introduced during its transmission or storage. Check digits and
parity are special cases of checksums, appropriate for small blocks of data (such as social Security bank account
numbers, social security no., computer words, single bytes, etc.). The simplest checksum algorithm is the so
called longitudinal parity check, which breaks the data into "words" with a fixed number n of bits, and then
computes the exclusive or of all those words. The result is appended to the message as an extra word. To check
the integrity of a message, the receiver computes the XOR of all its words, including the checksum; if the
result is not a word with n zeros; the receiver knows that a transmission error occurred.
D. Cyclic Redundancy Check (CRC)
Cyclic codes have favorable properties in that they are well suited for detecting burst errors. CRCs are
particularly easy to implement in hardware, and are therefore commonly used in digital networks and storage
devices such as hard disk drives. A CRCenabled device calculates a short, fixed-length binary sequence, known
as the check value or improperly the CRC, for each block of data to be sent or stored and appends it to the data,
forming a codeword. The idea behind CRC calculation is to look at the data as one large binary number. This
number is divided by a certain value and the remainder of the calculation is called the CRC. When a codeword is
received or read, the device either compares its check value with one freshly calculated from the data block, or
equivalently, performs a CRC on the whole codeword and compares the resulting check value with an expected
residue constant. If the check values do not match, then the block contains a data error and the device may seize
corrective action such as rereading or requesting the block is sent again, otherwise the data is assumed to be
error-free. CRCs are so called because the check (data verification) code is a redundancy (adds zero information
to the message) and the algorithm is based on cyclic codes. CRCs are popular because they are simple to
implement in binary hardware, are easy to analyze mathematically, and are particularly good at detecting
common errors caused by noise in transmission channels.
III. ERROR CORRECTION SCHEMES
The techniques that we have discussed so far can detect errors, but do not correct them. Error Correction
can be handled in two ways:
A. Backward Error Correction: In this scheme, when an error is discovered; the receiver has a back
channel to request the sender to retransmit the entire data unit, also known as ARQ.
B. Forward error correction: In this, receiver can use an error-correcting code, which automatically corrects
certain errors. Applications that require low latency (such as telephone conversations) cannot use Automatic
Repeat request (ARQ) ; they must use Forward Error Correction (FEC). By the time an ARQ system discovers
an error and re-transmits it, the re-sent data will arrive too late to be any good. Applications where the
transmitter immediately forgets the information as soon as it is sent (such as most television cameras) cannot use
ARQ; they must use FEC because when an error occurs, the original data is no longer available . This is also
why FEC is used in data storage systems such as RAID and distributed data store.
IV. AUTOMATIC REPEAT REQUEST
Automatic Repeat reQuest (ARQ) is an error control method for data transmission that makes use of
errordetection codes, acknowledgment and/or negative acknowledgment messages, and timeouts to achieve
reliable data transmission. An acknowledgment is a message sent by the receiver to indicate that it has
correctly received a data frame. Usually, when the transmitter does not receive the acknowledgment before
the timeout occurs (i.e., within a reasonable amount of time after sending the data frame), it retransmits the
frame until it is either correctly received. Three types of ARQ protocols are Stop-and-wait ARQ, Go-Back-N
ARQ, and Selective Repeat ARQ. ARQ is appropriate if the communication channel has varying or unknown
capacity, such as is the case on the Internet. However, ARQ requires the availability of a back channel, results in
possibly increased latency due to retransmissions, and requires the maintenance of buffers and timers for
retransmissions, which in the case of network congestion can put a strain on the server and overall network
capacity.
V. FORWARD ERROR CORRECTION
An error-correcting code (ECC) or forward error correction (FEC) code is a system of adding redundant data, or
parity data, to a message, such that it can be recovered by a receiver even when a number of errors (up to the
capability of the code being used) were introduced, either during the process of transmission, or on storage.
Since the receiver does not have to ask the sender for retransmission of the data, a back-channel is not required
in forward error correction, and it is therefore suitable for simplex communication such as broadcasting. Error-
correcting codes are frequently used in lower layer communication, as well as for reliable storage in media such
as CDs, DVDs, hard disks, and RAM. Error-correcting codes are usually distinguished between convolutional
codes and block codes: Convolutional codes are processed on a bit-by-bit basis. They are particularly suitable for
implementation in hardware, and the Viterbi decoder allows optimal decoding. Block codes are processed on a
block-byblock

74
basis. They were followed by a number of efficient codes, RS codes , Turbo codes [13] and low density parity
check code (LDPC) are relatively new constructions that can provide almost optimal efficiency. Shannon's
theorem is an important theorem in forward error correction and describes the maximum information rate at
which reliable communication is possible over a channel. This strict upper limit is expressed in terms of the
channel capacity. More specifically, the theorem says that there exist codes such that with increasing encoding
length the probability of error on a discrete memoryless channel can be made arbitrarily small, provided that
the code rate is smaller than the channel capacity.
A. BCH Codes
BCH codes were invented in 1959 by Hocquenghem, and independently in 1960 by Bose and Ray-Chaudhuri.
The codewords are formed by taking the remainder after dividing a polynomial representing our info bits by a
generator polynomial. The generator polynomial is selected to give the code its characteristics all codewords are
multiple of generator polynomial. The principal advantage of BCH codes is the ease with which they can be
decoded, via an elegant algebraic method known as syndrome decoding . This class of codes, is also highly
flexible, allowing control over block length and acceptable error thresholds, thus a custom code can be designed
to a given specification (subject to mathematical constraints). They have been widely used in communications
and data storage systems, including satellite communications, cellular networks, CD Rom, Mass Storage
Systems, wireless broadband, etc.
B. Hamming code
In telecommunication, a Hamming code is a linear error-correcting code named after its inventor, Richard
Hamming. Hamming codes can detect up to two simultaneous bit errors, and correct single-bit errors; thus,
reliable communication is possible when the Hamming distance between the transmitted and received bit
patterns is less than or equal to one. This means it is suitable for transmission medium situations where burst
errors do not occur. In particular, a singleerror-correcting and double-error-detecting variant commonly referred
to as SECDED.
C. Walsh-Hadamard code
In the field of mathematics, the Walsh-Hadamard code is an error correcting code over a binary alphabet that
allows reconstruction of any codeword if less than half its bits are corrupted. Furthermore, the Walsh-Hadamard
code is a locally decodable code which provides a way to recover the original message with high probability.
This gives rise to applications in complexity theory. It can also be shown that using list decoding; the original
message can be recovered as long as less than 1/2 of the bits in the received word have been corrupted. In coding
theory, the WH code is an example of a linear code over a binary alphabet that maps messages of length n to
codewords of length 2n. WH codes are mathematically orthogonal codes. As a result, a Walshencoded signal
appears as random noise to a CDMA capable mobile terminal, unless that terminal uses the same code as the one
used to encode the incoming signal. Walsh codes are used in direct sequence spread spectrum (DSSS) systems
such as QUALCOMM's CDMA, IS-95 and in frequency hopping spread spectrum (FHSS) systems to select the
target frequency for the next hop. Beside this, they are also used in power spectrum analysis, filtering,
processing speech and medical signals, multiplexing and coding in communications, characterizing non-linear
signals, solving non-linear differential equations, and logical design and analysis.
D. Reed–Solomon Codes
In coding theory, Reed–Solomon (RS) codes are non binary cyclic error correcting codes invented by Irving
S. Reed and Gustave Solomon in 1960. They described a systematic way of building codes that could detect and
correct multiple random symbol errors. RS codes work by adding t check symbols to the data, an RS code can
detect any combination of up to t erroneous symbols, and correct up to t/2 symbols. The choice of t is up to
the designer of the code. RS codes are usually constructed as systematic codes. Instead of sending s(x)
= p(x)g(x), the encoder will construct the transmitted polynomial s(x) such that it is evenly divisible
by g(x) and p(x) is apparent in the codeword. Ordinarily, the construction is done by multiplying p(x) by xt to
make room for the t check symbols, dividing that product by g(x) to find the remainder, and then
compensating for that remainder. The above properties of RS codes make them especially well-suited to
applications where errors occur in bursts . RS codes prominently used in consumer electronics such as CDs,
DVDs, Blue-ray Discs, in data transmission technologies such as DSL & WiMAX, in broadcast systems such as
DVB and ATSC, and in computer applications such as RAID 6 systems.
E. Low Density Parity Check Code (LDPC)
LDPC codes provide performance very close to the channel capacity (the theoretical maximum) using an iterated
soft-decision decoding approach, at linear time complexity in terms of their block length. Practical
implementations can draw heavily from the use of parallelism [10].The construction of a specific LDPC code
after this optimization falls into two main types of techniques:
a) Pseudo-random approaches
b) Combinatorial approaches
Construction by a pseudo-random approach builds on theoretical results that, for large block size, a

75
random construction gives good decoding performance. In general, pseudo-random codes have complex ncoders;
however pseudo-random codes with the best decoders can have simple encoders .Combinatorial approaches can
be used to optimize properties of small block-size LDPC codes or to create codes with simple encoders.
LDPC codes are now used in many recent high-speed communication standards, such as DVB-S2 (Digital video
broadcasting), WiMAX, High-Speed Wireless LAN (IEEE 802.11n), 10GBase-T Ethernet (802.3an) and
G.hn/G.9960 (ITU-T Standard for networking over power lines, phone lines and coaxial cable). Since 2009,
LDPC codes are also part of the Wi-Fi 802.11 standard as an optional part of 802.11n, in the High Throughput
(HT) PHY specification.
F. Turbo Codes
Turbo coding is an iterated soft-decoding scheme that combines two or more relatively simple convolutional
codes and an interleaver to produce a block code that can perform to within a fraction of a decibel of the
Shannon limit. The first turbo code, based on convolutional encoding, was introduced in 1993 by Berrou et al.
Predating LDPC codes in terms of practical application, they now provide similar performance. One of the
earliest commercial applications of turbo coding was the CDMA2000 1x digital cellular technology
specifically for Internet access, 1xEV-DO (TIA IS-856). Turbo codes find their major applications in field of
telecommunications:
Turbo codes are used extensively in 3G and 4G mobile telephony standards e.g. in HSPA, EV-DO and LTE.
MediaFLO, terrestrial mobile television system from Qualcomm. The interaction channel of satellite
communication systems, such as DVB-RCS. New NASA missions such as Mars Reconnaissance Orbiter now
use turbo codes, as an alternative to RS-Viterbi codes. Turbo coding such as block turbo coding and
convolutional turbo coding are used in IEEE 802.16 (WiMAX), a wireless metropolitan network standard.
Turbo codes are used for pictures, video, and mail transmissions. For voice transmission, however, convolutional
codes are used, because the decoding delay, the time it takes to decode the data, is a major drawback to turbo
codes. The several iterations required by turbo decoding make the delay unacceptable for realtime voice
communications and other applications that require instant data processing, like hard disk storage and optical
transmission.

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10.0:Internet Protocols

Background
The Internet protocols are the world‘s most popular open-system (nonproprietary) protocol suite
because they can be used to communicate across any set of interconnected networks and are equally
well suited for LAN and WAN communications. The Internet protocols consist of a suite of
communication protocols, of which the two best known are the Transmission Control Protocol (TCP)
and the Internet Protocol (IP). The Internet protocol suite not only includes lower-layer protocols
(such as TCP and IP), but it also specifies common applications such as electronic mail, terminal
emulation, and file transfer. This chapter provides a broad introduction to specifications that comprise
the Internet protocols. Discussions include IP addressing and key upper-layer protocols used in the
Internet. Specific routing protocols are addressed individually in Part 6, Routing Protocols.
Internet protocols were first developed in the mid-1970s, when the Defense Advanced Research
Projects Agency (DARPA) became interested in establishing a packet-switched network that would
facilitate communication between dissimilar computer systems at research institutions. With the goal
of heterogeneous connectivity in mind, DARPA funded research by Stanford University and Bolt,
Beranek, and Newman (BBN). The result of this development effort was the Internet protocol suite,
completed in the late 1970s.
TCP/IP later was included with Berkeley Software Distribution (BSD) UNIX and has since become
the foundation on which the Internet and the World Wide Web (WWW) are based.
Documentation of the Internet protocols (including new or revised protocols) and policies are
specified in technical reports called Request For Comments (RFCs), which are published and then
reviewed and analyzed by the Internet community. Protocol refinements are published in the new
RFCs. To illustrate the scope of the Internet protocols, Figure 30-1 maps many of the protocols of the
Internet protocol suite and their corresponding OSI layers. This chapter addresses the basic elements
and operations of these and other key Internet protocols

Figure 10-1 Internet protocols span the complete range of OSI model layers.

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Internet Protocol (IP)
The Internet Protocol (IP) is a network-layer (Layer 3) protocol that contains addressing information
and some control information that enables packets to be routed. IP is documented in RFC 791 and
is the primary network-layer protocol in the Internet protocol suite. Along with the Transmission
Control Protocol (TCP), IP represents the heart of the Internet protocols. IP has two primary
responsibilities: providing connectionless, best-effort delivery of datagrams through an
internetwork; and providing fragmentation and reassembly of datagrams to support data links with
different maximum-transmission unit (MTU) sizes.
IP Packet Format
An IP packet contains several types of information, as illustrated in Figure 10-2.

Figure 10-2 Fourteen fields comprise an IP packet.

The following discussion describes the IP packet fields illustrated in Figure 10-2:
• Version—Indicates the version of IP currently used.
• IP Header Length (IHL)—Indicates the datagram header length in 32-bit words.
• Type-of-Service—Specifies how an upper-layer protocol would like a current datagram to be
handled, and assigns datagrams various levels of importance.
• Total Length—Specifies the length, in bytes, of the entire IP packet, including the data and
header.
• Identification—Contains an integer that identifies the current datagram. This field is used to help
piece together datagram fragments.
• Flags—Consists of a 3-bit field of which the two low-order (least-significant) bits control
fragmentation. The low-order bit specifies whether the packet can be fragmented. The middle bit
specifies whether the packet is the last fragment in a series of fragmented packets. The third or
high-order bit is not used.
• Fragment Offset—Indicates the position of the fragment‘s data relative to the beginning of the
data in the original datagram, which allows the destination IP process to properly reconstruct the
original datagram.
• Time-to-Live—Maintains a counter that gradually decrements down to zero, at which point the
datagram is discarded. This keeps packets from looping endlessly.
• Protocol—Indicates which upper-layer protocol receives incoming packets after IP processing is
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complete.
• Header Checksum—Helps ensure IP header integrity.
• Source Address—Specifies the sending node.
• Destination Address—Specifies the receiving node
• Options—Allows IP to support various options, such as security.
• Data—Contains upper-layer information.
IP Addressing
As with any other network-layer protocol, the IP addressing scheme is integral to the process of
routing IP datagrams through an internetwork. Each IP address has specific components and follows
a basic format. These IP addresses can be subdivided and used to create addresses for subnetworks,
as discussed in more detail later in this chapter.
Each host on a TCP/IP network is assigned a unique 32-bit logical address that is divided into two
main parts: the network number and the host number. The network number identifies a network and
must be assigned by the Internet Network Information Center (InterNIC) if the network is to be part
of the Internet. An Internet Service Provider (ISP) can obtain blocks of network addresses from the
InterNIC and can itself assign address space as necessary. The host number identifies a host on a
network and is assigned by the local network administrator.
IP Address Format
The 32-bit IP address is grouped eight bits at a time, separated by dots, and represented in decimal
format (known as dotted decimal notation). Each bit in the octet has a binary weight (128, 64, 32,
16, 8, 4, 2, 1). The minimum value for an octet is 0, and the maximum value for an octet is 255.
Figure 10-3 illustrates the basic format of an IP address.

Figure 10-3 An IP address consists of 32 bits, grouped into four octets.

IP Address Classes
IP addressing supports five different address classes: A, B,C, D, and E. Only classes A, B, and C are
available for commercial use. The left-most (high-order) bits indicate the network class. Table 10-1
provides reference information about the five IP address classes.

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Table 10-1 Reference Information About the Five IP Address Classes

Figure 10-4 illustrates the format of the commercial IP address classes. (Note the high-order bits in
each class.)
Figure 10-4 IP address formats A, B, and C are available for commercial use.

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Figure 10-5 A range of possible values exists for the first octet of each address class.

IP Subnet Addressing
IP networks can be divided into smaller networks called subnetworks (or subnets). Subnetting provides the
network administrator with several benefits, including extra flexibility, more efficient use of network addresses,
and the capability to contain broadcast traffic (a broadcast will not cross a router).
Subnets are under local administration. As such, the outside world sees an organization as a single network and
has no detailed knowledge of the organization‘s internal structure.
A given network address can be broken up into many subnetworks. For example, 172.16.1.0,
172.16.2.0, 172.16.3.0, and 172.16.4.0 are all subnets within network 171.16.0.0. (All 0s in the host portion of an
address specifies the entire network.)
IP Subnet Mask
A subnet address is created by ―borrowing‖ bits from the host field and designating them as the subnet field. The
number of borrowed bits varies and is specified by the subnet mask. Figure 10-6 shows how bits are borrowed
from the host address field to create the subnet address field.
Figure 10-6 Bits are borrowed from the host address field to create the subnet address
field.

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Figure 10-7 A sample subnet mask consists of all binary 1s and 0s.

Subnet mask bits should come from the high-order (left-most) bits of the host field, as Figure 10-8 illustrates.
Details of Class B and C subnet mask types follow. Class A addresses are not discussed in this chapter because
they generally are subnetted on an 8-bit boundary.
Figure 10-8 Subnet mask bits come from the high-order bits of the host field.

Various types of subnet masks exist for Class B and C subnets. The default subnet mask for a Class B address
that has no subnetting is 255.255.0.0, while the subnet mask for a Class B address 171.16.0.0 that specifies eight
bits of subnetting is 255.255.255.0. The reason for this is that eight bits of subnetting or 2 8 – 2 (1 for the network
address and 1 for the broadcast address) = 254 subnets possible, with 28 – 2 = 254 hosts per subnet.
The subnet mask for a Class C address 192.168.2.0 that specifies five bits of subnetting is 255.255.255.248.With
five bits available for subnetting, 25 – 2 = 30 subnets possible, with 23 – 2 = 6 hosts per subnet. The reference
charts shown in table 30–2 and table 30–3 can be used when planning Class B and C networks to determine the
required number of subnets and hosts, and the appropriate subnet mask.

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Table 10-2 Class B Subnetting Reference Chart

How Subnet Masks are Used to Determine the Network Number

The router performs a set process to determine the network (or more specifically, the subnetwork) address. First,
the router extracts the IP destination address from the incoming packet and retrieves the internal subnet mask. It
then performs a logical AND operation to obtain the network number. This causes the host portion of the IP
destination address to be removed, while the destination network number remains. The router then looks up the
destination network number and matches it with an outgoing interface. Finally, it forwards the frame to the
destination IP address. Specifics regarding the logical AND operation are discussed in the following section.
Logical AND Operation
Three basic rules govern logically ―ANDing‖ two binary numbers. First, 1 ―ANDed‖ with 1 yields 1. Second, 1
―ANDed‖ with 0 yields 0. Finally, 0 ―ANDed‖ with 0 yields 0. The truth table provided in table 30–4 illustrates
the rules for logical AND operations.
Table 10-4 Rules for Logical AND Operations

Two simple guidelines exist for remembering logical AND operations: Logically ―ANDing‖ a 1 with
a 1 yields the original value, and logically ―ANDing‖ a 0 with any number yields 0.

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Figure 10-9 illustrates that when a logical AND of the destination IP address and the subnet mask is performed,
the subnetwork number remains, which the router uses to forward the packet.
Figure 10-9 Applying a logical AND the destination IP address and the subnet mask
produces the subnetwork number.

Address Resolution Protocol (ARP) Overview

For two machines on a given network to communicate, they must know the other machine‘s physical (orMAC)
addresses. By broadcasting Address Resolution Protocols (ARPs), a host can dynamically discover the MAC-
layer address corresponding to a particular IP network-layer address. After receiving aMAC-layer address, IP
devices create an ARP cache to store the recently acquired IP-to-MAC address mapping, thus avoiding having to
broadcast ARPS when they want to recontact a device. If the device does not respond within a specified time
frame, the cache entry is flushed.
In addition to the Reverse Address Resolution Protocol (RARP) is used to mapMAC-layer addresses to IP
addresses. RARP, which is the logical inverse of ARP, might be used by diskless workstations that do not know
their IP addresses when they boot. RARP relies on the presence of a RARP server with table entries of MAC-
layer-to-IP address mappings.
Internet Routing
Internet routing devices traditionally have been called gateways. In today‘s terminology, however, the term
gateway refers specifically to a device that performs application-layer protocol translation between devices.
Interior gateways refer to devices that perform these protocol functions between machines or networks under the
same administrative control or authority, such as a corporation‘s internal network. These are known as
utonomous systems. Exterior gateways perform protocol functions between independent networks. Routers
within the Internet are organized hierarchically. Routers used for information exchange within autonomous
systems are called interior routers, which use a variety of Interior Gateway Protocols (IGPs) to accomplish this
purpose. The Routing Information Protocol (RIP) is an example of an IGP. Routers that move information
between autonomous systems are called exterior routers. These routers use an exterior gateway protocol to
exchange information between autonomous systems. The Border Gateway Protocol (BGP) is an example of an
exterior gateway protocol.
IP Routing
IP routing protocols are dynamic. Dynamic routing calls for routes to be calculated automatically at regular
intervals by software in routing devices. This contrasts with static routing, where routers are established by the
network administrator and do not change until the network administrator changes them. An IP routing table,
which consists of destination address/next hop pairs, is used to enable dynamic routing. An entry in this table, for
example, would be interpreted as follows: to get to network 172.31.0.0, send the packet out Ethernet interface 0
(E0).
IP routing specifies that IP datagrams travel through internetworks one hop at a time. The entire route is not
known at the onset of the journey, however. Instead, at each stop, the next destination is calculated by matching
the destination address within the datagram with an entry in the current node‘s routing table. Each node‘s
involvement in the routing process is limited to forwarding packets based on internal information. The nodes do
not monitor whether the packets get to their final destination, nor does IP provide for error reporting back to the
source when routing anomalies occur. This task is left to another Internet protocol, the Internet Control-Message
Protocol (ICMP), which is discussed in the following section.

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Internet Control Message Protocol (ICMP)
The Internet Control Message Protocol (ICMP) is a network-layer Internet protocol that provides message
packets to report errors and other information regarding IP packet processing back to the source. ICMP is
documented in RFC 792.
ICMP Messages
ICMPs generate several kinds of useful messages, including Destination Unreachable, Echo Request and Reply,
Redirect, Time Exceeded, and Router Advertisement and Router Solicitation. If an ICMP message cannot be
delivered, no second one is generated. This is to avoid an endless flood of ICMP messages. When an ICMP
destination-unreachable message is sent by a router, it means that the router is unable to send the package to its
final destination. The router then discards the original packet. Two reasons exist for why a destination might be
unreachable. Most commonly, the source host has specified a nonexistent address. Less frequently, the router
does not have a route to the destination.
Destination-unreachable messages include four basic types: network unreachable, host unreachable, protocol
unreachable, and port unreachable. Network-unreachable messages usually mean that a failure has occurred in
the routing or addressing of a packet. Host-unreachable messages usually indicates delivery failure, such as a
wrong subnet mask. Protocol-unreachable messages generally mean that the destination does not support the
upper-layer protocol specified in the packet.
Port-unreachable messages imply that the TCP socket or port is not available. An ICMP echo-request message,
which is generated by the ping command, is sent by any host to test node reachability across an internetwork.
The ICMP echo-reply message indicates that the node can be successfully reached.
An ICMP Redirect message is sent by the router to the source host to stimulate more efficient routing. The router
still forwards the original packet to the destination. ICMP redirects allow host routing tables to remain small
because it is necessary to know the address of only one router, even if that router does not provide the best path.
Even after receiving an ICMP Redirect message, some devices might continue using the less-efficient route.
An ICMP Time-exceeded message is sent by the router if an IP packet‘s Time-to-Live field (expressed in hops
or seconds) reaches zero. The Time-to-Live field prevents packets from continuously circulating the
internetwork if the internetwork contains a routing loop. The router then discards the original packet.
ICMP Router-Discovery Protocol (IDRP)
IDRP uses Router-Advertisement and Router-Solicitation messages to discover the addresses of routers on
directly attached subnets. Each router periodically multicasts Router-Advertisement messages from each of its
interfaces. Hosts then discover addresses of routers on directly attached subnets by listening for these messages.
Hosts can use Router-Solicitation messages to request immediate advertisements rather than waiting for
unsolicited messages.
IRDP offers several advantages over other methods of discovering addresses of neighboring routers.
Primarily, it does not require hosts to recognize routing protocols, nor does it require manual configuration by an
administrator.
Router-Advertisement messages enable hosts to discover the existence of neighboring routers, but not which
router is best to reach a particular destination. If a host uses a poor first-hop router to reach a particular
destination, it receives a Redirect message identifying a better choice.
Transmission Control Protocol (TCP)
The TCP provides reliable transmission of data in an IP environment. TCP corresponds to the transport layer
(Layer 4) of the OSI reference model. Among the services TCP provides are stream data transfer, reliability,
efficient flow control, full-duplex operation, and multiplexing.
With stream data transfer, TCP delivers an unstructured stream of bytes identified by sequence numbers. This
service benefits applications because they do not have to chop data into blocks before handing it off to TCP.
Instead, TCP groups bytes into segments and passes them to IP for delivery.
TCP offers reliability by providing connection-oriented, end-to-end reliable packet delivery through an
internetwork. It does this by sequencing bytes with a forwarding acknowledgment number that indicates to the
destination the next byte the source expects to receive. Bytes not acknowledged within a specified time period
are retransmitted. The reliability mechanism of TCP allows devices to deal with lost, delayed, duplicate, or
misread packets. A time-out mechanism allows devices to detect lost packets and request retransmission.
TCP offers efficient flow control, which means that, when sending acknowledgments back to the source, the
receiving TCP process indicates the highest sequence number it can receive without overflowing its internal
buffers.
Full-duplex operation means that TCP processes can both send and receive at the same time.
Finally, TCP‘s multiplexing means that numerous simultaneous upper-layer conversations can be multiplexed
over a single connection.

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TCP Connection Establishment
To use reliable transport services, TCP hosts must establish a connection-oriented session with one another.
Connection establishment is performed by using a ―three-way handshake‖ mechanism.
A three-way handshake synchronizes both ends of a connection by allowing both sides to agree upon initial
sequence numbers. This mechanism also guarantees that both sides are ready to transmit data and know that the
other side is ready to transmit as well. This is necessary so that packets are not transmitted or retransmitted
during session establishment or after session termination.
Each host randomly chooses a sequence number used to track bytes within the stream it is sending and receiving.
Then, the three-way handshake proceeds in the following manner:
The first host (Host A) initiates a connection by sending a packet with the initial sequence number (X) and SYN
bit set to indicate a connection request. The second host (Host B) receives the SYN, records the sequence
number X, and replies by acknowledging the SYN (with an ACK = X + 1). Host B includes its own initial
sequence number (SEQ = Y). An ACK = 20 means the host has received bytes 0 through 19 and expects byte 20
next. This technique is called forward acknowledgment. Host A then acknowledges all bytes Host B sent with a
forward acknowledgment indicating the next byte Host A expects to receive (ACK = Y + 1). Data transfer then
can begin.
Positive Acknowledgment and Retransmission (PAR)
A simple transport protocol might implement a reliability-and-flow-control technique where the source sends
one packet, starts a timer, and waits for an acknowledgment before sending a new packet. If the acknowledgment
is not received before the timer expires, the source retransmits the packet. Such a technique is called positive
acknowledgment and retransmission (PAR).
By assigning each packet a sequence number, PAR enables hosts to track lost or duplicate packets caused by
network delays that result in premature retransmission. The sequence numbers are sent back in the
cknowledgments so that the acknowledgments can be tracked.
PAR is an inefficient use of bandwidth, however, because a host must wait for an acknowledgment before
sending a new packet, and only one packet can be sent at a time.
TCP Sliding Window
A TCP sliding window provides more efficient use of network bandwidth than PAR because it enables hosts to
send multiple bytes or packets before waiting for an acknowledgment.
In TCP, the receiver specifies the current window size in every packet. Because TCP provides a byte-stream
connection, window sizes are expressed in bytes. This means that a window is the number of data bytes that the
sender is allowed to send before waiting for an acknowledgment. Initial window sizes are indicated at connection
setup, but might vary throughout the data transfer to provide flow control. A window size of zero, for instance,
means ―Send no data.‖
In a TCP sliding-window operation, for example, the sender might have a sequence of bytes to send (numbered 1
to 10) to a receiver who has a window size of five. The sender then would place a window around the first five
bytes and transmit them together. It would then wait for an acknowledgment.
The receiver would respond with an ACK = 6, indicating that it has received bytes 1 to 5 and is expecting byte 6
next. In the same packet, the receiver would indicate that its window size is 5. The sender then would move the
sliding window five bytes to the right and transmit bytes 6 to 10. The receiver would respond with an ACK = 11,
indicating that it is expecting sequenced byte 11 next. In this packet, the receiver might indicate that its window
size is 0 (because, for example, its internal buffers are full). At this point, the sender cannot send any more bytes
until the receiver sends another packet with a window size greater than 0.

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TCP Packet Format
Figure 10-10 illustrates the fields and overall format of a TCP packet.
Figure 10-10 Twelve fields comprise a TCP packet.

TCP Packet Field Descriptions

The following descriptions summarize the TCP packet fields illustrated in Figure 30-10:
• Source Port and Destination Port—Identifies points at which upper-layer source and destination
processes receive TCP services.
• Sequence Number—Usually specifies the number assigned to the first byte of data in the current
message. In the connection-establishment phase, this field also can be used to identify an initial
sequence number to be used in an upcoming transmission.
• Acknowledgment Number—Contains the sequence number of the next byte of data the sender of
the packet expects to receive.
• Data Offset—Indicates the number of 32-bit words in the TCP header.
• Reserved—Remains reserved for future use.
• Flags—Carries a variety of control information, including the SYN and ACK bits used for
connection establishment, and the FIN bit used for connection termination.
• Window—Specifies the size of the sender‘s receive window (that is, the buffer space available for
incoming data).
• Checksum—Indicates whether the header was damaged in transit.
• Urgent Pointer—Points to the first urgent data byte in the packet.
• Options—Specifies various TCP options.
• Data—Contains upper-layer information.

User Datagram Protocol (UDP)

The User Datagram Protocol (UDP) is a connectionless transport-layer protocol (Layer 4) that belongs to the
Internet protocol family. UDP is basically an interface between IP and upper-layer processes. UDP protocol
ports distinguish multiple applications running on a single device from one another.
Unlike the TCP, UDP adds no reliability, flow-control, or error-recovery functions to IP. Because of UDP‘s
simplicity, UDP headers contain fewer bytes and consume less network overhead than TCP.
UDP is useful in situations where the reliability mechanisms of TCP are not necessary, such as in cases where a
higher-layer protocol might provide error and flow control.

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UDP is the transport protocol for several well-known application-layer protocols, including Network File System
(NFS), Simple Network Management Protocol (SNMP), Domain Name System (DNS), and Trivial File Transfer
Protocol (TFTP).
The UDP packet format contains four fields, as shown in Figure 10-11. These include source and destination
ports, length, and checksum fields.
Figure 10-11 A UDP packet consists of four fields.

Source and destination ports contain the 16-bit UDP protocol port numbers used to demultiplex datagrams for
receiving application-layer processes. A length field specifies the length of the UDP header and data. Checksum
provides an (optional) integrity check on the UDP header and data.
Internet Protocols Application-Layer Protocols
The Internet protocol suite includes many application-layer protocols that represent a wide variety
of applications, including the following:
• File Transfer Protocol (FTP)—Moves files between devices
• Simple Network-Management Protocol (SNMP)—Primarily reports anomalous network
conditions and sets network threshold values
• Telnet—Serves as a terminal emulation protocol
• X Windows—Serves as a distributed windowing and graphics system used for communication
between X terminals and UNIX workstations
• Network File System (NFS), External Data Representation (XDR), and Remote Procedure Call
(RPC)—Work together to enable transparent access to remote network resources
• Simple Mail Transfer Protocol (SMTP)—Provides electronic mail services
• Domain Name System (DNS)—Translates the names of network nodes into network addresses
Table 10-5 lists these higher-layer protocols and the applications that they support.

Table 10-5 Higher-Layer Protocols and Their Applications

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