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UNIT I FUNDAMENTALS OF ANALOG COMMUNICATION

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UNIT II DIGITAL COMMUNICATION


INTRODUCTION:

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FREQUENCY SHIFT KEYING:

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UNIT III DIGITAL TRANSMISSION


INTRODUCTION:

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Delta Modulation (DM) :

Delta Modulation (DM) :

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Let m[ n ] = m(nT s ) , n = 0,1,2,


where Ts is the sampling

period and m( nT s ) is a sample of m(t ).

The error signal is


e[ n ] = m[ n ] mq [ n 1]

(3.52)

eq [ n ] = sgn( e[ n ] )

(3.53)

mq [ n ] = mq [ n 1] +eq [ n ]

where mq [ n ] is the quantizer


the quantized

(3.54)
output , eq [ n ] is

version of e[ n ] , and is the step size

The modulator consists of a comparator, a quantizer, and an accumulator. The output of


the accumulator is

Differential Pulse-Code Modulation (DPCM):


Usually PCM has the sampling rate higher than the Nyquist rate .The encode signal
contains redundant information. DPCM can efficiently remove this redundancy.

Figure 3.28 DPCM system. (a) Transmitter. (b) Receiver.

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[n ]
e[n ] = m[n] m
[n] is a prediction
m

(3.74)

value.
The quantizer output is
eq [n] = e[n] + q[n]

(3.75)

where q[n] is quantizati on error.


The prediction filter input is
[n ] +e[n] +q[n] (3.77)
mq [ n] = m
m[n ]
mq [n] = m[ n] + q[ n]

(3.78)

Processing Gain:
The (SNR)
(SNR)

of the DPCM system is

M2
Q2

(3.79)

where M2 and Q2 are variances of m[ n ] ( E[ m[ n]] = 0) and q[ n ]


(SNR)

=(

M2 E2
)(
)
E2 Q2

= G p (SNR ) Q

(3.80)

where E2 is the variance of the prediction s error


and the signal - to - quantizati on noise ratio is
(SNR ) Q =

E2
Q2

Processing Gain, G p =

(3.81)

M2
(3.82)
E2

Design a prediction filter to maximize G p (minimize E2 )

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NEAR EAST UNIVERSITY


FACULTY OF ENGINEERING
DEPARTMENT OF COMPUTER ENGINEERING

COM 318
DATA COMMUNICATIONS
LECTURE NOTES

Prepared by
Dr. Tayseer Alshanableh

Nicosia-2007

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TABLE OF CONTENTS

CH. 1: INTRODUCTION TO DATA COMMUNICATIONS


CH. 2: DATA TRANSMISSION & SIGNALS
CH. 3: TRANSMISSION MEDIA
CH. 4: ENCODING, MODULATING & TRANSMISSION CODES
CH. 5: TRANSMISSION OF DIGITAL DATA: INTERFACES & MODEMS
CH. 6: MULTIPLEXING
CH. 7: ERROR DETECTION AND CORRECTION

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CHAPTER 1
INTRODUCTION TO DATA COMMUNICATIONS

COMPUTER NETWORK

Interconnected collection of autonomous computers that are able to exchange information.

No master/slave relationship between computers in the network.

DATA COMMUNICATIONS

Transmission of signals in a reliable and efficient matter.


-

COMMUNICATION MODEL (SYSTEM)

The purpose of a communications system is to exchange data between two entities.

Source: entity that generates data; eg. a person who speaks into the phone, or a
computer sending data to the modem.

Transmitter: a device to transform/encode the signal generated by the source.


- the transformed signal is actually sent over the transmission system.
eg. a modem transforms digital data to analog signal that can be handled by the
telephone network.

Transmission System (Channel): medium that allows the transfer of a signal from
one point to another.
eg. a telephone network for a computer/modem.

Receiver: a device to decode the received signal for handling by destination device.
eg. A modem converts the received analog data back to digital for the use by the
computer.

Destination: entity that finally uses the data.


eg. Computer on other end of a receiving modem.

Data Communications
Data communications is the transfer of information that is in digital form, before it enters the
communication system.
- Basic Elements of a Communication System
Signal s(t)

Channel r(t)

Transmitter

Receiver

n(t)
Information
source & input
transducer

Noise

Output
transducer

Basic Elements of a Communication System


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Information: generated by the source may be in the form of voice, a picture or a plain
text. An essential feature of any source that generates information is that its output is
described in probabilistic terms; that is, the output is not deterministic.
A transducer is usually required to convert the output of a source in an electrical signal
that is suitable for transmission.

Transmitter: a transmitter converts the electrical signal into a form that is suitable for
transmission through the physical channel or transmission medium. In general, a
transmitter performs the matching of the message signal to the channel by a process
called modulation.
The choice of the type of modulation is based on several factors, such as:
- the amount of bandwidth allocated,
- the type of noise and interference that the signal encounters in transmission over the
channel,
- and the electronic devices that are available for signal amplification prior to
transmission.

Channel: the communication channel is the physical medium that connects the
transmitter to the receiver. The physical channel may be a pair of wires that carry the
electrical signals, or an optical fibre that carries the information on a modulated light
beam or free space at which the information-bearing signal are electromagnetic waves.

Receiver: the function of a receiver is to recover the message signal contained in the
received signal. The main operations performed by a receiver are demodulation,
filtering and decoding.

Analog and Digital Data Transmission


-

Data are entries that convey information.

Signals are electrical encoding (representation) of data.

Signalling is the act of propagation of signals through a suitable medium.

The terms analog and digital correspond to continuous and discrete, respectively. These
two terms are frequently used in data communications.

Analog data takes on continuous values on some interval. The most familiar example of
analog data is audio signal. Frequency components of speech may be found between
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20 Hz and 20 kHz. The basic speech energy is concentrated between 300-3400 Hz. The
frequencies up to 4000 Hz add very little to the intelligibility of human ear.
Another common example of analog data is video. The outputs of many sensors, such as
temperature and pressure sensors, are also examples of analog data.
Digital data takes on discrete values; eg. a computers output.
-

Analog transmission is a means of transmitting analog signals regardless of their


content. The data may be analog or digital.

Digital transmission is the transfer of information through a medium in digital form. A


digital signal can be transmitted only for a limited distance.

Data communications is the transfer of information that is in digital form, before it


enters the communication system.

Two methods of sending data from computer A to computer B. both cases are
examples of data communications, because the original data is digital in nature.
Digital
Source

Modem

Modem

Analog Transmission

B
Digital Transmission

Two ways of transmitting analog information. In either cases it is not data


communications, because the original information is not digital.
Digital Transmission

ADC

DAC

Analog

Destination

Source

Analog Transmission

ADC: Analog-Digital-Converter
DAC: Digital-Analog-Converter
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Digital Communication System
Up to this point, we have described an electrical communication system in rather broad terms
based on the implicit assumption that the message signal is a continuous time-varying
waveform. We refer to such continuous-time signal waveforms as analog signals and to the
corresponding information sources that produce such signals as analog sources. Analog
signals can be transmitted directly via carrier modulation over the communication channel
and demodulated accordingly at the receiver. We call such a communication system an analog
communication system.
Alternatively, an analog source output may be converted into a digital form and the message
can be transmitted via digital modulation and demodulated as a digital signal at the receiver.
There are some potential advantages to transmitting an analog signal by means of digital
modulation. The most important reason is that signal fidelity is better controlled through
digital transmission than analog transmission. In particular, digital transmission allows us to
regenerate the digital signal in long-distance transmission, thus eliminating effects of noise at
each regeneration point. In contrast, the noise added in analog transmission is amplified
analog with the signal when amplifiers are used periodically to boost the signal level in longdistance transmission. Another reason for choosing digital transmission over analog is that the
analog message signal may be highly redundant. With digital processing, redundancy may be
removed prior to modulation, thus conserving channel bandwidth. Yet a third reason may be
that digital communication systems are often cheaper to implement.
In some applications, the information to be transmitted is inherently digital, e.g., in the form
of English text, computer data, etc. In such cases, the information source that generates the
data is called a discrete (digital) source.
In a digital communication system, the functional operations performed at the transmitter and
receiver must be expanded to include message signal discrimination at the transmitter and
message signal synthesis or interpolation at the receiver. Additional functions include
redundancy removal, and channel coding and decoding.
Figure 1.2 illustrates the functional diagram and the basic elements of a digital
communication system. The source output may be either an analog signal, such as audio or
video signal, or a digital signal, such as the output of a Teletype machine, which is discrete in
time and has a finite number of output characters. In a digital communication system, the
messages produced by the source are usually converted into a sequence of binary digits.

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Information
source & input
transducer

Source
encoder

Channel
encoder

Digital
modulator

Data Communication System

Output
Signal

Output
transducer

Source
decoder

Channel
decoder

Channel

Digital
demodulator

Figure 1.2. Basic elements of a digital communication system


Ideally, we would like to represent the source output (message) by as few binary digits as
possible. In other words, we seek an efficient representation of the source output that results
in little or no redundancy. The process of efficiently converting the output of either an analog
or a digital source into a sequence of binary digits is called source encoder or data
compression.
The sequence of binary digits from the source encoder, which we call the information
sequence, is passed to the channel encoder. The purpose of the channel encoder is to introduce
in a controlled manner some redundancy in the binary information sequence, which can be
used at the receiver to overcome the effects of noise and interference encountered in the
transmission of the signal through the channel. Thus, the added redundancy serves to increase
the reliability of the received data and improves the fidelity of the received signal. In effect,
redundancy in the information sequence aids the receiver in decoding the desired information
sequence. For example, a (trivial) form of encoding of the binary information sequence is
simply to repeat each binary digit m times, where m is some positive integer. More
sophisticated (nontrivial) encoding involves taking k information bits at a time and mapping
each k-bit sequence into a unique n-bit sequence, called a code word. The amount of
redundancy introduced by encoding the data in this manner is measured by the ratio n/k. The
reciprocal of this ratio, namely, k/n is called the rate of the code or, simply, the code rate.
The binary sequence at the output of the channel encoder is passed to the digital modulator,
which serves as the interface to the communications channel. Since nearly all of the
communication channels encountered in practice are capable of transmitting electrical signals
(waveforms), the primary purpose of the digital modulator is to map the binary information
sequence into signal waveforms. To elaborate on this point, let us suppose that the coded
information sequence is to be transmitted one bit at a time at some uniform rate R bits/s. The
digital modulator may simply map the binary digit 0 into a waveform s0(t) and the binary digit
1 into a waveform s1(t). In this manner, each bit from the channel encoder is transmitted
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separately. We call this binary modulation. Alternatively, the modulator may transmit b coded
information bits at a time by using M =2b distinct waveform si(t), I = 0, 1, , m-1, one
waveform for each of the 2b possible b-bits sequences. We call this M-ary modulation (M
>2). Note that a new b-bit sequence enters the modulator every b/R seconds. Hence, when the
channel bit rate R is fixed, the amount of time available to transmit one of the M waveforms
corresponding to a b-bit sequence is b times the period in a system that uses binary
modulation.
At the receiving end of a digital communications system, the digital demodulator processes
the channel-corrupted transmitted waveform and reduces each waveform to a single number
that represents an estimate of the transmitted data symbol. For example, when binary
modulation is used, the demodulator may process the received waveform and decide on
whether the transmitted bit is a 0 or 1. In such a case, we say the demodulator has made a
binary decision. As one alternative, the demodulator may make a ternary decision; that is, it
decides that the transmitted bit is either a 0 or 1 or it makes no decision at all, depending on
the apparent quality of the received signal. When no decision is made on a particular bit, we
say that the demodulator has inserted an erasure in the demodulated data. Using the
redundancy in the transmitted data, the decoder attempts to fill in the positions where erasures
occurred. Viewing the decision process performed by the demodulator as a form of
quantization, we observe that binary and ternary decisions are special cases of a demodulator
that quantizes to Q levels, where Q 2 In general, if the digital communications system
employs M-ary modulation, where m = 0,1, , M represent the M possible transmitted symbols, each corresponding to k =log2 M bits, the demodulator may make A Q-ary decision,
where Q M .In the extreme case where no quantization is performed, Q = .
When there is no redundancy in the transmitted information, the demodulator must decide
which of the M waveforms was transmitted in any given time interval. Consequently, Q = M,
and since there is no redundancy in the transmitted information, no discrete channel decoder
is used following the demodulator. On the other hand, when there is redundancy introduced
by a discrete channel encoder at the transmitter, the Q-ary output from the demodulator
occurring every k/R seconds is fed to the decoder, which attempts to reconstruct the original
information sequence from knowledge of the code used by the channel encoder and the
redundancy contained in the received data. A measure of how well the demodulator and
encoder perform is the frequency with which errors occur in the decoded sequence. More
precisely, the average probability of a bit-error at the output of the decoder is a measure of the
performance of the demodulator-decoder combination. In general, the probability of error is a
function of the code characteristics, the types of waveforms used to transmit the information
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over the channel, the transmitter power, the characteristics of the channel (i.e., the amount of
noise), the nature of the interference, etc., and the method of demodulation and decoding.
These items and their effect on performance will be discussed in detail in subsequent
chapters.
As a final step, when an analog output is desired, the source decoder accepts the output
sequence from the channel decoder, and from knowledge of the source encoding method
used, attempts to reconstruct the original signal from the source. Due to channel decoding
errors and possible distortion introduced by the source encoder and, perhaps, the source
decoder, the signal at the output of the source decoder is an approximation to the original
source output. The difference or some function of the difference between the original signal
and the reconstructed signal is a measure of the distortion introduced by the digital
communications system.
Early Work in Digital Communications
Although Morse is responsible for the development of the first electrical digital
communication system (telegraphy), the beginnings of what we now regard as modem digital
communications stem from the work of Nyquist (1924), who investigated the problem of
determining the maximum signalling rate that can be used over a telegraph channel of a given
bandwidth without intersymbol interference. He formulated a model of a telegraph system in
which a transmitted signal has the general form

S (t ) a n g (t nT )
n

Where g(t) represents a basic pulse shape and {an} is the binary data sequence of {1}
transmitted at a rate of 1/T bits per second. Nyquist set out to determine the optimum pulse
shape that was bandlimited to W Hz and maximised the bit rate 1/T under the constraint that
the pulse caused no intersymbol interference at the sampling times k/T, k = 0, 1, 2,.... His
studies led him to conclude that the maximum pulse rate1/T is 2W pulses per second. This rate
is now called the Nyquist rate. Moreover, this pulse rate can be achieved by using the pulses
g(t) = (sin2 Wt)/2 Wt. This pulse shape allows the recovery of the data without intersymbol
interference at the sampling instants Nyquists result is equivalent to a version of the sampling
theorem for bandlimited signals, which was later stated precisely by Shannon (1948). The
sampling theorem states that a signal of bandwidth W can be reconstructed from samples
taken at the Nyquist rate of 2W samples per second using the interpolation formula

S (t ) s(
n

n sin 2W (t n / 2W )
)
2W 2W (t n / 2W )

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In light of Nyquist's work Hartley (1928) considered the issue of the amount of data that can
be transmitted reliably over a bandlimited channel when multiple amplitude levels are used.
Due to the presence of noise and other interference, Hartley postulated that the receiver could
reliably estimate the received signal amplitude to some accuracy, say A. This investigation
led Hartley to conclude that there is maximum data rate that can be communicated reliably
over a bandlimited channel when the maximum signal amplitude is limited to Amax (fixed
power constraint) and the amplitude resolution is A.
Another significant advance in the development of communications was the work of Wiener
(1942) who considered the problem of estimating a desired signal waveform s(t) in the
presence of additive noise n(t), based on observation of the received signal r(t) = s(t) + n(t).
This problem arises in signal demodulation. Wiener determined the linear filter whose output
is the best mean-square approximation to the desired signal s(t). The resulting filter is called
the optimum linear (Wiener) filter. Hartleys and Nyquist results on the maximum
transmission rate of digital information were precursors to the work of Shannon (1948 a, b)
who established the mathematical foundations for information theory and derived the
fundamental limits for digital communication systems. In his pioneering work, Shannon
formulated the basic problem of reliable transmission of information in statistical terms, using
probabilistic models for information sources and communication channels. Based on such a
statistical formulation, he adopted a logarithmic measure for the information content of a
source. He also demonstrated that the effect of a transmitter power constraint, a bandwidth
constraint, and additive noise can be associated with the channel and incorporated into a
single parameter, called the channel capacity For example, in the case of an additive white
(spectrally flat) Gaussian noise interference, an ideal bandlimited channel of bandwidth W has
a capacity C given by
P
C = W log2(1 + ) bits/s
WN0
where P is the average transmitted power and N0 is the power spectral density of the additive
noise. The significance of the channel capacity is as follows: If the information rate R from
the source is less than C (R < C), then it is theoretically possible to achieve reliable (errorfree) transmission through the channel by appropriate coding. On the other hand, if R > C,
reliable transmission is not possible regardless of the amount of signal processing performed
at the transmitter and receiver. Thus, Shannon established basic limits on communication of
information and gave birth to a new field that is now called information theory.
Initially the fundamental work of Shannon had a relatively small impact on the design and
development of new digital communications systems. In part, this was due to the small
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demand for digital information transmission during the 1950's. Another reason was the
relatively large complexity and, hence, the high cost of digital hardware required to achieve
the high efficiency and high reliability predicted by Shannon's theory.
Another important contribution to the field of digital communications is the work of
Kotelnikov (1947), which provided a coherent analysis of the various digital communication
systems based on a geometrical approach. Kotelnikov approach was later expanded by
Wozencraft and Jacobs (1965).
The increase in the demand for data transmission during the last three decades, coupled with
the development of more sophisticated integrated circuits, has led to the development of very
efficient and more reliable digital communications systems. In the course of these
developments, Shannon's original results and the generalization of his results on maximum
transmission limits over a channel and on bounds on the performance achieved have served as
benchmarks for any given communications system design. The theoretical limits derived by
Shannon and other researchers that contributed to the development of information theory
serve as an ultimate goal in the continuing efforts to design and develop more efficient digital
communications systems.
Following Shannon's publications name the classic work of Hamming (1950) on error
detecting and error-correcting codes to combat the detrimental effects of channel noise.
Hamming's work stimulated many researchers in the years that followed, and a variety of new
and powerful codes were discovered, many of which are used today in the implementation of
modem communication systems.

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CHAPTER II
DATA TRANSMISSION & SIGNALS

Data Transmission
Concepts and Terminology

Transmission Terminology
Transmission from transmitter to receiver goes over some transmission medium using
electromagnetic waves.
- Guided Media: waves are guided along a physical path; twisted pair, optical fibre,
coaxial cable.
- Unguided Media: waves are not guided; air waves radio waves.
- Direct Link: signal goes from transmitter to receiver without intermediate devices,
other than amplifiers and repeaters.
- Point-to Point Link: guided media with direct link between two devices.
- Multipoint Guided Configuration: more than two devices can share the same
medium.

Frequency, Spectrum, & Bandwidth

Signal is generated by a transmitter and transmitted over a medium.

Signal is a function of time or frequency.

A signal is any function that carries information. Based on the range of variation of
independent variables, signals can be divided into two classes: continuous-time (or
analog) signals and discrete-time (or digital) signals. A signal is a function of time, but
can also be expressed as a function of frequency; that is, the signal consists of components
of different frequencies.

Analog and Digital Signals

Information can be analog or digital. Analog information is continuous. Digital


information is discrete.
Signals can be analog or digital. Analog signals can have any value in a range; while
digital signals can have only a limited number of values.

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Effective Bandwidth
Narrow band of frequencies containing most of the energy of the signal.

DC Component
Component of zero frequency; changes the average amplitude of the signal to non-zero.

Relationship between Data Rate and Bandwidth


-

Any transmitter/receiver system can accommodate only a limited range of frequencies.


* The range for FM radio transmission is 88-108 MHz

This limits the data rate that can be carried over the transmission medium.

Consider a square wave. Suppose that we let the positive pulse to be binary 1 and the
negative pulse to be binary 0. Then, the waveform represents the binary stream
1010 and duration (period) of each pulse is 1/2f. Thus, the data rate is equal to 2f
bits per second (bps) or the data rate is equal to twice the fundamental frequency of
the digital signal. It can be shown that the frequency-domain representation of this
waveform is:

1
s(t ) sin(2kft )
k 1 k

This waveform has infinite number of frequency components and infinite bandwidth.

Peak amplitude of the kth frequency component is 1/k, so most of the energy is
concentrated in the first few frequencies.

Ex
Consider a digital transmission system capable of transmitting signals with a bandwidth of
4 MHz.

Case 1
Approximate the square wave with a waveform of the first three sinusoidal components
sin (2ft ) 1 / 3 sin(2 (3 f )t ) 1 / 5 sin(2 (5 f )t )

If f = 106 cycles per second, or 1 MHz, the bandwidth of the signal

s(t ) sin (2 10 6 t ) 1 / 3 sin(2 3 10 6 t ) 1 / 5 sin(2 5 10 6 t )


is 5x106-106 = 4 MHz
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For f = 1 MHz, the period of the fundamental frequency is T = 1/10 6 = 1s. If the
waveform is a bit string of 1s and 0s, then one bit occurs every 0.5 s for a data rate of
2x106 bps or 2 Mbps.

Case 2
Assume a bandwidth of 8 MHz and f = 2 MHz; this gives us the signal bandwidth as
(5x2x106)-(2x106) = 8 MHz
But T = 1/f = 0.5 s, so that the time needed for one bit is 0.25 s, giving a data rate of
4 Mbps. Other things being equal, doubling the bandwidth doubles the potential data rate.

Case 3
Let us represent the signal by the first two components of the sinusoid as

sin (2ft ) 1/ 3sin(2 (3 f )t )


Assume that f = 2 MHz and T = 1/f = 0.5 s so that the time needed for one bit is 0.25 s,
giving a data rate of 4 Mbps.
Bandwidth of the signal is
(3x2x106)-(2x106) = 4 MHz
A given bandwidth can support various data rates depending on the ability of the receiver
to differentiate between 0 and 1 in the presence of noise and other impairments.

Ex
If a periodic signal is decomposed into five waves with frequencies of 100, 300, 500, 700,
and 900 Hz, what is the bandwidth of the signal?

Let fh be the highest frequency, fl be the lowest frequency, and B be the bandwidth, then
B = fh - fl
= 900-100 = 800 Hz
Digital Signals
Data can be represented by a digital signal. For example, a 1 can be encoded as a positive
voltage and a 0 as a zero voltage.

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Noise
In the absence of a signal, a transmission medium ideally has no electrical signal present.
In practice, however, there is what we call line noise level, because of random
perturbations on the line even when no signal is being transmitted. An important
parameter associated with a transmission medium, therefore, is the ratio of the average
power in a received signal, S , to the power in the noise level, N . The ratio S / N is
known as the signal-to-noise ratio (SNR) and normally is expressed in decibels, that is:
S
SNR = 10 log 10 dB
N

- A high SNR ratio means a good-quality signal.


- A low SNR ratio means a low-quality signal.
The theoretical maximum data rate of transmission channel is related to the SNR ratio
and we can determine this rate using a formula attributed to Shannon and Hartley. This is
known as the Shannon-Hartley Law, which states:
S

C W log 2 1 bps
N

3.32 W log 10 1 bps


N

where C is the data rate in bps, W is the bandwidth of the line channel in Hz, S is the
average signal power in watts and N is the random noise power in watts.
Ex
Consider a voice channel with BW of 2,800 Hz. A typical value of S/N for a telephone
line is 20 dB. What is the channel capacity?

Solution
SNR = 20 dB
20 = 10 log10 (S/N) S/N = 100
W = 2,800 Hz

S
S

C W log 2 1 bps 3.32 W log 10 1 bps


N
N

3.32 (2800) log10 1 100

C = 18,632 bps
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CHAPTER 3
TRANSMISSION MEDIA

There are two basic categories of transmission media: guided and unguided media.
Guided transmission media use cabling system that guides the data signals along a specific
path. Data signals are bound by the cabling system. Guided media is also known as bound
media. Cabling is meant in a generic sense, and is not meant to be interpreted as copper
wire cabling only.
Unguided transmission media consists of a means for the data signals to travel but nothing
to guide them along a specific path. The data signals are not bound to a cabling media and are
therefore often called unbound media.
Transmission Media: Guided
There four basic types of guided media:
a. Open Wire
b. Twisted Pair
c. Coaxial Cable
d. Optical Fibre
Open Wire
Open wire is traditionally used to describe the electrical wire strung along power poles. There
is a single wire strung between poles. No shielding or protection from noise interference is
used. We are going to extend the traditional definition of open wire to include any data signal
path without shielding or protection from noise interference. This can include multi conductor
cables or single wires. This medium is susceptible to a large degree of noise and interference
and consequently is not acceptable for data transmission except for short distances under
20 ft.

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Twisted Pair
The wires in twisted pair cabling are twisted together in pairs. Each pair consists of a wire
used for the positive data signal and a wire used for the negative data signal. Any noise that
appears on one wire of the pair will also occur on the other wire. Since the wires have
opposite polarities, they are 180 degrees out of phase. When noise appears on both wires, it
cancels or nulls itself out at the receiving end. Twisted pair cables are most effectively used in
systems that use a balanced line method of transmission: polar line coding (Manchester
encoding) as opposed to unipolar line coding.

Unshielded Twisted Pair


The degree of reduction in noise interference is determined specifically by the number of
turns per foot. Increasing the number of turns per foot reduces the noise interference. To
further improve noise rejection, a foil or wire braid shield is woven around the twisted
pairs. This shield can be woven around individual pairs or around a multi-pair conductor
(several pairs).

Shielded Twisted Pair


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Cables with a shield are called shielded twisted pair and are commonly abbreviated STP.
Cables without a shield are called unshielded twisted pair or UTP. Twisting the wires together
results in a characteristic impedance for the cable. Typical impedance for UTP is 100 Ohm for
Ethernet 10BaseT cable.
UTP or unshielded twisted pair cable is used in Ethernet 10BaseT and can also be used with
Token Ring. It uses the RJ line of connectors (RJ45, RJ11, etc..).
STP or shielded twisted pair is used with the traditional Token Ring cabling or ICS-IBM
Cabling System. It requires a custom connector. IBM STP (shielded twisted pair) has a
characteristic impedance of 150 Ohm.
Coaxial Cable
Coaxial cable consists of two conductors. The inner conductor is held inside an insulator with
the other conductor woven around it providing a shield. An insulating protective coating
called a jacket covers the outer conductor.

Coaxial Cable
The outer shield protects the inner conductor from outside electrical signals. The distance
between the outer conductor (shield) and inner conductor, plus the type of material used for
insulating the inner conductor determine the cable properties or impedance. Typical
impedances for coaxial cables are 75 Ohms for TV cable, 50 Ohms for Ethernet Thinnet and
Thicknet. The excellent control of the impedance characteristics of the cable allow higher data
rates to be transferred than with twisted pair cable.
Optical fibre
Optical fibre consists of thin glass fibres that can carry information at frequencies in the
visible light spectrum and beyond. The typical optical fibre consists of a very narrow strand of
glass called the core. Around the core is a concentric layer of glass called the cladding. A
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typical core diameter is 62.5 microns (1 micron = 10-6 m). Typically Cladding has a diameter
of 125 microns. Coating the cladding is a protective coating consisting of plastic, it is called
the Jacket.
Fibre Optic Cables

Just as standard electric cables come in a variety of sizes, shapes, and types, fibre optic cables
are available in different configurations. The simplest cable is just a single strand of fibre,
whereas complex cables are made up of multiple fibres with different layers and other
elements.

The portion of a fibre optic cable (core) that carries the light is made from either glass or
plastic. Another name for glass is silica. Special techniques have been developed to create
nearly perfect optical glass or plastic, which is transparent to light. Such materials can carry
light over a long distance. Glass has superior optical characteristics over plastic. However,
glass is far more expensive and more fragile than plastic. Although the plastic is less
expensive and more flexible, its attenuation of light is greater. For a given intensity, light will
travel a greater distance in glass than in plastic. For very long distance transmission, glass is
certainly preferred. For shorter distances, plastic is much more practical.

All fibres consist of a number of substructures including:

A core, which carries most of the light, surrounded by


A cladding, which bends the light and confines it to the core, surrounded by
A substrate layer (in some fibres) of glass which does not carry light, but adds to the diameter
and strength of the fibre, covered by
A primary buffer coating, this provides the first layer of mechanical protection, covered by
A secondary buffer coating, this protects the relatively fragile primary coating.
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taking a longer path. The result is that the light rays arrive at the receiver at different times.
The signal becomes longer than the original signal. LED light sources are used. Typical Core:
62.5 microns.
Graded index has a gradual change in the core's refractive index. This causes the light rays to
be gradually bent back into the core path. This is represented by a curved reflective path in the
attached drawing. The result is a better receive signal than with step index. LED light sources
are used. Typical Core: 62.5 microns.
Note: Both step index and graded index allow more than one light source to be used (different
colours simultaneously), so multiple channels of data can be run at the same time!
Single mode has separate distinct refractive indexes for the cladding and core. The light ray
passes through the core with relatively few reflections off the cladding. Single mode is used
for a single source of light (one colour) operation. It requires a laser and the core is very
small: 9 microns.
Basic Construction of Fibre-Optic Cables
There are two basic ways of classifying fibre optic cables. The first way is an indication of
how the index of refraction varies across the cross section of the cable. The second way of
classification is by mode. Mode refers to the various paths that the light rays can take in
passing through the fibre. Usually these two methods of classification are combined to define
the types of cable. There are two basic ways of defining the index of refraction variation
across a cable. These are step index and graded index. Step index refers to the fact that there
is a sharply defined step in the index of refraction where the fibre core and the cladding
interface. It means that the core has one constant index of refraction n1, while the cladding has
another constant index of refraction n2.
The other type of cable has a graded index. In this type of cable, the index of refraction of the
core is not constant. Instead, the index of refraction varies smoothly and continuously over the
diameter of the core. As you get closer to the centre of the core, the index of refraction
gradually increases, reaching a peak at the centre and then declining as the other outer edge of
the core is reached. The index of refraction of the cladding is constant.
Mode refers to the number of paths for the light rays in the cable. There are two
classifications: single mode and multimode. In single mode, light follows a single path
through the core. In multimode, the light takes many paths through the core.
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repetition rate. When this is done, proper operation occurs. But with pulses at a lower
frequency, less information can be handled.

2. Single Mode Cable


In a single mode, or mono-mode, step-index fibre cable the core is so small that the total
number modes or paths through the core are minimised and modal dispersion is essentially
eliminated. The typical core sizes are 5 to 15 m. The output pulse has essentially the same
duration as the input pulse (see Figure 6).

The single mode step index fibres are by far the best since the pulse repetition rate can be high
and the maximum amount of information can be carried. For very long distance transmission
and maximum information content, single-mode step-index fibre cables should be used.

The main problem with this type of cable is that because of its extremely small size, it is
difficult to make and is, therefore, very expensive. Handling, splicing, and making
interconnections are also more difficult. Finally, for proper operation an expensive, super
intense light source such as a laser must be used. For long distances, however, this is the type
of cable preferred.
Cross section

Index profile

Beam path

n1
n2

cladding

Input

Output

core

Figure 6

3. Multimode graded-index fibre cables


These cables have a several modes or paths of transmission through the cable, but they are
much more orderly and predictable. Figure 7 shows the typical paths of the light beams.
Because of the continuously varying index of refraction across the core, the light rays are bent
smoothly and converge repeatedly at points along the cable.

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850 nM - wavelength of light source

Typically 62.5/125 (core diameter/cladding diameter)

Multimode - can run many light sources.

Outdoor cable specifications:

Laser Light Source

1 dB/Km Attenuation (loses 1 dB of signal per kilometer)

1170 nM - wavelength of light source

Monomode (single mode)

Advantages of Optical Fibre:

Noise immunity: RFI and EMI immune (RFI - Radio Frequency Interference, EMI Electromagnetic Interference)

Security: cannot tap into cable.

Large Capacity due to BW (bandwidth)

No corrosion

Longer distances than copper wire

Smaller and lighter than copper wire

Faster transmission rate

Disadvantages of optical fibre:

Physical vibration will show up as signal noise!

Limited physical arc of cable. Bend it too much and it will break!

Difficult to splice

The cost of optical fibre is a trade-off between capacity and cost. At higher transmission
capacity, it is cheaper than copper. At lower transmission capacity, it is more expensive.

Media versus Bandwidth


The following table compares the usable bandwidth of the different guided transmission
media.

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Cable Type

Bandwidth

Open Cable

0 - 5 MHz

Twisted Pair

0 - 100 MHz

Coaxial Cable

0 - 600 MHz

Optical Fibre

0 - 1 GHz

Transmission Media: Unguided


Unguided transmission media is data signals that flow through the air. They are not guided or
bound to a channel to follow. They are classified by the type of wave propagation.
RF Propagation
There are three types of RF (radio frequency) propagation:

Ground Wave

Sky Wave

Line of Sight (LOS)

Ground wave propagation follows the curvature of the Earth. Ground waves have carrier
frequencies up to 2 MHz. AM radio is an example of ground wave propagation.

Sky wave propagation bounces off of the Earth's ionospheric layer in the upper atmosphere.
It is sometimes called double hop propagation. It operates in the frequency range of 30-85
MHz. Because it depends on the Earth's ionosphere, it changes with the weather and time of
day. The signal bounces off of the ionosphere and back to earth. Ham radios operate in this
range.
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Line of sight propagation transmits exactly in the line of sight. The receive station must be
in the view of the transmit station. It is sometimes called space waves or troposphere
propagation. It is limited by the curvature of the Earth for ground-based stations (100 km,
from horizon to horizon). Reflected waves can cause problems. Examples of line of sight
propagation are: FM radio, microwave and satellite.

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Radio Frequencies
The frequency spectrum operates from 0 Hz (DC) to gamma rays (1019 Hz).
Name

Frequency (Hertz)

Gamma Rays

1019

X-Rays

1017

Ultra-Violet Light

7.5 x 1015

Visible Light

4.3 x 1014

Infrared Light

3 x 1011

Examples

EHF - Extremely High Frequencies 30 GHz (Giga = 109)

Radar

SHF - Super High Frequencies

3 GHz

Satellite & Microwaves

UHF - Ultra High Frequencies

300 MHz (Mega = 106)

UHF TV (Ch. 14-83)

VHF - Very High Frequencies

30 MHz

FM & TV (Ch2 - 13)

HF - High Frequencies

3 MHz2

Short Wave Radio

MF - Medium Frequencies

300 kHz (kilo = 103)

AM Radio

LF Low Frequencies

30 kHz

Navigation

VLF - Very Low Frequencies

3 kHz

Submarine Communications

VF - Voice Frequencies

300 Hz

Audio

ELF - Extremely Low Frequencies 30 Hz

Power Transmission

Radio frequencies are in the range of 300 kHz to 10 GHz. We are seeing an emerging
technology called wireless LANs. Some use radio frequencies to connect the workstations
together, some use infrared technology.
Microwave
Microwave transmission is line of sight transmission. The transmit station must be in visible
contact with the receive station. This sets a limit on the distance between stations depending
on the local geography. Typically the line of sight due to the Earth's curvature is only 100 km
to the horizon! Repeater stations must be placed so the data signal can hop, skip and jump
across the country.

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Microwaves operate at high operating frequencies of 3 to 10 GHz. This allows them to carry
large quantities of data due to their large bandwidth.
Advantages:
a. They require no right of way acquisition between towers.
b. They can carry high quantities of information due to their high operating frequencies.
c. Low cost land purchase: each tower occupies only a small area.
d. High frequency/short wavelength signals require small antennae.
Disadvantages:
a. Attenuation by solid objects: birds, rain, snow and fog.
b. Reflected from flat surfaces like water and metal.
c. Diffracted (split) around solid objects.
d. Refracted by atmosphere, thus causing beam to be projected away from receiver.
Satellite
Satellites are transponders (units that receive on one frequency and retransmit on another) that
are set in geostationary orbits directly over the equator. These geostationary orbits are 36,000
km from the Earth's surface. At this point, the gravitational pull of the Earth and the
centrifugal force of Earth's rotation are balanced and cancel each other out. Centrifugal force
is the rotational force placed on the satellite that wants to fling it out into space.

The uplink is the transmitter of data to the satellite. The downlink is the receiver of data.
Uplinks and downlinks are also called Earth stations because they are located on the Earth.

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The footprint is the "shadow" that the satellite can transmit to, the shadow being the area that
can receive the satellite's transmitted signal.

Iridium Telecom System


The Iridium Telecom System is a new satellite system that will be the largest private
aerospace project. It is a mobile telecom system intended to compete with cellular phones. It
relies on satellites in lower Earth orbit (LEO). The satellites will orbit at an altitude of 900 10,000 km in a polar, non-stationary orbit. Sixty-six satellites are planned. The user's handset
will require less power and will be cheaper than cellular phones. There will be 100% coverage
of the Earth.

Unfortunately, although the Iridium project was planned for 1996-1998, with 1.5 million
subscribers by end of the decade, it looked very financially unstable.

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4PSK characteristics
Dibit

Phase

00
01
10
11

0o
90o
180o
270o

01

10

00

11
Constellation diagram

8PSK Characteristics
Tribit

Phase

000
001
010
011
100
101
110
111

0o
45o
90o
135o
180o
225o
270o
315o

010
001

011

000

100

111

101
110

Bit rate of 8-PSk is three as that of 2-PSK

Constellation diagram

Bandwidth for PSK


The min. BW required for PSK transmission is the same as that required for ASK
transmission.
BW = Nbaud
Ex: Find the bandwidth for a 4-PSK signal transmitting at 2,000 bps. Transmission is in halfduplex mode.
Solution
For 4-PSK the baud rate is half of the bit rate.
The baud rate is therefore 1,000. A PSK signal requires a bandwidth equal to its baud rate.
Therefore the bandwidth is 1,000 Hz.
Ex: Given a bandwidth of 5,000 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 5,000. But
in 8-PSK, the bit rate is three times the baud rate. So the bit rate is 15,000 bps.

Quadrature Amplitude Modulation (QAM)


QAM means combining ASK and PSK in such a way that we have maximum contrast
between each bit, dibit, tribit, quadbit, and so on.
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Bit

Nbaud= N
0

Dibit

Nbit= N
1

Nbaud= N
0

Tribit
0

Nbit= 3N
0

Nbaud= N
0

Nbit= 2N

Nbaud= N

Quadbit
0

Nbit= 4N
1

Bit/ Baud Rate Comparison


Modulation

Units

Bits/Bauds

Baud Rate

Bit Rate

ASK, FSK, 2-PSK

Bit

4-PSK, 4-QAM

Dibit

2N

8-PSK, 8-QAM

Tribit

3N

16-QAM

Quadbit

4N

32-QAM

Pentabit

5N

64-QAM

Hexabit

6N

128-QAM

Septabit

7N

256-QAM

Octabit

8N

Ex
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 points 45o apart. Since 23 = 8, three bits are
transmitted with each signal element. Therefore, the baud rate is
4800/3 = 1600 baud
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TRANSMISSION CODES

Transmission Codes
Binary-Coded Decimal (also called 8421 BCD)
In BCD, four bits are used to encode one decimal character. Four bits give 16 binary
combinations. Since there are 10 decimal characters, 0 through 9, only 10 of the 16 possible
combinations are necessary for encoding in BCD. The remaining 6 combinations are said to
be invalid.

Decimal

BCD

0000

0001

0010

1010

0011

1011

0100

1100

0101

1101

0110

1110

0111

1111

1000

1001

Not valid in BCD

Ex
Convert 36710 to BCD
Solution
36710 = 0011 0110 0111
Ex
Convert 124910 to BCD
Solution
124910 = 0001 0010 0100 1001
Ex
Convert 5810 to BCD
Solution
5810 = 0101 1000
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BCD Addition:
Straight binary addition is performed as long as the result does not exceed a decimal value
of 9.
Ex
Add the decimal numbers 3 and 4 in BCD
Solution
3

0011

+4

0100

0111

Ex
Add the decimal numbers 63 and 24 in BCD
Solution
63

0110 0011

+ 24

0010 0100

87

1000 0111

When the sum of two numbers exceeds 9, an invalid BCD number is obtained. The invalid
number can be converted to a valid number by adding 0110 (6) to it.
Ex
Add the decimal numbers 9 and 6 in BCD
Solution
9

1001

+6

0110

15

1111not valid in BCD


+ 0110add 6 for correction
0001 0101correct BCD number

Ex
Add the decimal numbers 46 and 79 in BCD
Solution
46

0100 0110

+ 79

0111 1001

125

1011 1111
+ 0110 0110
0001 0010 0101
1
2
5
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Excess-3 Code
Excess-3 code is very similar to 8421 BCD code. The only difference is that 3 is added to the
decimal before it is encoded into a four-word.

Decimal

BCD

Excess-3

0000

0011

0001

0100

0010

0101

0000

0011

0110

0001

0100

0111

0010

0101

1000

1101

0110

1001

1110

0111

1010

1111

1000

1011

1001

1100

Not valid in Excess-3

Ex
Add the decimal numbers 9 and 7 in Excess-3
Solution
9

1100

+7

1010

16

1 0110

Gray Code
The disadvantage of the previous codes is that several bits change state between adjacent
counts. The Gray code is unique in that successive counts result in only one bit change. For
example, 7 (0111) to 8 (1000) in binary, or BCD, all four bits change state. In Gray code,
however, 7 (0100) to 8 (1100) require a single bit change.
The switching noise generated by the associated circuits may be intolerable in some
environments. The same change with Gray code undergoes only a single bit change
consequently, less noise is generated. Shaft encoders used for receiver tuning dials often use
Gray code.
The Gray code is widely used for encoding the position of the rotary shaft and for data
transmission using PSK.

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Decimal

Binary

Gray Code

0000

0000

0001

0001

0010

0011

0011

0010

0100

0110

0101

0111

0110

0101

0111

0100

1000

1100

1001

1101

10

1010

1111

Binary-to-Gray Conversion
-

The given binary code is shifted to the right by one bit.

Discard the last bit (the LSB) from the obtained bits.

Exclusive-ORing the given and obtained bits result in the equivalent Gray code.
MSB

LSB
x

16th position

x
1st position

Ex
Compute the Gray code for the binary number 11010
Solution
Binary code

1 1 0 1 0

Gray code

1 1 0 1
1 0 1 1 1

Ex
Compute the Gray code for the binary number 10001101
Solution
Binary code

1 0 0 0 1 1 0 1

Gray code

1 0 0 0 1 1 0
1 1 0 0 1 0 1 1

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Ex
1

digits

weights

___________________
8

results

+
13
Octal Numbers
The octal numbering system is used by computer programmers to represent binary
numbers in compact form. Also called base 8.
Octal numbers use 8 symbols: 0,1,2,3,4,5,6,7.

Octal weights
Position

Fifth

Fourth

Third

Second

First

Weight

84 (4096)

83 (512)

82 (64)

81 (8)

80 (1)

Ex
3

digits

512

64

weights

___________________
1,536 256

56

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+
1,849
Hexadecimal Numbers
Hexadecimal numbering system, like octal, is used by computer programmers to represent
binary numbers in compact form. Also called base 16.
Hexadecimal uses 16 symbols: 0,1,2,3,4,5,6,7,8,9,A,B,C,D,E,F.

Hexadecimal weights
Position

Fifth

Fourth

Third

Second

First

Weight

164 (65,536)

163 (4,096)

162 (256)

161 (16)

160 (1)

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Ex
3

digits

4,096

256

16

weights

_____________________
12,288 1,024 112 1

results

+
13,425
Decimal

Binary

Octal

Hexadecimal

0000

0001

0010

0011

0100

0101

0110

0111

1000

10

1001

11

10

1010

12

11

1011

13

12

1100

14

13

1101

15

14

1110

16

15

1111

17

Transformations
- From Other Systems to Decimal
a) From Binary to Decimal
1

Binary

64

32

16

weights

64

weighted results

+
78

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b) Hexadecimal to Decimal

Hexadecimal

16

weights

64

14

weighted results

Octal

64

weights

64

weighted results

+
78
c) Octal to Decimal

+
78
- From Binary to Octal or Hexadecimal

1 0 0 1 1 1 0

Group by three
From right to
left

Group by four
from right to left

1 0 0

1 1 1 0

Octal

Hexadecimal

- From Octal or Hexadecimal to Binary

Octal
1

Hexadecimal
6

1 0 0 1 1 1 0

1 0 0

Binary

1 1 1 0
Binary

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- From Decimal to Other Systems
a) From Decimal to Binary

Decimal

19

39

78

b) From Decimal to Octal

Binary
Code

(divide by 8)
1

78

c) From Decimal to Hexadecimal

Binary
Code

(divide by 16)
4

78

Remainder

Remainder

Remainder

(divide by 2)

Binary
Code

Morse Code
Morse code is one of the oldest electrical transmission codes. The digital code system is made
up of a series of dots and dashes, representing the alphabet and decimal numbers system. A
dash is three times the duration of a dot.

A
B
C
D
.
.
1
2
.
ASCII Code
The American Standard Code for Information Interchange (ASCII) is the most widely used
alphanumeric code for transmission and data processing.
ASCII is a seven-bit code that can be represented by two hexadecimal characters for
simplicity. The MS hexadecimal character in this case never exceed 7

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Ex
What is the ASCII code for the letter H (uppercase) in binary and hexadecimal?

Solution
Letter H is located in column 4 and row 8.
Binary:

100 1000

Hexadecimal: $ 48

Ex
What is the ASCII code for the letter k (lowercase) in binary and hexadecimal?

Solution
Letter k is located in column 6 and row B:
Binary:

110 1011

Hexadecimal: $ 6B

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UNIT-5
ADVANCED MOBILE PHONE SYSTEM
Advanced Mobile Phone Service (AMPS) is a standard system
for analog signal cellular telephone service in the United States
and is also used in other countries.
It is based on the initial electromagnetic radiation spectrum
allocation for cellular service by the Federal Communications
Commission (FCC) in 1970. Introduced by AT&T in 1983,
AMPS became one of the most widely deployed cellular system
in the United States.

AMPS allocates frequency ranges within the 800 and 900


Megahertz (MHz) spectrum to cellular telephone. Each service
provider can use half of the 824-849 MHz range for receiving
signals from cellular phones and half the 869-894 MHz range
for transmitting to cellular phones.

The bands are divided into 30 kHz sub-bands, called channels.


The receiving channels are called reverse channels and the
sending channels are called forward channels. The division of
the spectrum into sub-band channels is achieved by using
frequency division multiple access (FDMA).
The signals received from a transmitter cover an area called a
cell. As a user moves out of the cell's area into an adjacent cell,
the user begins to pick up the new cell's signals without any
noticeable transition.

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The signals in the adjacent cell are sent and received on


different channels than the previous cell's signals to so that the
signals don't interfere with each other.
The analog service of AMPS has been updated with digital
cellular service by adding to FDMA a further subdivision of
each channel using time division multiple access (TDMA). This
service is known as digital AMPS (D-AMPS). Although AMPS
and D-AMPS originated for the North American cellular
telephone market , they are now used worldwide with over 74
million subscribers, according to Ericsson, one of the major
cellular phone manufacturers.
GSM

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GSM MSs consist of:

Mobile Equipment

Subscriber Identity Module

FUNCTIONS OF MS
Voice and data transmission & receipt
Frequency and time synchronization
Monitoring of power and signal quality of the surrounding cells
Provision of location updates even during inactive state

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complete orthonormal set. The data signal is also binary and the time
multiplication is achieved with a simple XOR function. This is
usually a Gilbert cell mixer in the circuitry.
Synchronous CDMA exploits mathematical properties of
orthogonality between vectors representing the data strings. For
example, binary string 1011 is represented by the vector (1, 0, 1, 1).
Vectors can be multiplied by taking their dot product, by summing the
products of their respective components (for example, if u = (a, b) and
v = (c, d), then their dot product uv = ac + bd). If the dot product is
zero, the two vectors are said to be orthogonal to each other. Some
properties of the dot product aid understanding of how W-CDMA
works. If vectors a and b are orthogonal, then
and:

Each user in synchronous CDMA uses a code orthogonal to the


others' codes to modulate their signal. An example of four mutually
orthogonal digital signals is shown in the figure. Orthogonal codes
have a cross-correlation equal to zero; in other words, they do not
interfere with each other. In the case of IS-95 64 bit Walsh codes are
used to encode the signal to separate different users. Since each of the
64 Walsh codes are orthogonal to one another, the signals are
channelized into 64 orthogonal signals. The following example
demonstrates how each user's signal can be encoded and decoded.

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Example

An example of four mutually orthogonal digital signals.


Start with a set of vectors that are mutually orthogonal. (Although
mutual orthogonality is the only condition, these vectors are usually
constructed for ease of decoding, for example columns or rows from
Walsh matrices.) An example of orthogonal functions is shown in the
picture on the right. These vectors will be assigned to individual users
and are called the code, chip code, or chipping code. In the interest of
brevity, the rest of this example uses codes, v, with only two bits.
Each user is associated with a different code, say v. A 1 bit is
represented by transmitting a positive code, v, and a 0 bit is
represented by a negative code, v. For example, if v = (v0, v1) = (1,
1) and the data that the user wishes to transmit is (1, 0, 1, 1), then the
transmitted
symbols
would
be
(v, v, v, v) = (v0, v1, v0, v1, v0, v1, v0, v1) = (1, 1, 1, 1, 1, 1, 1,
1). For the purposes of this article, we call this constructed vector the
transmitted vector.

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Each sender has a different, unique vector v chosen from that set, but
the construction method of the transmitted vector is identical.
Now, due to physical properties of interference, if two signals at a
point are in phase, they add to give twice the amplitude of each signal,
but if they are out of phase, they subtract and give a signal that is the
difference of the amplitudes. Digitally, this behaviour can be
modelled by the addition of the transmission vectors, component by
component.
If sender0 has code (1, 1) and data (1, 0, 1, 1), and sender1 has code
(1, 1) and data (0, 0, 1, 1), and both senders transmit simultaneously,
then this table describes the coding steps:
Step Encode sender0

Encode sender1

code0 = (1, 1), data0 = (1, 0, code1 = (1, 1), data1 = (0, 0, 1,
1, 1)
1)

encode0 = 2(1, 0, 1, 1) (1, 1, encode1 = 2(0, 0, 1, 1) (1, 1, 1,


1, 1) = (1, 1, 1, 1)
1) = (1, 1, 1, 1)

signal0 = encode0 code0 signal1 = encode1 code1


= (1, 1, 1, 1) (1, 1) = (1, 1, 1, 1) (1, 1)
= (1, 1, 1, 1, 1, 1, 1, 1)
= (1, 1, 1, 1, 1, 1, 1, 1)

Because signal0 and signal1 are transmitted at the same time into the
air, they add to produce the raw signal:
(1, 1, 1, 1, 1, 1, 1, 1) + (1, 1, 1, 1, 1, 1, 1, 1) = (0, 2,
2, 0, 2, 0, 2, 0)
This raw signal is called an interference pattern. The receiver then
extracts an intelligible signal for any known sender by combining the
sender's code with the interference pattern, the receiver combines it
with the codes of the senders. The following table explains how this
works and shows that the signals do not interfere with one another:
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Step Decode sender0

Decode sender1

code0 = (1, 1), signal = (0, 2, code1 = (1, 1), signal = (0, 2,
2, 0, 2, 0, 2, 0)
2, 0, 2, 0, 2, 0)

decode0 = pattern.vector0

decode0 = ((0, 2), (2, 0), (2, decode1 = ((0, 2), (2, 0), (2,
0), (2, 0)).(1, 1)
0), (2, 0)).(1, 1)

decode0 = ((0 + 2), (2 + 0), (2 decode1 = ((0 2), (2 + 0), (2


+ 0), (2 + 0))
+ 0), (2 + 0))

data0=(2, 2, 2, 2), meaning (1, data1=(2, 2, 2, 2), meaning


0, 1, 1)
(0, 0, 1, 1)

decode1 = pattern.vector1

Further, after decoding, all values greater than 0 are interpreted as 1


while all values less than zero are interpreted as 0. For example, after
decoding, data0 is (2, 2, 2, 2), but the receiver interprets this as (1, 0,
1, 1). Values of exactly 0 means that the sender did not transmit any
data, as in the following example:
Assume signal0 = (1, 1, 1, 1, 1, 1, 1, 1) is transmitted alone. The
following table shows the decode at the receiver:
Step Decode sender0

Decode sender1

code0 = (1, 1), signal = (1, 1, code1 = (1, 1), signal = (1, 1,
1, 1, 1, 1, 1, 1)
1, 1, 1, 1, 1, 1)

decode0 = pattern.vector0

decode0 = ((1, 1), (1, 1), (1, decode1 = ((1, 1), (1, 1), (1,
1), (1, 1)).(1, 1)
1), (1, 1)).(1, 1)

decode0 = ((1 + 1), (1 1),(1 + decode1 = ((1 1), (1 + 1),(1

decode1 = pattern.vector1

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1), (1 + 1))
4

1), (1 1))

data0 = (2, 2, 2, 2), meaning data1 = (0, 0, 0, 0), meaning no


(1, 0, 1, 1)
data

When the receiver attempts to decode the signal using sender1's code,
the data is all zeros, therefore the cross correlation is equal to zero and
it is clear that sender1 did not transmit any data.
Asynchronous CDMA
See also: Direct-sequence spread spectrum and near-far problem
When mobile-to-base links cannot be precisely coordinated,
particularly due to the mobility of the handsets, a different approach is
required. Since it is not mathematically possible to create signature
sequences that are both orthogonal for arbitrarily random starting
points and which make full use of the code space, unique "pseudorandom" or "pseudo-noise" (PN) sequences are used in asynchronous
CDMA systems. A PN code is a binary sequence that appears random
but can be reproduced in a deterministic manner by intended
receivers. These PN codes are used to encode and decode a user's
signal in Asynchronous CDMA in the same manner as the orthogonal
codes in synchronous CDMA (shown in the example above). These
PN sequences are statistically uncorrelated, and the sum of a large
number of PN sequences results in multiple access interference (MAI)
that is approximated by a Gaussian noise process (following the
central limit theorem in statistics). Gold codes are an example of a PN
suitable for this purpose, as there is low correlation between the
codes. If all of the users are received with the same power level, then
the variance (e.g., the noise power) of the MAI increases in direct
proportion to the number of users. In other words, unlike synchronous
CDMA, the signals of other users will appear as noise to the signal of
interest and interfere slightly with the desired signal in proportion to
number of users.
All forms of CDMA use spread spectrum process gain to allow
receivers to partially discriminate against unwanted signals. Signals
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encoded with the specified PN sequence (code) are received, while


signals with different codes (or the same code but a different timing
offset) appear as wideband noise reduced by the process gain.
Since each user generates MAI, controlling the signal strength is an
important issue with CDMA transmitters. A CDM (synchronous
CDMA), TDMA, or FDMA receiver can in theory completely reject
arbitrarily strong signals using different codes, time slots or frequency
channels due to the orthogonality of these systems. This is not true for
Asynchronous CDMA; rejection of unwanted signals is only partial.
If any or all of the unwanted signals are much stronger than the
desired signal, they will overwhelm it. This leads to a general
requirement in any asynchronous CDMA system to approximately
match the various signal power levels as seen at the receiver. In
CDMA cellular, the base station uses a fast closed-loop power control
scheme to tightly control each mobile's transmit power.
Advantages of asynchronous CDMA over other techniques
Efficient practical utilization of the fixed frequency spectrum
In theory CDMA, TDMA and FDMA have exactly the same spectral
efficiency but practically, each has its own challenges power control
in the case of CDMA, timing in the case of TDMA, and frequency
generation/filtering in the case of FDMA.
TDMA systems must carefully synchronize the transmission times of
all the users to ensure that they are received in the correct time slot
and do not cause interference. Since this cannot be perfectly
controlled in a mobile environment, each time slot must have a guardtime, which reduces the probability that users will interfere, but
decreases the spectral efficiency. Similarly, FDMA systems must use
a guard-band between adjacent channels, due to the unpredictable
doppler shift of the signal spectrum because of user mobility. The
guard-bands will reduce the probability that adjacent channels will
interfere, but decrease the utilization of the spectrum.

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Flexible allocation of resources


Asynchronous CDMA offers a key advantage in the flexible
allocation of resources i.e. allocation of a PN codes to active users. In
the case of CDM (synchronous CDMA), TDMA, and FDMA the
number of simultaneous orthogonal codes, time slots and frequency
slots respectively are fixed hence the capacity in terms of number of
simultaneous users is limited. There are a fixed number of orthogonal
codes, time slots or frequency bands that can be allocated for CDM,
TDMA, and FDMA systems, which remain underutilized due to the
bursty nature of telephony and packetized data transmissions. There is
no strict limit to the number of users that can be supported in an
asynchronous CDMA system, only a practical limit governed by the
desired bit error probability, since the SIR (Signal to Interference
Ratio) varies inversely with the number of users. In a bursty traffic
environment like mobile telephony, the advantage afforded by
asynchronous CDMA is that the performance (bit error rate) is
allowed to fluctuate randomly, with an average value determined by
the number of users times the percentage of utilization. Suppose there
are 2N users that only talk half of the time, then 2N users can be
accommodated with the same average bit error probability as N users
that talk all of the time. The key difference here is that the bit error
probability for N users talking all of the time is constant, whereas it is
a random quantity (with the same mean) for 2N users talking half of
the time.
In other words, asynchronous CDMA is ideally suited to a mobile
network where large numbers of transmitters each generate a
relatively small amount of traffic at irregular intervals. CDM
(synchronous CDMA), TDMA, and FDMA systems cannot recover
the underutilized resources inherent to bursty traffic due to the fixed
number of orthogonal codes, time slots or frequency channels that can
be assigned to individual transmitters. For instance, if there are N time
slots in a TDMA system and 2N users that talk half of the time, then
half of the time there will be more than N users needing to use more
than N time slots. Furthermore, it would require significant overhead
to continually allocate and deallocate the orthogonal code, time slot or
frequency channel resources. By comparison, asynchronous CDMA
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transmitters simply send when they have something to say, and go off
the air when they don't, keeping the same PN signature sequence as
long as they are connected to the system.
Spread-spectrum characteristics of CDMA
Most modulation schemes try to minimize the bandwidth of this
signal since bandwidth is a limited resource. However, spread
spectrum techniques use a transmission bandwidth that is several
orders of magnitude greater than the minimum required signal
bandwidth. One of the initial reasons for doing this was military
applications including guidance and communication systems. These
systems were designed using spread spectrum because of its security
and resistance to jamming. Asynchronous CDMA has some level of
privacy built in because the signal is spread using a pseudo-random
code; this code makes the spread spectrum signals appear random or
have noise-like properties. A receiver cannot demodulate this
transmission without knowledge of the pseudo-random sequence used
to encode the data. CDMA is also resistant to jamming. A jamming
signal only has a finite amount of power available to jam the signal.
The jammer can either spread its energy over the entire bandwidth of
the signal or jam only part of the entire signal.[9]
CDMA can also effectively reject narrow band interference. Since
narrow band interference affects only a small portion of the spread
spectrum signal, it can easily be removed through notch filtering
without much loss of information. Convolution encoding and
interleaving can be used to assist in recovering this lost data. CDMA
signals are also resistant to multipath fading. Since the spread
spectrum signal occupies a large bandwidth only a small portion of
this will undergo fading due to multipath at any given time. Like the
narrow band interference this will result in only a small loss of data
and can be overcome.
Another reason CDMA is resistant to multipath interference is
because the delayed versions of the transmitted pseudo-random codes
will have poor correlation with the original pseudo-random code, and
will thus appear as another user, which is ignored at the receiver. In
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other words, as long as the multipath channel induces at least one chip
of delay, the multipath signals will arrive at the receiver such that they
are shifted in time by at least one chip from the intended signal. The
correlation properties of the pseudo-random codes are such that this
slight delay causes the multipath to appear uncorrelated with the
intended signal, and it is thus ignored.
Some CDMA devices use a rake receiver, which exploits multipath
delay components to improve the performance of the system. A rake
receiver combines the information from several correlators, each one
tuned to a different path delay, producing a stronger version of the
signal than a simple receiver with a single correlation tuned to the
path delay of the strongest signal.[10]
Frequency reuse is the ability to reuse the same radio channel
frequency at other cell sites within a cellular system. In the FDMA
and TDMA systems frequency planning is an important consideration.
The frequencies used in different cells must be planned carefully to
ensure signals from different cells do not interfere with each other. In
a CDMA system, the same frequency can be used in every cell,
because channelization is done using the pseudo-random codes.
Reusing the same frequency in every cell eliminates the need for
frequency planning in a CDMA system; however, planning of the
different pseudo-random sequences must be done to ensure that the
received signal from one cell does not correlate with the signal from a
nearby cell.[11]
Since adjacent cells use the same frequencies, CDMA systems have
the ability to perform soft hand offs. Soft hand offs allow the mobile
telephone to communicate simultaneously with two or more cells. The
best signal quality is selected until the hand off is complete. This is
different from hard hand offs utilized in other cellular systems. In a
hard hand off situation, as the mobile telephone approaches a hand
off, signal strength may vary abruptly. In contrast, CDMA systems
use the soft hand off, which is undetectable and provides a more
reliable and higher quality signal.[11]

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Collaborative CDMA
In a recent study, a novel collaborative multi-user transmission and
detection scheme called Collaborative CDMA[12] has been
investigated for the uplink that exploits the differences between users
fading channel signatures to increase the user capacity well beyond
the spreading length in multiple access interference (MAI) limited
environment. The authors show that it is possible to achieve this
increase at a low complexity and high bit error rate performance in
at fading channels, which is a major research challenge for
overloaded CDMA systems. In this approach, instead of using one
sequence per user as in conventional CDMA, the authors group a
small number of users to share the same spreading sequence and
enable group spreading and despreading operations. The new
collaborative multi-user receiver consists of two stages: group multiuser detection (MUD) stage to suppress the MAI between the groups
and a low complexity maximum-likelihood detection stage to recover
jointly the co-spread users data using minimum Euclidean distance
measure and users channel gain coefcients. In CDM signal security
is high.
HAND OFF
In cellular telecommunications, the term handover or handoff refers
to the process of transferring an ongoing call or data session from one
channel connected to the core network to another channel. In satellite
communications it is the process of transferring satellite control
responsibility from one earth station to another without loss or
interruption of service.
Purpose
In telecommunications there may be different reasons why a handover
might be conducted:

when the phone is moving away from the area covered by one
cell and entering the area covered by another cell the call is
transferred to the second cell in order to avoid call termination
when the phone gets outside the range of the first cell;

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when the capacity for connecting new calls of a given cell is


used up and an existing or new call from a phone, which is
located in an area overlapped by another cell, is transferred to
that cell in order to free-up some capacity in the first cell for
other users, who can only be connected to that cell;
in non-CDMA networks when the channel used by the phone
becomes interfered by another phone using the same channel in
a different cell, the call is transferred to a different channel in
the same cell or to a different channel in another cell in order to
avoid the interference;
again in non-CDMA networks when the user behaviour
changes, e.g. when a fast-travelling user, connected to a large,
umbrella-type of cell, stops then the call may be transferred to a
smaller macro cell or even to a micro cell in order to free
capacity on the umbrella cell for other fast-traveling users and to
reduce the potential interference to other cells or users (this
works in reverse too, when a user is detected to be moving faster
than a certain threshold, the call can be transferred to a larger
umbrella-type of cell in order to minimize the frequency of the
handovers due to this movement);
in CDMA networks a handover (see further down) may be
induced in order to reduce the interference to a smaller
neighboring cell due to the "near-far" effect even when the
phone still has an excellent connection to its current cell;
etc.

The most basic form of handover is when a phone call in progress is


redirected from its current cell (called source) to a new cell (called
target). In terrestrial networks the source and the target cells may be
served from two different cell sites or from one and the same cell site
(in the latter case the two cells are usually referred to as two sectors
on that cell site). Such a handover, in which the source and the target
are different cells (even if they are on the same cell site) is called
inter-cell handover. The purpose of inter-cell handover is to maintain
the call as the subscriber is moving out of the area covered by the
source cell and entering the area of the target cell.

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A special case is possible, in which the source and the target are one
and the same cell and only the used channel is changed during the
handover. Such a handover, in which the cell is not changed, is called
intra-cell handover. The purpose of intra-cell handover is to change
one channel, which may be interfered or fading with a new clearer or
less fading channel.
Types of handover
In addition to the above classification of inter-cell and intra-cell
classification of handovers, they also can be divided into hard and soft
handovers:

A hard handover is one in which the channel in the source cell is


released and only then the channel in the target cell is engaged.
Thus the connection to the source is broken before or 'as' the
connection to the target is madefor this reason such handovers
are also known as break-before-make. Hard handovers are
intended to be instantaneous in order to minimize the disruption
to the call. A hard handover is perceived by network engineers
as an event during the call. It requires the least processing by the
network providing service. When the mobile is between base
stations, then the mobile can switch with any of the base
stations, so the base stations bounce the link with the mobile
back and forth. This is called ping-ponging.
A soft handover is one in which the channel in the source cell is
retained and used for a while in parallel with the channel in the
target cell. In this case the connection to the target is established
before the connection to the source is broken, hence this
handover is called make-before-break. The interval, during
which the two connections are used in parallel, may be brief or
substantial. For this reason the soft handover is perceived by
network engineers as a state of the call, rather than a brief event.
Soft handovers may involve using connections to more than two
cells: connections to three, four or more cells can be maintained
by one phone at the same time. When a call is in a state of soft
handover, the signal of the best of all used channels can be used
for the call at a given moment or all the signals can be combined
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to produce a clearer copy of the signal. The latter is more


advantageous, and when such combining is performed both in
the downlink (forward link) and the uplink (reverse link) the
handover is termed as softer. Softer handovers are possible
when the cells involved in the handovers have a single cell site.
Comparison of handovers
An advantage of the hard handover is that at any moment in time one
call uses only one channel. The hard handover event is indeed very
short and usually is not perceptible by the user. In the old analog
systems it could be heard as a click or a very short beep; in digital
systems it is unnoticeable. Another advantage of the hard handoff is
that the phone's hardware does not need to be capable of receiving
two or more channels in parallel, which makes it cheaper and simpler.
A disadvantage is that if a handover fails the call may be temporarily
disrupted or even terminated abnormally. Technologies which use
hard handovers, usually have procedures which can re-establish the
connection to the source cell if the connection to the target cell cannot
be made. However re-establishing this connection may not always be
possible (in which case the call will be terminated) and even when
possible the procedure may cause a temporary interruption to the call.
One advantage of the soft handovers is that the connection to the
source cell is broken only when a reliable connection to the target cell
has been established and therefore the chances that the call will be
terminated abnormally due to failed handovers are lower. However,
by far a bigger advantage comes from the mere fact that
simultaneously channels in multiple cells are maintained and the call
could only fail if all of the channels are interfered or fade at the same
time. Fading and interference in different channels are unrelated and
therefore the probability of them taking place at the same moment in
all channels is very low. Thus the reliability of the connection
becomes higher when the call is in a soft handover. Because in a
cellular network the majority of the handovers occur in places of poor
coverage, where calls would frequently become unreliable when their
channel is interfered or fading, soft handovers bring a significant
improvement to the reliability of the calls in these places by making
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the interference or the fading in a single channel not critical. This


advantage comes at the cost of more complex hardware in the phone,
which must be capable of processing several channels in parallel.
Another price to pay for soft handovers is use of several channels in
the network to support just a single call. This reduces the number of
remaining free channels and thus reduces the capacity of the network.
By adjusting the duration of soft handovers and the size of the areas in
which they occur, the network engineers can balance the benefit of
extra call reliability against the price of reduced capacity.
Possibility of handover
While theoretically speaking soft handovers are possible in any
technology, analog or digital, the cost of implementing them for
analog technologies is prohibitively high and none of the technologies
that were commercially successful in the past (e.g. AMPS, TACS,
NMT, etc.) had this feature. Of the digital technologies, those based
on FDMA also face a higher cost for the phones (due to the need to
have multiple parallel radio-frequency modules) and those based on
TDMA or a combination of TDMA/FDMA, in principle, allow not so
expensive implementation of soft handovers. However, none of the
2G (second-generation) technologies have this feature (e.g. GSM, DAMPS/IS-136, etc.). On the other hand, all CDMA based
technologies, 2G and 3G (third-generation), have soft handovers. On
one hand, this is facilitated by the possibility to design not so
expensive phone hardware supporting soft handovers for CDMA and
on the other hand, this is necessitated by the fact that without soft
handovers CDMA networks may suffer from substantial interference
arising due to the so-called near-far effect.
In all current commercial technologies based on FDMA or on a
combination of TDMA/FDMA (e.g. GSM, AMPS, IS-136/DAMPS,
etc.) changing the channel during a hard handover is realised by
changing the pair of used transmit/receive frequencies.

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Implementations
For the practical realisation of handoffs in a cellular network each cell
is assigned a list of potential target cells, which can be used for
handing-off calls from this source cell to them. These potential target
cells are called neighbours and the list is called neighbour list.
Creating such a list for a given cell is not trivial and specialised
computer tools are used. They implement different algorithms and
may use for input data from field measurements or computer
predictions of radio wave propagation in the areas covered by the
cells.
During a call one or more parameters of the signal in the channel in
the source cell are monitored and assessed in order to decide when a
handover may be necessary. The downlink (forward link) and/or
uplink (reverse link) directions may be monitored. The handover may
be requested by the phone or by the base station (BTS) of its source
cell and, in some systems, by a BTS of a neighbouring cell. The
phone and the BTSs of the neighbouring cells monitor each other
others' signals and the best target candidates are selected among the
neighbouring cells. In some systems, mainly based on CDMA, a
target candidate may be selected among the cells which are not in the
neighbour list. This is done in an effort to reduce the probability of
interference due to the aforementioned near-far effect.
In analog systems the parameters used as criteria for requesting a hard
handover are usually the received signal power and the received
signal-to-noise ratio (the latter may be estimated in an analog system
by inserting additional tones, with frequencies just outside the
captured voice-frequency band at the transmitter and assessing the
form of these tones at the receiver). In non-CDMA 2G digital systems
the criteria for requesting hard handover may be based on estimates of
the received signal power, bit error rate (BER) and block error/erasure
rate (BLER), received quality of speech (RxQual), distance between
the phone and the BTS (estimated from the radio signal propagation
delay) and others. In CDMA systems, 2G and 3G, the most common
criterion for requesting a handover is Ec/Io ratio measured in the pilot
channel (CPICH) and/or RSCP.
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In CDMA systems, when the phone in soft or softer handoff is


connected to several cells simultaneously, it processes the received in
parallel signals using a rake receiver. Each signal is processed by a
module called rake finger. A usual design of a rake receiver in mobile
phones includes three or more rake fingers used in soft handoff state
for processing signals from as many cells and one additional finger
used to search for signals from other cells. The set of cells, whose
signals are used during a soft handoff, is referred to as the active set.
If the search finger finds a sufficiently-strong signal (in terms of high
Ec/Io or RSCP) from a new cell this cell is added to the active set.
The cells in the neighbour list (called in CDMA neighbouring set) are
checked more frequently than the rest and thus a handoff with a
neighbouring cell is more likely, however a handoff with others cells
outside the neighbor list is also allowed (unlike in GSM, IS136/DAMPS, AMPS, NMT, etc.).

Satellite
From Wikipedia, the free encyclopedia
This article is about artificial satellites. For natural satellites, also
known as moons, see Natural satellite. For other uses, see Satellite
(disambiguation).
This article is outdated. Please update this article to reflect
recent events or newly available information. (December 2013)

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Play media

NASA's Earth-observing fleet as of June 2012.

An animation depicting the orbits of GPS satellites in medium Earth


orbit.

A full-size model of the Earth observation satellite ERS 2

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In the context of spaceflight, a satellite is an artificial object which


has been intentionally placed into orbit. Such objects are sometimes
called artificial satellites to distinguish them from natural satellites
such as the Moon.
The world's first artificial satellite, the Sputnik 1, was launched by the
Soviet Union in 1957. Since then, thousands of satellites have been
launched into orbit around the Earth. Some satellites, notably space
stations, have been launched in parts and assembled in orbit. Artificial
satellites originate from more than 50 countries and have used the
satellite launching capabilities of ten nations. A few hundred satellites
are currently operational, whereas thousands of unused satellites and
satellite fragments orbit the Earth as space debris. A few space probes
have been placed into orbit around other bodies and become artificial
satellites to the Moon, Mercury, Venus, Mars, Jupiter, Saturn, Vesta,
Eros, and the Sun.
Satellites are used for a large number of purposes. Common types
include military and civilian Earth observation satellites,
communications satellites, navigation satellites, weather satellites, and
research satellites. Space stations and human spacecraft in orbit are
also satellites. Satellite orbits vary greatly, depending on the purpose
of the satellite, and are classified in a number of ways. Well-known
(overlapping) classes include low Earth orbit, polar orbit, and
geostationary orbit.
About 6,600 satellites have been launched. The latest estimates are
that 3,600 remain in orbit.[1] Of those, about 1,000 are operational;[2][3]
the rest have lived out their useful lives and are part of the space
debris. Approximately 500 operational satellites are in low-Earth
orbit, 50 are in medium-Earth orbit (at 20,000 km), the rest are in
geostationary orbit (at 36,000 km).[4]
Satellites are propelled by rockets to their orbits. Usually the launch
vehicle itself is a rocket lifting off from a launch pad on land. In a
minority of cases satellites are launched at sea (from a submarine or a
mobile maritime platform) or aboard a plane (see air launch to orbit).

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Satellites are usually semi-independent computer-controlled systems.


Satellite subsystems attend many tasks, such as power generation,
thermal control, telemetry, attitude control and orbit control.
History of artificial satellites

Sputnik 1: The first artificial satellite to orbit Earth.


The first artificial satellite was Sputnik 1, launched by the Soviet
Union on October 4, 1957, and initiating the Soviet Sputnik program,
with Sergei Korolev as chief designer (there is a crater on the lunar far
side which bears his name). This in turn triggered the Space Race
between the Soviet Union and the United States.
Sputnik 1 helped to identify the density of high atmospheric layers
through measurement of its orbital change and provided data on
radio-signal distribution in the ionosphere. The unanticipated
announcement of Sputnik 1's success precipitated the Sputnik crisis in
the United States and ignited the so-called Space Race within the
Cold War.
Sputnik 2 was launched on November 3, 1957 and carried the first
living passenger into orbit, a dog named Laika.[9]
In May, 1946, Project RAND had released the Preliminary Design of
an Experimental World-Circling Spaceship, which stated, "A satellite
vehicle with appropriate instrumentation can be expected to be one of
the most potent scientific tools of the Twentieth Century."[10] The
United States had been considering launching orbital satellites since
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1945 under the Bureau of Aeronautics of the United States Navy. The
United States Air Force's Project RAND eventually released the
above report, but did not believe that the satellite was a potential
military weapon; rather, they considered it to be a tool for science,
politics, and propaganda. In 1954, the Secretary of Defense stated, "I
know of no American satellite program."[11]
On July 29, 1955, the White House announced that the U.S. intended
to launch satellites by the spring of 1958. This became known as
Project Vanguard. On July 31, the Soviets announced that they
intended to launch a satellite by the fall of 1957.
Following pressure by the American Rocket Society, the National
Science Foundation, and the International Geophysical Year, military
interest picked up and in early 1955 the Army and Navy were
working on Project Orbiter, two competing programs: the army's
which involved using a Jupiter C rocket, and the civilian/Navy
Vanguard Rocket, to launch a satellite. At first, they failed: initial
preference was given to the Vanguard program, whose first attempt at
orbiting a satellite resulted in the explosion of the launch vehicle on
national television. But finally, three months after Sputnik 2, the
project succeeded; Explorer 1 became the United States' first artificial
satellite on January 31, 1958.[12]
In June 1961, three-and-a-half years after the launch of Sputnik 1, the
Air Force used resources of the United States Space Surveillance
Network to catalog 115 Earth-orbiting satellites.[13]
Early satellites were constructed as "one-off" designs. With growth in
geosynchronous (GEO) satellite communication, multiple satellites
began to be built on single model platforms called satellite buses. The
first standardized satellite bus design was the HS-333 GEO commsat,
launched in 1972.
The largest artificial satellite currently orbiting the Earth is the
International Space Station.

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1U CubeSat ESTCube-1, developed mainly by the students from the


University of Tartu, carries out a tether deployment experiment on the
low Earth orbit.
Space Surveillance Network
Main article: United States Space Surveillance Network
The United States Space Surveillance Network (SSN), a division of
The United States Strategic Command, has been tracking objects in
Earth's orbit since 1957 when the Soviets opened the space age with
the launch of Sputnik I. Since then, the SSN has tracked more than
26,000 objects. The SSN currently tracks more than 8,000 man-made
orbiting objects. The rest have re-entered Earth's atmosphere and
disintegrated, or survived re-entry and impacted the Earth. The SSN
tracks objects that are 10 centimeters in diameter or larger; those now
orbiting Earth range from satellites weighing several tons to pieces of
spent rocket bodies weighing only 10 pounds. About seven percent
are operational satellites (i.e. ~560 satellites), the rest are space
debris.[14] The United States Strategic Command is primarily
interested in the active satellites, but also tracks space debris which
upon reentry might otherwise be mistaken for incoming missiles.
A search of the NSSDC Master Catalog at the end of October 2010
listed 6,578 satellites launched into orbit since 1957, the latest being
Chang'e 2, on 1 October 2010.[15]
Non-military satellite services
There are three basic categories of non-military satellite services:[16]

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Fixed satellite services


Fixed satellite services handle hundreds of billions of voice, data, and
video transmission tasks across all countries and continents between
certain points on the Earth's surface.
Mobile satellite systems
Mobile satellite systems help connect remote regions, vehicles, ships,
people and aircraft to other parts of the world and/or other mobile or
stationary communications units, in addition to serving as navigation
systems.
Scientific research satellites (commercial and noncommercial)
Scientific research satellites provide meteorological information, land
survey data (e.g. remote sensing), Amateur (HAM) Radio, and other
different scientific research applications such as earth science, marine
science, and atmospheric research.
Types

MILSTAR: A communication satellite

"Killer Satellites" are satellites that are designed to destroy


enemy warheads, satellites, and other space assets.
Astronomical satellites are satellites used for observation of
distant planets, galaxies, and other outer space objects.

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Biosatellites are satellites designed to carry living organisms,


generally for scientific experimentation.
Communications satellites are satellites stationed in space for
the purpose of telecommunications. Modern communications
satellites typically use geosynchronous orbits, Molniya orbits or
Low Earth orbits.
Miniaturized satellites are satellites of unusually low masses
and small sizes.[17] New classifications are used to categorize
these satellites: minisatellite (500100 kg), microsatellite (below
100 kg), nanosatellite (below 10 kg).[citation needed]
Navigational satellites are satellites which use radio time
signals transmitted to enable mobile receivers on the ground to
determine their exact location. The relatively clear line of sight
between the satellites and receivers on the ground, combined
with ever-improving electronics, allows satellite navigation
systems to measure location to accuracies on the order of a few
meters in real time.
Reconnaissance satellites are Earth observation satellite or
communications satellite deployed for military or intelligence
applications. Very little is known about the full power of these
satellites, as governments who operate them usually keep
information pertaining to their reconnaissance satellites
classified.
Earth observation satellites are satellites intended for nonmilitary uses such as environmental monitoring, meteorology,
map making etc. (See especially Earth Observing System.)
Tether satellites are satellites which are connected to another
satellite by a thin cable called a tether.
Weather satellites are primarily used to monitor Earth's
weather and climate.[18]
Recovery satellites are satellites that provide a recovery of
reconnaissance, biological, space-production and other payloads
from orbit to Earth.
Manned spacecraft (spaceships) are large satellites able to put
humans into (and beyond) an orbit, and return them to Earth.
Spacecraft including spaceplanes of reusable systems have

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major propulsion or landing facilities. They can be used as


transport to and from the orbital stations.

International Space Station as seen from Space

Space stations are man-made orbital structures that are


designed for human beings to live on in outer space. A space
station is distinguished from other manned spacecraft by its lack
of major propulsion or landing facilities. Space stations are
designed for medium-term living in orbit, for periods of weeks,
months, or even years.
A Skyhook is a proposed type of tethered satellite/ion powered
space station that serves as a terminal for suborbital launch
vehicles flying between the Earth and the lower end of the
Skyhook, as well as a terminal for spacecraft going to, or
arriving from, higher orbit, the Moon, or Mars, at the upper end
of the Skyhook.[19][20]

BLUETOOTH
This article is about a wireless technology standard. For the medieval
King of Denmark, see Harald Bluetooth.
Bluetooth

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Developed Bluetooth Special Interest


Group
by
Industry

Mobile personal
networks

area

Mobile phones, Personal


Compatible
computers,
Laptop
hardware
computers
Physical
range

Up to 60 metres[1]

Bluetooth is a wireless technology standard for exchanging data over


short distances (using short-wavelength UHF radio waves in the ISM
band from 2.4 to 2.485 GHz[2]) from fixed and mobile devices, and
building personal area networks (PANs). Invented by telecom vendor
Ericsson in 1994,[3] it was originally conceived as a wireless
alternative to RS-232 data cables. It can connect several devices,
overcoming problems of synchronization.
Bluetooth is managed by the Bluetooth Special Interest Group (SIG),
which has more than 20,000 member companies in the areas of
telecommunication, computing, networking, and consumer
electronics.[4] Bluetooth was standardized as IEEE 802.15.1, but the
standard is no longer maintained. The SIG oversees the development
of the specification, manages the qualification program, and protects
the trademarks.[5] To be marketed as a Bluetooth device, it must be
qualified to standards defined by the SIG.[6] A network of patents is
required to implement the technology, which is licensed only for that
qualifying device.

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Implementation
Bluetooth operates in the range of 24002483.5 MHz (including
guard bands). This is in the globally unlicensed (but not unregulated)
Industrial, Scientific and Medical (ISM) 2.4 GHz short-range radio
frequency band. Bluetooth uses a radio technology called frequencyhopping spread spectrum. The transmitted data are divided into
packets and each packet is transmitted on one of the 79 designated
Bluetooth channels. Each channel has a bandwidth of 1 MHz.
Bluetooth 4.0 uses 2 MHz spacing which allows for 40 channels. The
first channel starts at 2402 MHz and continues up to 2480 MHz in
1 MHz steps. It usually performs 1600 hops per second, with
Adaptive Frequency-Hopping (AFH) enabled.[12]
Originally, Gaussian frequency-shift keying (GFSK) modulation was
the only modulation scheme available; subsequently, since the
introduction of Bluetooth 2.0+EDR, /4-DQPSK and 8DPSK
modulation may also be used between compatible devices. Devices
functioning with GFSK are said to be operating in basic rate (BR)
mode where an instantaneous data rate of 1 Mbit/s is possible. The
term Enhanced Data Rate (EDR) is used to describe /4-DPSK and
8DPSK schemes, each giving 2 and 3 Mbit/s respectively. The
combination of these (BR and EDR) modes in Bluetooth radio
technology is classified as a "BR/EDR radio".
Bluetooth is a packet-based protocol with a master-slave structure.
One master may communicate with up to seven slaves in a piconet; all
devices share the master's clock. Packet exchange is based on the
basic clock, defined by the master, which ticks at 312.5 s intervals.
Two clock ticks make up a slot of 625 s; two slots make up a slot
pair of 1250 s. In the simple case of single-slot packets the master
transmits in even slots and receives in odd slots; the slave, conversely,
receives in even slots and transmits in odd slots. Packets may be 1, 3
or 5 slots long, but in all cases the master transmit will begin in even
slots and the slave transmit in odd slots.

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Communication and connection


A master Bluetooth device can communicate with a maximum of
seven devices in a piconet (an ad-hoc computer network using
Bluetooth technology), though not all devices reach this maximum.
The devices can switch roles, by agreement, and the slave can become
the master (for example, a headset initiating a connection to a phone
will necessarily begin as master, as initiator of the connection; but
may subsequently prefer to be slave).
The Bluetooth Core Specification provides for the connection of two
or more piconets to form a scatternet, in which certain devices
simultaneously play the master role in one piconet and the slave role
in another.
At any given time, data can be transferred between the master and one
other device (except for the little-used broadcast mode.[citation needed])
The master chooses which slave device to address; typically, it
switches rapidly from one device to another in a round-robin fashion.
Since it is the master that chooses which slave to address, whereas a
slave is (in theory) supposed to listen in each receive slot, being a
master is a lighter burden than being a slave. Being a master of seven
slaves is possible; being a slave of more than one master is
difficult.[citation needed] The specification is vague as to required behavior
in scatternets.
Many USB Bluetooth adapters or "dongles" are available, some of
which also include an IrDA adapter.[citation needed]
Uses
Class

Max. permitted power Typ. range[13]


(mW)

(dBm)

(m)

100

20

~100

2.5

~10

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~1

Bluetooth is a standard wire-replacement communications protocol


primarily designed for low-power consumption, with a short range
based on low-cost transceiver microchips in each device.[14] Because
the devices use a radio (broadcast) communications system, they do
not have to be in visual line of sight of each other, however a quasi
optical wireless path must be viable.[4] Range is power-classdependent, but effective ranges vary in practice; see the table on the
right.
Version

Data rate Max. application throughput

1.2

1 Mbit/s >80 kbit/s

2.0 + EDR 3 Mbit/s >80 kbit/s


3.0 + HS 24 Mbit/s See Version 3.0 + HS
4.0

24 Mbit/s See Version 4.0 LE

The effective range varies due to propagation conditions, material


coverage, production sample variations, antenna configurations and
battery conditions. Most Bluetooth applications are in indoor
conditions, where attenuation of walls and signal fading due to signal
reflections will cause the range to be far lower than the specified lineof-sight ranges of the Bluetooth products. Most Bluetooth
applications are battery powered Class 2 devices, with little difference
in range whether the other end of the link is a Class 1 or Class 2
device as the lower powered device tends to set the range limit. In
some cases the effective range of the data link can be extended when
a Class 2 devices is connecting to a Class 1 transceiver with both
higher sensitivity and transmission power than a typical Class 2
device.[15] Mostly however the Class 1 devices have a similar
sensitivity to Class 2 devices. Connecting two Class 1 devices with
both high sensitivity and high power can allow ranges far in excess of
the typical 100m, depending on the throughput required by the
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application. Some such devices allow open field ranges of up to 1 km


and beyond between two similar devices without exceeding legal
emission limits.[16][17][18]
While the Bluetooth Core Specification does mandate minimal for
range, the range of the technology is application-specific and not
limited. Manufacturers may tune their implementations to the range
needed for individual use cases.
Bluetooth profiles
Main article: Bluetooth profile
To use Bluetooth wireless technology, a device has to be able to
interpret certain Bluetooth profiles, which are definitions of possible
applications and specify general behaviours that Bluetooth enabled
devices use to communicate with other Bluetooth devices. These
profiles include settings to parametrize and to control the
communication from start. Adherence to profiles saves the time for
transmitting the parameters anew before the bi-directional link
becomes effective. There are a wide range of Bluetooth profiles that
describe many different types of applications or use cases for
devices.[19][20]
List of applications

A typical Bluetooth mobile phone headset.

Wireless control of and communication between a mobile phone


and a handsfree headset. This was one of the earliest
applications to become popular.
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Wireless control of and communication between a mobile phone


and a Bluetooth compatible car stereo system.
Wireless control of and communication with tablets and
speakers such as iPad and Android devices.
Wireless Bluetooth headset and Intercom. Idiomatically, a
headset is sometimes called "a Bluetooth".
Wireless networking between PCs in a confined space and
where little bandwidth is required.
Wireless communication with PC input and output devices, the
most common being the mouse, keyboard and printer.
Transfer of files, contact details, calendar appointments, and
reminders between devices with OBEX.
Replacement of previous wired RS-232 serial communications
in test equipment, GPS receivers, medical equipment, bar code
scanners, and traffic control devices.
For controls where infrared was often used.
For low bandwidth applications where higher USB bandwidth is
not required and cable-free connection desired.
Sending small advertisements from Bluetooth-enabled
advertising hoardings to other, discoverable, Bluetooth
devices.[21]
Wireless bridge between two Industrial Ethernet (e.g.,
PROFINET) networks.
Three seventh and eighth generation game consoles, Nintendo's
Wii.[22] and Sony's PlayStation 3, use Bluetooth for their
respective wireless controllers.
Dial-up internet access on personal computers or PDAs using a
data-capable mobile phone as a wireless modem.
Short range transmission of health sensor data from medical
devices to mobile phone, set-top box or dedicated telehealth
devices.[23]
Allowing a DECT phone to ring and answer calls on behalf of a
nearby mobile phone.
Real-time location systems (RTLS), are used to track and
identify the location of objects in real-time using Nodes or
tags attached to, or embedded in the objects tracked, and

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Readers that receive and process the wireless signals from


these tags to determine their locations.[24]
Personal security application on mobile phones for prevention of
theft or loss of items. The protected item has a Bluetooth marker
(e.g., a tag) that is in constant communication with the phone. If
the connection is broken (the marker is out of range of the
phone) then an alarm is raised. This can also be used as a man
overboard alarm. A product using this technology has been
available since 2009.[25]
Calgary, Alberta, Canada's Roads Traffic division uses data
collected from travelers' Bluetooth devices to predict travel
times and road congestion for motorists.[26]

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