TOPIC:

Data communication,
Telecommunications, and
networking

In partial fulfilment of the requirements

for the course
Living in the IT Era

Submitted by:

Submitted to:
MA’AM MARIA ERESA PLASTINA
TELECOMMUNICATION
Telecommunication, science, and practice of transmitting information by
electromagnetic means. Modern telecommunication centres on the problems
involved in transmitting large volumes of information over long distances
without damaging loss due to noise and interference. The basic components
of a modern digital telecommunications system must be capable of
transmitting voice, data, radio, and television signals. Digital transmission is
employed in order to achieve high reliability and because the cost of digital
switching systems is much lower than the cost of analog systems. In order to
use digital transmission, however, the analog signals that make up most
voice, radio, and television communication must be subjected to a process of
analog-to-digital conversion. (In data transmission this step is bypassed
because the signals are already in digital form; most television, radio, and
voice communication, however, use the analog system and must be
digitized.) In many cases, the digitized signal is passed through a source
encoder, which employs a number of formulas to reduce redundant binary
information. After source encoding, the digitized signal is processed in a
channel encoder, which introduces redundant information that allows errors
to be detected and corrected. The encoded signal is made suitable for
transmission by modulation onto a carrier wave and may be made part of a
larger signal in a process known as multiplexing. The multiplexed signal is
then sent into a multiple-access transmission channel. After transmission,
the above process is reversed at the receiving end, and the information is
extracted.
This article describes the components of a digital telecommunications
system as outlined above. For details on specific applications that utilize
telecommunications systems, see the articles telephone, telegraph, fax,
radio, and television. Transmission over electric wire, radio wave, and optical
fibre is discussed in telecommunications media.

Analog-to-Digital conversion
In transmission of speech, audio, or
video information, the object is high
fidelity—that is, the best possible
reproduction of the original message
without the degradations imposed by
signal distortion and noise. The basis of
relatively noise-free and distortion-free
telecommunication is the binary signal.
The simplest possible signal of any kind
that can be employed to transmit
messages, the binary signal consists of
only two possible values. These values
are represented by the binary digits, or
bits, 1 and 0. Unless the noise and
distortion picked up during transmission
are great enough to change the binary
signal from one value to another, the
correct value can be determined by the
receiver so that perfect reception can
occur.
If the information to be transmitted is
already in binary form (as in data
communication), there is no need for the
signal to be digitally encoded. But
ordinary voice communications taking place by way of a telephone are not in
binary form; neither is much of the information gathered for transmission
from a space probe, nor are the television or radio signals gathered for
transmission through a satellite link. Such signals, which continually vary
among a range of values, are said to be analog, and in digital
communications systems analog signals must be converted to digital form.
The process of making this signal conversion is called analog-to-digital (A/D)
conversion.

Sampling
Analog-to-digital conversion begins with sampling or measuring the
amplitude of the analog waveform at equally spaced discrete instants of
time. The fact that samples of a continually varying wave may be used to
represent that wave relies on the assumption that the wave is constrained in
its rate of variation. Because a communications signal is actually a complex
wave—essentially the sum of a number of component sine waves, all of
which have their own precise amplitudes and phases—the rate of variation of
the complex wave can be measured by the frequencies of oscillation of all its
components. The difference between the maximum rate of oscillation (or
highest frequency) and the minimum rate of oscillation (or lowest frequency)
of the sine waves making up the signal is known as the bandwidth (B) of the
signal. Bandwidth thus represents the maximum frequency range occupied
by a signal. In the case of a voice signal having a minimum frequency of 300
hertz and a maximum frequency of 3,300 hertz, the bandwidth is 3,000
hertz, or 3 kilohertz. Audio signals generally occupy about 20 kilohertz of
bandwidth, and standard video signals occupy approximately 6 million hertz,
or 6 megahertz.
The concept of bandwidth is central to all telecommunications. In analog-to-
digital conversion, there is a fundamental theorem that the analog signal
may be uniquely represented by discrete samples spaced no more than one
over twice the bandwidth (1/2B) apart. This theorem is commonly referred to
as the sampling theorem, and the sampling interval (1/2B seconds) is
referred to as the Nyquist interval (after the Swedish-born American
electrical engineer Harry Nyquist). As an example of the Nyquist interval, in
past telephone practice the bandwidth, commonly fixed at 3,000 hertz, was
sampled at least every 1/6,000 second. In current practice 8,000 samples are
taken per second, in order to increase the frequency range and the fidelity of
the speech representation.

Quantization
In order for a sampled signal to be stored or transmitted in digital form, each
sampled amplitude must be converted to one of a finite number of possible
values, or levels. For ease in conversion to binary form, the number of levels
is usually a power of 2—that is, 8, 16, 32, 64, 128, 256, and so on, depending
on the degree of precision required. In digital transmission of voice, 256
levels are commonly used because tests have shown that this provides
adequate fidelity for the average telephone listener.
The input to the quantizer is a sequence of sampled amplitudes for which
there are an infinite number of possible values. The output of the quantizer,
on the other hand, must be restricted to a finite number of levels. Assigning
infinitely variable amplitudes to a limited number of levels inevitably
introduces inaccuracy, and inaccuracy results in a corresponding amount of
signal distortion. (For this reason quantization is often called a “lossy”
system.) The degree of inaccuracy depends on the number of output levels
used by the quantizer. More quantization levels increase the accuracy of the
representation, but they also increase the storage capacity or transmission
speed required. Better performance with the same number of output levels
can be achieved by judicious placement of the output levels and the
amplitude thresholds needed for assigning those levels. This placement in
turn depends on the nature of the waveform that is being quantized.
Generally, an optimal quantizer places more levels in amplitude ranges
where the signal is more likely to occur and fewer levels where the signal is
less likely. This technique is known as nonlinear quantization. Nonlinear
quantization can also be accomplished by passing the signal through a
compressor circuit, which amplifies the signal’s weak components and
attenuates its strong components. The compressed signal, now occupying a
narrower dynamic range, can be quantized with a uniform, or linear, spacing
of thresholds and output levels. In the case of the telephone signal, the
compressed signal is uniformly quantized at 256 levels, each level being
represented by a sequence of eight bits. At the receiving end, the
reconstituted signal is expanded to its original range of amplitudes. This
sequence of compression and expansion, known as companding, can yield an
effective dynamic range equivalent to 13 bits.

Bit mapping
In the next step in the digitization process, the output of the quantizer is
mapped into a binary sequence. An encoding table that might be used to
generate the binary sequence is shown below:

Encoding table.
It is apparent that 8 levels require three binary digits, or bits; 16 levels
require four bits; and 256 levels require eight bits. In general 2n levels
require n bits.
In the case of 256-level voice quantization, where each level is represented
by a sequence of 8 bits, the overall rate of transmission is 8,000 samples per
second times 8 bits per sample, or 64,000 bits per second. All 8 bits must be
transmitted before the next sample appears. In order to use more levels,
more binary samples would have to be squeezed into the allotted time slot
between successive signal samples. The circuitry would become more costly,
and the bandwidth of the system would become correspondingly greater.
Some transmission channels (telephone wires are one example) may not
have the bandwidth capability required for the increased number of binary
samples and would distort the digital signals. Thus, although the accuracy
required determines the number of quantization levels used, the resultant
binary sequence must still be transmitted within the bandwidth tolerance
allowed.

Source encoding
As is pointed out in analog-to-digital conversion, any available
telecommunications medium has a limited capacity for data transmission.
This capacity is commonly measured by the parameter called bandwidth.
Since the bandwidth of a signal increases with the number of bits to be
transmitted each second, an important function of a digital communications
system is to represent the digitized signal by as few bits as possible—that is,
to reduce redundancy. Redundancy reduction is accomplished by a source
encoder, which often operates in conjunction with the analog-to-digital
converter.

Huffman codes
In general, fewer bits on the average will be needed if the source encoder
considers the probabilities at which different quantization levels are likely to
occur. A simple example will illustrate this concept. Assume a quantizing
scale of only four levels: 1, 2, 3, and 4. Following the usual standard of binary
encoding, each of the four levels would be mapped by a two-bit code word.
But also assume that level 1 occurs 50 percent of the time, that level 2
occurs 25 percent of the time, and that levels 3 and 4 each occur 12.5
percent of the time. Using variable-bit code words might cause more efficient
mapping of these levels to be achieved. The variable-bit encoding rule would
use only one bit 50 percent of the time, two bits 25 percent of the time, and
three bits 25 percent of the time. On average it would use 1.75 bits per
sample rather than the two bits per sample used in the standard code.
Thus, for any given set of levels and associated probabilities, there is an
optimal encoding rule that minimizes the number of bits needed to represent
the source. This encoding rule is known as the Huffman code, after the
American D.A. Huffman, who created it in 1952. Even more efficient encoding
is possible by grouping sequences of levels together and applying the
Huffman code to these sequences.

The Lempel-Ziv algorithm

The design and performance of the Huffman code depends on the designers’
knowing the probabilities of different levels and sequences of levels. In many
cases, however, it is desirable to have an encoding system that can adapt to
the unknown probabilities of a source. A very efficient technique for encoding
sources without needing to know their probable occurrence was developed in
the 1970s by the Israelis Abraham Lempel and Jacob Ziv. The Lempel-Ziv
algorithm works by constructing a codebook out of sequences encountered
previously. For example, the codebook might begin with a set of four 12-bit
code words representing four possible signal levels. If two of those levels
arrived in sequence, the encoder, rather than transmitting two full code
words (of length 24), would transmit the code word for the first level (12 bits)
and then an extra two bits to indicate the second level. The encoder would
then construct a new code word of twelve bits for the sequence of two levels,
so that even fewer bits would be used thereafter to represent that particular
combination of levels. The encoder would continue to read quantization
levels until another sequence arrived for which there was no code word. In
this case the sequence without the last level would be in the codebook, but
not the whole sequence of levels. Again, the encoder would transmit the
code word for the initial sequence of levels and then an extra two bits for the
last level. The process would continue until all 4,096 possible 12-bit
combinations had been assigned as code words.
In practice, standard algorithms for compressing binary files use code words
of twelve bits and transmit 1 extra bit to indicate a new sequence. Using
such a code, the Lempel-Ziv algorithm can compress transmissions of English
text by about 55 percent, whereas the Huffman code compresses the
transmission by only 43 percent.

Run length codes

Certain signal sources are known to produce “runs,” or long sequences of
only 1s or 0s. In these cases it is more efficient to transmit a code for the
length of the run rather than all the bits that represent the run itself. One
source of long runs is the fax machine. A fax machine works by scanning a
document and mapping very small areas of the document into either a black
pixel (picture element) or a white pixel. The document is divided into a
number of lines (approximately 100 per inch), with 1,728 pixels in each line
(at standard resolution). If all black pixels were mapped into 1s and all white
pixels into 0s, then the scanned document would be represented by
1,857,600 bits (for a standard American 11-inch page). At older modem
transmission speeds of 4,800 bits per second, it would take 6 minutes 27
seconds to send a single page. If, however, the sequence of 0s and 1s were
compressed using a run-length code, significant reductions in transmission
time would be made.
The code for fax machines is actually a combination of a run-length code and
a Huffman code; it can be explained as follows: A run-length code maps run
lengths into code words, and the codebook is partitioned into two parts. The
first part contains symbols for runs of lengths that are a multiple of 64; the
second part is made up of runs from 0 to 63 pixels. Any run length would
then be represented as a multiple of 64 plus some remainder. For example, a
run of 205 pixels would be sent using the code word for a run of length 192
(3 × 64) plus the code word for a run of length 13. In this way the number of
bits needed to represent the run is decreased significantly. In addition,
certain runs that are known to have a higher probability of occurrence are
encoded into code words of short length, further reducing the number of bits
that need to be transmitted. Using this type of encoding, typical
compressions for facsimile transmission range between 4 to 1 and 8 to 1.
Coupled to higher modem speeds, these compressions reduce the
transmission time of a single page to between 48 seconds and 1 minute 37
seconds.

Channel encoding
As described in Source encoding, one purpose of the source encoder is to
eliminate redundant binary digits from the digitized signal. The strategy of
the channel encoder, on the other hand, is to add redundancy to the
transmitted signal—in this case so that errors caused by noise during
transmission can be corrected at the receiver. The process of encoding for
protection against channel errors is called error-control coding. Error-control
codes are used in a variety of applications, including satellite
communication, deep-space communication, mobile radio communication,
and computer networking.
There are two commonly employed methods for protecting electronically
transmitted information from errors. One method is called forward error
control (FEC). In this method information bits are protected against errors by
the transmitting of extra redundant bits, so that if errors occur during
transmission the redundant bits can be used by the decoder to determine
where the errors have occurred and how to correct them. The second method
of error control is called automatic repeat request (ARQ). In this method
redundant bits are added to the transmitted information and are used by the
receiver to detect errors. The receiver then signals a request for a repeat
transmission. Generally, the number of extra bits needed simply to detect an
error, as in the ARQ system, is much smaller than the number of redundant
bits needed both to detect and to correct an error, as in the FEC system.

Repetition codes
One simple, but not usually implemented, FEC method is to send each data
bit three times. The receiver examines the three transmissions and decides
by majority vote whether a 0 or 1 represents a sample of the original signal.
In this coded system, called a repetition code of block-length three and rate
one-third, three times as many bits per second are used to transmit the
same signal as are used by an uncoded system; hence, for a fixed available
bandwidth only one-third as many signals can be conveyed with the coded
system as compared with the uncoded system. The gain is that now at least
two of the three coded bits must be in error before a reception error occurs.

The Hamming code

Another simple example of an FEC code is known as the Hamming code. This
code is able to protect a four-bit information signal from a single error on the
channel by adding three redundant bits to the signal. Each sequence of
seven bits (four information bits plus three redundant bits) is called a code
word. The first redundant bit is chosen so that the sum of ones in the first
three information bits plus the first redundant bit amounts to an even
number. (This calculation is called a parity check, and the redundant bit is
called a parity bit.) The second parity bit is chosen so that the sum of the
ones in the last three information bits plus the second parity bit is even, and
the third parity bit is chosen so that the sum of ones in the first, second, and
fourth information bits and the last parity bit is even. This code can correct a
single channel error by recomputing the parity checks. A parity check that
fails indicates an error in one of the positions checked, and the two
subsequent parity checks, by process of elimination, determine the precise
location of the error. The Hamming code thus can correct any single error
that occurs in any of the seven positions. If a double error occurs, however,
the decoder will choose the wrong code word.

Convolutional encoding
The Hamming code is called a block code because information is blocked into
bit sequences of finite length to which a number of redundant bits are added.
When k information bits are provided to a block encoder, n − k redundancy
bits are appended to the information bits to form a transmitted code word of
n bits. The entire code word of length n is thus completely determined by
one block of k information bits. In another channel-encoding scheme, known
as convolutional encoding, the encoder output is not naturally segmented
into blocks but is instead an unending stream of bits. In convolutional
encoding, memory is incorporated into the encoding process, so that the
preceding M blocks of k information bits, together with the current block of k
information bits, determine the encoder output. The encoder accomplishes
this by shifting among a finite number of “states,” or “nodes.” There are
several variations of convolutional encoding, but the simplest example may
be seen in what is known as the (n,1) encoder, in which the current block of k
information bits consists of only one bit. At each given state of the (n,1)
encoder, when the information bit (a 0 or a 1) is received, the encoder
transmits a sequence of n bits assigned to represent that bit when the
encoder is at that current state. At the same time, the encoder shifts to one
of only two possible successor states, depending on whether the information
bit was a 0 or a 1. At this successor state, in turn, the next information bit is
represented by a specific sequence of n bits, and the encoder is again shifted
to one of two possible successor states. In this way, the sequence of
information bits stored in the encoder’s memory determines both the state of
the encoder and its output, which is modulated and transmitted across the
channel. At the receiver, the demodulated bit sequence is compared to the
possible bit sequences that can be produced by the encoder. The receiver
determines the bit sequence that is most likely to have been transmitted,
often by using an efficient decoding algorithm called Viterbi decoding (after
its inventor, A.J. Viterbi). In general, the greater the memory (i.e., the more
states) used by the encoder, the better the error-correcting performance of
the code—but only at the cost of a more complex decoding algorithm. In
addition, the larger the number of bits (n) used to transmit information, the
better the performance—at the cost of a decreased data rate or larger
bandwidth.
Coding and decoding processes similar to those described above are
employed in trellis coding, a coding scheme used in high-speed modems.
However, instead of the sequence of bits that is produced by a convolutional
encoder, a trellis encoder produces a sequence of modulation symbols. At
the transmitter, the channel-encoding process is coupled with the
modulation process, producing a system known as trellis-coded modulation.
At the receiver, decoding and demodulating are performed jointly in order to
optimize the performance of the error-correcting algorithm.

Modulation
In many telecommunications systems, it is necessary to represent an
information-bearing signal with a waveform that can pass accurately through
a transmission medium. This assigning of a suitable waveform is
accomplished by modulation, which is the process by which some
characteristic of a carrier wave is varied in accordance with an information
signal, or modulating wave. The modulated signal is then transmitted over a
channel, after which the original information-bearing signal is recovered
through a process of demodulation.
Modulation is applied to information signals for a number of reasons, some of
which are outlined below.
1. Many transmission channels are characterized by limited passbands—
that is, they will pass only certain ranges of frequencies without
seriously attenuating them (reducing their amplitude). Modulation
methods must therefore be applied to the information signals in order
to “frequency translate” the signals into the range of frequencies that
are permitted by the channel. Examples of channels that exhibit
passband characteristics include alternating-current-coupled coaxial
cables, which pass signals only in the range of 60 kilohertz to several
hundred megahertz, and fibre-optic cables, which pass light signals
only within a given wavelength range without significant attenuation.
In these instances frequency translation is used to “fit” the information
signal to the communications channel.
2. In many instances a communications channel is shared by multiple
users. In order to prevent mutual interference, each user’s information
signal is modulated onto an assigned carrier of a specific frequency.
When the frequency assignment and subsequent combining is done at
a central point, the resulting combination is a frequency-division
multiplexed signal, as is discussed in Multiplexing. Frequently there is
no central combining point, and the communications channel itself acts
as a distributed combine. An example of the latter situation is the
broadcast radio bands (from 540 kilohertz to 600 megahertz), which
permit simultaneous transmission of multiple AM radio, FM radio, and
television signals without mutual interference as long as each signal is
assigned to a different frequency band.
3. Even when the communications channel can support direct
transmission of the information-bearing signal, there are often practical
reasons why this is undesirable. A simple example is the transmission
of a three kilohertz (i.e., voiceband) signal via radio wave. In free space
the wavelength of a three-kilohertz signal is 100 kilometres (60 miles).
Since an effective radio antenna is typically as large as half the
wavelength of the signal, a three-kilohertz radio wave might require an
antenna up to 50 kilometres in length. In this case translation of the
 voice frequency to a higher frequency would allow the use of a much
smaller antenna.

Analog modulation
As is noted in analog-to-digital conversion, voice signals, as well as audio and
video signals, are inherently analog in form. In most modern systems these
signals are digitized prior to transmission, but in some systems the analog
signals are still transmitted directly without converting them to digital form.
There are two commonly used methods of modulating analog signals. One
technique, called amplitude modulation, varies the amplitude of a fixed-
frequency carrier wave in proportion to the information signal. The other
technique, called frequency modulation, varies the frequency of a fixed-
amplitude carrier wave in proportion to the information signal.

Digital modulation
In order to transmit computer data and
other digitized information over a
communications channel, an analog
carrier wave can be modulated to
reflect the binary nature of the digital
baseband signal. The parameters of
the carrier that can be modified are the
amplitude, the frequency, and the
phase.

Amplitude-shift keying
If amplitude is the only parameter of the carrier wave to be altered by the
information signal, the modulating method is called amplitude-shift keying
(ASK). ASK can be considered a digital version of analog amplitude
modulation. In its simplest form, a burst of radio frequency is transmitted
only when a binary 1 appears and is stopped when a 0 appears. In another
variation, the 0 and 1 are represented in the modulated signal by a shift
between two preselected amplitudes.

Frequency-shift keying
If frequency is the parameter chosen to be a function of the information
signal, the modulation method is called frequency-shift keying (FSK). In the
simplest form of FSK signaling, digital data is transmitted using one of two
frequencies, whereby one frequency is used to transmit a 1 and the other
frequency to transmit a 0. Such a scheme was used in the Bell 103
voiceband modem, introduced in 1962, to transmit information at rates up to
300 bits per second over the public switched telephone network. In the Bell
103 modem, frequencies of 1,080 +/- 100 hertz and 1,750 +/- 100 hertz
were used to send binary data in both directions.

Phase-shift keying
When phase is the parameter altered by the information signal, the method
is called phase-shift keying (PSK). In the simplest form of PSK a single radio
frequency carrier is sent with a fixed phase to represent a 0 and with a 180°
phase shift—that is, with the opposite polarity—to represent a 1. PSK was
employed in the Bell 212 modem, which was introduced about 1980 to
transmit information at rates up to 1,200 bits per second over the public
switched telephone network.
Advanced methods
In addition to the elementary forms of digital modulation described above,
there exist more advanced methods that result from a superposition of
multiple modulating signals. An example of the latter form of modulation is
quadrature amplitude modulation (QAM). QAM signals actually transmit two
amplitude-modulated signals in phase quadrature (i.e., 90° apart), so that
four or more bits are represented by each shift of the combined signal.
Communications systems that employ QAM include digital cellular systems in
the United States and Japan as well as most voiceband modems transmitting
above 2,400 bits per second.
A form of modulation that combines convolutional codes with QAM is known
as trellis-coded modulation (TCM), which is described in Channel encoding.
Trellis-coded modulation forms an essential part of most of the modern
voiceband modems operating at data rates of 9,600 bits per second and
above, including V.32 and V.34 modems.

Multiplexing
Because of the installation cost of a communications channel, such as a
microwave link or a coaxial cable link, it is desirable to share the channel
among multiple users. Provided that the channel’s data capacity exceeds
that required to support a single user, the channel may be shared through
the use of multiplexing methods. Multiplexing is the sharing of a
communications channel through local combining of signals at a common
point. Two types of multiplexing are commonly employed: frequency-division
multiplexing and time-division multiplexing.

Frequency-division multiplexing
In frequency-division multiplexing
(FDM), the available bandwidth of a
communications channel is shared
among multiple users by frequency
translating, or modulating, each of the
individual users onto a different carrier
frequency. Assuming sufficient
frequency separation of the carrier
frequencies that the modulated signals
do not overlap, recovery of each of the
FDM signals is possible at the receiving end. In order to prevent overlap of
the signals and to simplify filtering, each of the modulated signals is
separated by a guard band, which consists of an unused portion of the
available frequency spectrum. Each user is assigned a given frequency band
for all time.
While each user’s information signal may be either analog or digital, the
combined FDM signal is inherently an analog waveform. Therefore, an FDM
signal must be transmitted over an analog channel. Examples of FDM are
found in some of the old long-distance telephone transmission systems,
including the American N- and L-carrier coaxial cable systems and analog
point-to-point microwave systems. In the L-carrier system a hierarchical
combining structure is employed in which 12 voiceband signals are
frequency-division multiplexed to form a group signal in the frequency range
of 60 to 108 kilohertz. Five group signals are multiplexed to form a
supergroup signal in the frequency range of 312 to 552 kilohertz,
corresponding to 60 voiceband signals, and 10 supergroup signals are
multiplexed to form a master group signal. In the L1 carrier system, deployed
in the 1940s, the master group was transmitted directly over coaxial cable.
For microwave systems, it was frequency modulated onto a microwave
carrier frequency for point-to-point transmission. In the L4 system,
developed in the 1960s, six master groups were combined to form a jumbo
group signal of 3,600 voiceband signals.

Time-division multiplexing
Digital multiplexing, as employed in
the North American telephone
systemIn time-division multiplexing
(TDM), 24 digitized voice signals, each
at 64 kilobits per second, are assigned
successive time slots in a 1.544-
megabits-per-second signal.
Combined signals are further
combined to form data streams of
increasing bitrate and voice-carrying
capacity.
Multiplexing also may be conducted through the interleaving of time
segments from different signals onto a single transmission path—a process
known as time-division multiplexing (TDM). Time-division multiplexing of
multiple signals is possible only when the available data rate of the channel
exceeds the data rate of the total number of users. While TDM may be
applied to either digital or analog signals, in practice it is applied almost
always to digital signals. The resulting composite signal is thus also a digital
signal.
In a representative TDM system, data from multiple users are presented to a
time-division multiplexer. A scanning switch then selects data from each of
the users in sequence to form a composite TDM signal consisting of the
interleaved data signals. Each user’s data path is assumed to be time-
aligned or synchronized to each of the other users’ data paths and to the
scanning mechanism. If only one bit were selected from each of the data
sources, then the scanning mechanism would select the value of the arriving
bit from each of the multiple data sources. In practice, however, the
scanning mechanism usually selects a slot of data consisting of multiple bits
of each user’s data; the scanner switch is then advanced to the next user to
select another slot, and so on. Each user is assigned a given time slot for all
time.
Most modern telecommunications systems employ some form of TDM for
transmission over long-distance routes. The multiplexed signal may be sent
directly over cable systems, or it may be modulated onto a carrier signal for
transmission via radio wave. Examples of such systems include the North
American T carriers as well as digital point-to-point microwave systems. In T1
systems, introduced in 1962, 24 voiceband signals (or the digital equivalent)
are time-division multiplexed together. The voiceband signal is a 64-kilobit-
per-second data stream consisting of 8-bit symbols transmitted at a rate of
8,000 symbols per second. The TDM process interleaves 24 8-bit time slots
together, along with a single frame-synchronization bit, to form a 193-bit
frame. The 193-bit frames are formed at the rate of 8,000 frames per
second, resulting in an overall data rate of 1.544 megabits per second. For
transmission over more recent T-carrier systems, T1 signals are often further
multiplexed to form higher-data-rate signals—again using a hierarchical
scheme.

Multiple access
Multiplexing is defined as the sharing of a communications channel through
local combining at a common point. In many cases, however, the
communications channel must be efficiently shared among many users that
are geographically distributed and that sporadically attempt to communicate
at random points in time. Three schemes have been devised for efficient
sharing of a single channel under these conditions; they are called
frequency-division multiple access (FDMA), time-division multiple access
(TDMA), and code-division multiple access (CDMA). These techniques can be
used alone or together in telephone systems, and they are well illustrated by
the most advanced mobile cellular systems.

Frequency-division multiple access

In FDMA the goal is to divide the frequency spectrum into slots and then to
separate the signals of different users by placing them in separate frequency
slots. The difficulty is that the frequency spectrum is limited and that there
are typically many more potential communicators than there are available
frequency slots. In order to make efficient use of the communications
channel, a system must be devised for managing the available slots. In the
advanced mobile phone system (AMPS), the cellular system employed in the
United States, different callers use separate frequency slots via FDMA. When
one telephone call is completed, a network-managing computer at the
cellular base station reassigns the released frequency slot to a new caller. A
key goal of the AMPS system is to reuse frequency slots whenever possible in
order to accommodate as many callers as possible. Locally within a cell,
frequency slots can be reused when corresponding calls are terminated. In
addition, frequency slots can be used simultaneously by multiple callers
located in separate cells. The cells must be far enough apart geographically
that the radio signals from one cell are sufficiently attenuated at the location
of the other cell using the same frequency slot.

Time-division multiple access

In TDMA the goal is to divide time into slots and separate the signals of
different users by placing the signals in separate time slots. The difficulty is
that requests to use a single communications channel occur randomly, so
that on occasion the number of requests for time slots is greater than the
number of available slots. In this case information must be buffered, or
stored in memory, until time slots become available for transmitting the
data. The buffering introduces delay into the system. In the IS54 cellular
system, three digital signals are interleaved using TDMA and then
transmitted in a 30-kilohertz frequency slot that would be occupied by one
analog signal in AMPS. Buffering digital signals and interleaving them in time
causes some extra delay, but the delay is so brief that it is not ordinarily
noticed during a call. The IS54 system uses aspects of both TDMA and FDMA.

Code-division multiple access

In CDMA, signals are sent at the same time in the same frequency band.
Signals are either selected or rejected at the receiver by recognition of a
user-specific signature waveform, which is constructed from an assigned
spreading code. The IS95 cellular system employs the CDMA technique. In
IS95 an analog speech signal that is to be sent to a cell site is first quantized
and then organized into one of a number of digital frame structures. In one
frame structure, a frame of 20 milliseconds’ duration consists of 192 bits. Of
these 192 bits, 172 represent the speech signal itself, 12 form a cyclic
redundancy check that can be used for error detection, and 8 form an
encoder “tail” that allows the decoder to work properly. These bits are
formed into an encoded data stream. After interleaving of the encoded data
stream, bits are organized into groups of six. Each group of six bits indicates
which of 64 possible waveforms to transmit. Each of the waveforms to be
transmitted has a particular pattern of alternating polarities and occupies a
certain portion of the radio-frequency spectrum. Before one of the
waveforms is transmitted, however, it is multiplied by a code sequence of
polarities that alternate at a rate of 1.2288 megahertz, spreading the
bandwidth occupied by the signal and causing it to occupy (after filtering at
the transmitter) about 1.23 megahertz of the radio-frequency spectrum. At
the cell site one user can be selected from multiple users of the same 1.23-
megahertz bandwidth by its assigned code sequence.
CDMA is sometimes referred to as spread-spectrum multiple access (SSMA),
because the process of multiplying the signal by the code sequence causes
the power of the transmitted signal to be spread over a larger bandwidth.
Frequency management, a necessary feature of FDMA, is eliminated in
CDMA. When another user wishes to use the communications channel, it is
assigned a code and immediately transmits instead of being stored until a
frequency slot opens.

Telecommunications Network
telecommunications network, electronic system of links and switches, and
the controls that govern their operation, which allows for data transfer and
exchange among multiple users.
When several users of telecommunications media wish to communicate with
one another, they must be organized into some form of network. In theory,
each user can be given a direct point-to-point link to all the other users in
what is known as a fully connected topology (similar to the connections
employed in the earliest days of telephony), but in practice this technique is
impractical and expensive—especially for a large and dispersed network.
Furthermore, the method is inefficient since most of the links will be idle at
any given time. Modern telecommunications networks avoid these issues by
establishing a linked network of switches, or nodes, such that each user is
connected to one of the nodes. Each link in such a network is called a
communications channel. Wire, fibre-optic cable, and radio waves may be
used for different communications channels.

Types of networks
Switched communications network
A switched communications network transfers data from source to
destination through a series of network nodes. Switching can be done in one
of two ways. In a circuit-switched network, a dedicated physical path is
established through the network and is held for as long as communication is
necessary. An example of this type of network is the traditional (analog)
telephone system. A packet-switched network, on the other hand, routes
digital data in small pieces called packets, each of which proceeds
independently through the network. In a process called store-and-forward,
each packet is temporarily stored at each intermediate node, then forwarded
when the next link becomes available. In a connection-oriented transmission
scheme, each packet takes the same route through the network, and thus all
packets usually arrive at the destination in the order in which they were sent.
Conversely, each packet may take a different path through the network in a
connectionless or datagram scheme. Since datagrams may not arrive at the
destination in the order in which they were sent, they are numbered so that
they can be properly reassembled. The latter is the method that is used for
transmitting data through the Internet.
Broadcast network
A broadcast network avoids the
complex routing procedures of a
switched network by ensuring that
each node’s transmissions are
received by all other nodes in the
network. Therefore, a broadcast
network has only a single communications channel. A wired local area
network (LAN), for example, may be set up as a broadcast network, with one
user connected to each node and the nodes typically arranged in a bus, ring,
or star topology, as shown in the figure. Nodes connected together in a
wireless LAN may broadcast via radio or optical links. On a larger scale,
many satellite radio systems are broadcast networks, since each Earth
station within the system can typically hear all messages relayed by a
satellite.
Network access
Since all nodes can hear each transmission in a broadcast network, a
procedure must be established for allocating a communications channel to
the node or nodes that have packets to transmit and at the same time
preventing destructive interference from collisions (simultaneous
transmissions). This type of communication, called multiple access, can be
established either by scheduling (a technique in which nodes take turns
transmitting in an orderly fashion) or by random access to the channel.
Scheduled access
In a scheduling method known as time-division multiple access (TDMA), a
time slot is assigned in turn to each node, which uses the slot if it has
something to transmit. If some nodes are much busier than others, then
TDMA can be inefficient, since no data are passed during time slots allocated
to silent nodes. In this case a reservation system may be implemented, in
which there are fewer time slots than nodes and a node reserves a slot only
when it is needed for transmission.
A variation of TDMA is the process of polling, in which a central controller
asks each node in turn if it requires channel access, and a node transmits a
packet or message only in response to its poll. “Smart” controllers can
respond dynamically to nodes that suddenly become very busy by polling
them more often for transmissions. A decentralized form of polling is called
token passing. In this system a special “token” packet is passed from node to
node. Only the node with the token is authorized to transmit; all others are
listeners.
Random access
Scheduled access schemes have several disadvantages, including the large
overhead required for the reservation, polling, and token passing processes
and the possibility of long idle periods when only a few nodes are
transmitting. This can lead to extensive delays in routing information,
especially when heavy traffic occurs in different parts of the network at
different times—a characteristic of many practical communications networks.
Random-access algorithms were designed specifically to give nodes with
something to transmit quicker access to the channel. Although the channel is
vulnerable to packet collisions under random access, various procedures
have been developed to reduce this probability.
Carrier sense multiple access
One random-access method that reduces the chance of collisions is called
carrier sense multiple access (CSMA). In this method a node listens to the
channel first and delays transmitting when it senses that the channel is busy.
Because of delays in channel propagation and node processing, it is possible
that a node will erroneously sense a busy channel to be idle and will cause a
collision if it transmits. In CSMA, however, the transmitting nodes will
recognize that a collision has occurred: the respective destinations will not
acknowledge receipt of a valid packet. Each node then waits a random time
before sending again (hopefully preventing a second collision). This method
is commonly employed in packet networks with radio links, such as the
system used by amateur radio operators.
It is important to minimize the time that a communications channel spends
in a collision state, since this effectively shuts down the channel. If a node
can simultaneously transmit and receive (usually possible on wire and fibre-
optic links but not on radio links), then it can stop sending immediately upon
detecting the beginning of a collision, thus moving the channel out of the
collision state as soon as possible. This process is called carrier sense
multiple access with collision detection (CSMA/CD), a feature of the popular
wired Ethernet. (For more information on Ethernet, see computer: Local area
networks.)
Spread-spectrum multiple access
Since collisions are so detrimental to network performance, methods have
been developed to allow multiple transmissions on a broadcast network
without necessarily causing mutual packet destruction. One of the most
successful is called spread-spectrum multiple access (SSMA). In SSMA
simultaneous transmissions will cause only a slight increase in bit error
probability for each user if the channel is not too heavily loaded. Error-free
packets can be obtained by using an appropriate control code.
Disadvantages of SSMA include wider signal bandwidth and greater
equipment cost and complexity compared with conventional CSMA.
Open systems interconnection
Different communication
requirements necessitate different
network solutions, and these different
network protocols can create
significant problems of compatibility
when networks are interconnected
with one another. In order to
overcome some of these
interconnection problems, the open
systems interconnection (OSI) was
approved in 1983 as an international
standard for communications
architecture by the International Organization for Standardization (ISO) and
the International Telegraph and Telephone Consultative Committee (CCITT).
The OSI model, as shown in the figure, consists of seven layers, each of
which is selected to perform a well-defined function at a different level of
abstraction. The bottom three layers provide for the timely and correct
transfer of data, and the top four ensure that arriving data are recognizable
and useful. While all seven layers are usually necessary at each user
location, only the bottom three are normally employed at a network node
since nodes are concerned only with timely and correct data transfer from
point to point.
Data recognition and use
The application layer is difficult to generalize since its content is specific to
each user. For example, distributed databases used in the banking and
airline industries require several access and security issues to be solved at
this level. Network transparency (making the physical distribution of
resources irrelevant to the human user) also is managed at this level. The
presentation layer, on the other hand, performs functions that are requested
sufficiently often that a general solution is warranted. These functions are
often placed in a software library that is accessible by several users running
different applications. Examples are text conversion, data compression, and
data encryption.
User interface with the network is performed by the session layer, which
manages the process of connecting to another computer, verifying user
authenticity, and establishing a reliable communication process. This layer
also ensures that files which can be altered by several network users are
kept in order. Data from the session layer are accepted by the transport
layer, which separates the data stream into smaller units, if necessary, and
ensures that all arrive correctly at the destination. If fast throughput is
needed, the transport layer may establish several simultaneous paths in the
network and send different parts of the data over each path. Conversely, if
low cost is a requirement, then the layer may time-multiplex several users’
data over one path through the network. Flow control is also regulated at this
level, ensuring that data from a fast source will not overrun a slow
destination.
Data transfer
The network layer breaks data into packets and determines how the packets
are routed within the network, which nodes (if any) will check packets for
errors along the route, and whether congestion control is needed in a heavily
loaded network. The data-link layer transforms a raw communications
channel into a line that appears essentially free of transmission errors to the
network layer. This is done by breaking data up into data frames,
transmitting them sequentially, and processing acknowledgment frames sent
back to the source by the destination. This layer also establishes frame
boundaries and implements recovery procedures from lost, damaged, or
duplicated frames. The physical layer is the transmission medium itself,
along with various electric and mechanical specifications.

INTERNET
The Internet is a virtual networking medium that can be connected and used
on various devices. It enables the users to send, receive, collect, store,
update, delete, and more. Internet usage is expanding its boundaries every
day. A few of the Internet’s major uses are e-commerce, e-learning,
knowledge sharing, social connectivity, etc. So, in this topic, we are going to
learn about what are the uses of internet.
This is the primary uses of the internet which are as follows:
 Seek information on anything across the globe on a real-time basis.
 Communicate, and collaborate with others.
 Telecommute to the office or work from home.
 Do transactions with business entities.
 Download files from a remote.
 Get educated and entertained.
 Carry out social.
  Do group activities.
 Collect operational data from remote equipment (stationary as well as
moving).
 Process data while it is streamed to the central server.
 Get real-time data on the surrounding devices, systems, and whether
to automate activities.
 Design a decision-taking system against the decision support system.
 Connect people, stakeholders, machines, and everything.
Top 9 Uses of Internet

1. Electronic Mail (email)

The first major use of the internet is Email. People thronged to email for
sharing information, data files, Photos, Videos, Business communications,
and more. This has enabled faster communication between people and
improved business efficiency. An email has reduced paper usage and the
load on physical mail systems.
Though other latest collaboration tools provide many rich features, they
cannot de-popularize Email, which still rules official and personal
communication. There are many free Email websites offering mail services,
and practically every individual has an Email address connected by email.
Email concepts paved the way for developing many innovative tools for
improved collaboration.
2. FTP File Transfer
This is the second major use case for the internet in the early days. FTP is
the file transfer protocol that enables data exchange between two
stakeholders over internet media in a secure way. The data exchange may
occur between two business entities or customers with business and vice
versa. Usually, E-mail restricts the size of a file that can be shared, and it is
not secure to share sensitive and confidential data across public networks.
FTP concept is still in use in mobile apps for file downloading.
3. Search Engines
These engines locate the information one seeks, available on whichever
server across the globe (world wide web). Google, Yahoo, and MSN are
renowned search engines today. One can search for anything on this site,
and the search question can be in any format. People have started using the
word Google as a generic verb synonymous with search.
4. E-Commerce
The Internet enables the selling of goods and services in online mode. Many
e-commerce platform vendors like Amazon and Ola aggregate several
products/services available in the market and sell them through their portal
to customers. Products are procured by platform vendors, stored in their
warehouses, packed, and distributed by them in their brand. Customers get a
good discount and don’t have to visit physical stores.
5. Online Banking
Called Net banking, it allows doing banking transactions at ease sitting at
home or while on mobile. Footfalls in the bank branches have come down
appreciably with almost all the services available in net banking 24×7. Any
amount of money can be transferred instantaneously through this facility. E-
Banking supports Electricity bills, Telephone bills, and other services
payments.
6. Cashless Transactions
Bill Payment at merchandise outlets through debit cards, credit cards, and
UPI gateway is increasing. Cash circulation gets reduced in the system to the
extent of the growth of these transactions. It’s growing by more than 50%
yearly and is expected to grow by 10 times over the next 5 years.
7. Education
The Internet offers a wealth of educational material on any subject with
structured navigation and search facilities. One can seek any reading
material, and the internet will get it for them from any server in any part of
the world, and people need not have to go to libraries to go through books.
Those who cannot attend physical (face-to-face) classes can take an online
course where they get connected to the teacher, in the other part of the
world, in video mode and get taught on the subject backed up by other
audio-visual tools.
8. Collaboration
Online chat tools like messenger, Skype, and other video conferencing tools
help people to get connected 24 x7 and have hassle-free business and
personal discussions. This avoids unwanted travel by people and saves their
time for productive use. The Internet has also facilitated work from home
with seamless connectivity to the office and avoiding daily commuting.
9. Social Networking
Internet connects people online and enables them to form social groups.
Information, Ideas, views, and opinions on social/political issues are
exchanged. Political and social organization uses this platform to promote
their interest among the public.

Applications of Internet
With the evolution of IOT, Artificial Intelligence technologies, supported by
computing resources in the cloud, more and more new applications are being
developed over the internet layer, and a few of them are:
1. Tracking the Vehicle – Fleet Management system
2. Monitoring the health of the moving vehicle – Telematics
3. Autonomous and Driverless vehicle – 5G networks
4. Remote diagnostics and triggering preventive maintenance of
equipment
5. Monitoring Children in the home from outside
6. Online streaming of events
7. Entertainment – Contents sharing platform (OTT), Internet TV, Web
Serials
8. Connected Machines – Manufacturing 0

Web Application
A web-application is an application program that is usually stored on a
remote server, and users can access it through the use of Software known as
web-browser.
In general, a web application can contain online shops (or we can also say
them e-commerce shops), webmail's, calculators, social media platforms,
etc. There is also some kind of web application that usually requires a special
kind of web browser to access them. We cannot access those kinds of web
applications by using regular web- browsers. However, most of the web
applications available on the internet can be accessed using a standard web
browser.
a web application usually uses a combination of the server-side scripts such
as PHP, ASP, for handling the information/ data storage and retrieval of the
data.
Some of them also use the client-side scripts such as JavaScript, HTML to
represent the data/information in front of the users, and some of the web
applications are also using both server-side and client-side at the same time.
It allows the users to communicate with the organization or companies by
using the online form, online forums, shopping carts, content management
system, and much more.
Apart from that web applications also allow its users to create documents,
share them, or share the data/ information. By using the web application,
users can collaborate on same projects by event when they are not available
on the same geographical location.
The Flow of the Web Application
Let's understand how the flow of the typical web application looks like.

1. In general, a user sends a request to the web-server using web

browsers such as Google Chrome, Microsoft Edge, Firefox, etc
over the internet.
2. Then, the request is forwarded to the appropriate web application
server by the web-server.
3. Web application server performs the requested operations/ tasks
like processing the database, querying the databases;
produces the result of the requested data.
4. The obtained result is sent to the web-server by the web application
server along with the requested data/information or processed data.
5. The web server responds to the user with the requested or processed
data/information and provides the result to the user's screen .
Benefits of a web application
Let see some of the significant benefits offered by a web application:
o Any typical web application can run or accessible on any operating
system such as the Windows, Mac, Linux as long as the browser is
compatible.
o A web application is usually not required to install in the hard drive of
the computer system, thus it eliminates all the issues related to the
space limitation.
o All the users are able to access the same version of the web
application, which eliminates all compatibility issues.
o It also reduces software piracy in subscription-based web applications,
for example, SAAS (or Software as a service).
o They also reduce the expense for end-users, business owners because
the maintenance needed by the business is significantly less.
o Web applications are flexible. A user can work from any geographical
location as long as he has a working internet connection.
o It just takes a moment to create a new user by providing a username,
password, and URL, and it's all.
o After the availability of the cloud, storage space is now virtually
unlimited as long as you can afford it.
o A web application can be programmed to run on a wide variety of
operating systems, unlike native applications that can run on a
particular platform.
o Any standard web application is developed with some basic
programming languages like HTML, CSS that are compatible and well
known among the IT professionals.
Disadvantages of the Web Applications
As we all know, there are two sides of anything; if something has some
advantages, it may also have limitations/ disadvantages. Consider the
following disadvantages of the web applications.
o Internet connection is necessary to access any web application, and
without an internet connection, anyone can't use any of the web
applications. It is very typical to get an internet connection in our
modern cities, still rural area internet connectivity not so well.
o Several people in business believe that their data on the cloud
environment is no that secure and likes to stick with old methods; they
even don't want to use new methods.
o As we all know that many users like to use different web browsers
according to their needs and choices. So while creating a web
application, you must remember that your application must support
several web browsers, including new and old versions of browsers.
o Speed-related issues are also affecting the web application's
performance because there are several factors on which the
performance of a web application depends, and these all factors affect
the performance of the web application in their own way.
o If a user's web application faces any kind of issues, or if he does not
have a good quality corporate website, his web application will not be
going to run correctly, smoothly.
 o A user must have to spend enough money to maintain the good
condition of his web application, provide an update whenever an issue
occurs, and make an attractive user interface, which is not so cheap at
all.
o A web application must be programmed/ coded in such a way that it
will be run regardless of the device's operating system.
o A web application may face some issues while running on Windows,
Android, or several other operating systems if it is not responsive.
There are several advantages and disadvantages of web applications; it is
impossible to discuss them all at once. So in the above, we only discussed
some critical and useful points that may help you quickly understand the
pros and cons of the web application.

REFERENCES:
Uses of Internet | Top 9 Uses of Internet along with Applications
([Link])
Telecommunication | Technology, Examples, Devices, & Facts | Britannica
Telecommunication | Technology, Examples, Devices, & Facts | Britannica