DCS Lecture Notes - 1

LECTURE NOTES
ON
DATA COMMUNICATION
II B.Tech II semester (JNTUH-R15)
Mrs. J.SRAVANA
Assistant professor
ELECTRONICS AND COMMUNICATION ENGINEERING
INSTITUTE OF AERONAUTICAL ENGINEERING

DUNDIGAL, HYDERABAD - 500043
UNIT-1
INTRODUCTION TO DATA COMMUNICATIONS AND NETWORKING

Data communication-(A40409) Page 3
UNIT -2
MULTIPLEXERS
Introduction
It has been observed that most of the individual data-communicating devices typically
require modest data rate. But, communication media usually have much higher bandwidth. As a
consequence, two communicating stations do not utilize the full capacity of a data link.
Moreover, when many nodes compete to access the network, some efficient techniques for
utilizing the data link are very essential. When the bandwidth of a medium is greater than
individual signals to be transmitted through the channel, a medium can be shared by more than
one channel of signals. The process of making the most effective use of the available channel
capacity is called Multiplexing.
For efficiency, the channel capacity can be shared among a number of communicating
stations just like a large water pipe can carry water to several separate houses at once. Most
common use of multiplexing is in long-haul communication using coaxial cable, microwave and
optical fibre.
Figure 2.1 depicts the functioning of multiplexing functions in general. The multiplexer is
connected to the demultiplexer by a single data link. The multiplexer combines (multiplexes)
data from these n input lines and transmits them through the high capacity data link, which is
being demultiplexed at the other end and is delivered to the appropriate output lines. Thus,
Multiplexing can also be defined as a technique that allows simultaneous transmission of
multiple signals across a single data link.
Figure 2.1 Basic concept of multiplexing
Multiplexing techniques can be categorized into the following three types:
Frequency-division multiplexing (FDM): It is most popular and is used extensively in

radio and TV transmission. Here the frequency spectrum is divided into several logical channels,
giving each user exclusive possession of a particular frequency band.
Time-division Multiplexing (TDM): It is also called synchronous TDM, which is
commonly used for multiplexing digitized voice stream. The users take turns using the entire
channel for short burst of time.
Statistical TDM: This is also called asynchronous TDM, which simply improves on the
efficiency of synchronous TDM.
Frequency-Division Multiplexing (FDM) :
In frequency division multiplexing, the available bandwidth of a single physical medium

is subdivided into several independent frequency channels. Independent message signals are
translated into different frequency bands using modulation techniques, which are combined by a
linear summing circuit in the multiplexer, to a composite signal.
The resulting signal is then transmitted along the single channel by electromagnetic
means as shown in Fig. 2.2. Basic approach is to divide the available bandwidth of a single
physical medium into a number of smaller, independent frequency channels. Using modulation,
independent message signals are translated into different frequency bands. All the modulated
signals are combined in a linear summing circuit to form a composite signal for transmission.
The carriers used to modulate the individual message signals are called sub-carriers, shown as
f1, f2, , fn in Fig. 2.3 (a).
Figure 2.2 Basic concept of FDM
At the receiving end the signal is applied to a bank of band-pass filters, which separates
individual frequency channels. The band pass filter outputs are then demodulated and distributed
to different output channels as shown in Fig. 2.3(b).

Figure 2.3 (a) FDM multiplexing process, (b) FDM demultiplexing process
Figure 2.4 Use of guard bands in FDM
If the channels are very close to one other, it leads to inter-channel cross talk. Channels must be
separated by strips of unused bandwidth to prevent inter-channel cross talk. These unused
channels between each successive channel are known as guard bands as shown in Fig. 2.4.
FDM are commonly used in radio broadcasts and TV networks. Since, the frequency
band used for voice transmission in a telephone network is 4000 Hz, for a particular cable of 48
KHz bandwidth, in the 70 to 108 KHz range, twelve separate 4 KHz sub channels could be used
for transmitting twelve different messages simultaneously. Each radio and TV station, in a
certain broadcast area, is allotted a specific broadcast frequency, so that independent channels
can be sent simultaneously in different broadcast area. For example, the AM radio uses 540 to
1600 KHz frequency bands while the FM radio uses 88 to 108 MHz frequency bands.

TIME-DIVISION MULTIPLEXING (TDM) :
In frequency division multiplexing, all signals operate at the same time with different
frequencies, but in Time-division multiplexing all signals operate with same frequency at
different times. This is a base band transmission system, where an electronic commutator
sequentially samples all data source and combines them to form a composite base band signal,
which travels through the media and is being demultiplexed into appropriate independent
message signals by the corresponding commutator at the receiving end. The incoming data from
each source are briefly buffered. Each buffer is typically one bit or one character in length. The
buffers are scanned sequentially to form a composite data stream. The scan operation is
sufficiently rapid so that each buffer is emptied before more data can arrive. Composite data rate
must be at least equal to the sum of the individual data rates. The composite signal can be
transmitted directly or through a modem. The multiplexing operation is shown in Fig. 2. 7
Figure 2.7 Time division multiplexing operation
As shown in the Fig 2.7 the composite signal has some dead space between the successive
sampled pulses, which is essential to prevent interchannel cross talks. Along with the sampled
pulses, one synchronizing pulse is sent in each cycle. These data pulses along with the control
information form a frame. Each of these frames contain a cycle of time slots and in each frame,
one or more slots are dedicated to each data source. The maximum bandwidth (data rate) of a
TDM system should be at least equal to the same data rate of the sources.

Synchronous TDM is called synchronous mainly because each time slot is preassigned to
a fixed source. The time slots are transmitted irrespective of whether the sources have any data to
send or not. Hence, for the sake of simplicity of implementation, channel capacity is wasted.
Although fixed assignment is used TDM, devices can handle sources of different data rates. This
is done by assigning fewer slots per cycle to the slower input devices than the faster devices.
Both multiplexing and demultiplexing operation for synchronous TDM are shown in Fig. 2. 8.

STATISTICAL TIME DIVISION MULTIPLEXING:
One drawback of the TDM approach, as discussed earlier, is that many of the time slots
in the frame are wasted. It is because, if a particular terminal has no data to transmit at particular
instant of time, an empty time slot will be transmitted. An efficient alternative to this
synchronous TDM is statistical TDM, also known as asynchronous TDM or Intelligent TDM. It
dynamically allocates the time slots on demand to separate input channels, thus saving the
channel capacity. As with Synchronous TDM, statistical multiplexers also have many I/O lines
with a buffer associated to each of them. During the input, the multiplexer scans the input
buffers, collecting data until the frame is filled and send the frame. At the receiving end, the
demultiplexer receives the frame and distributes the data to the appropriate buffers. The
difference between synchronous TDM and asynchronous TDM is illustrated with the help of Fig.
2.9. It may be noted that many slots remain unutilised in case synchronous TDM, but the slots
are fully utilized leading to smaller time for transmission and better utilization of bandwidth of
the medium. In case of statistical TDM, the data in each slot must have an address part, which
identifies the source of data. Since data arrive from and are distributed to I/O lines unpredictably,
address information is required to assure proper delivery as shown in Fig. 2.10.This leads to
more overhead per slot. Relative addressing can be used to reduce overhead
2.9 Synchronous vs. Asynchronous TDM

2.10 Address overhead in asynchronous TDM

TRANSMISSION MEDIA

OPTICAL FIBER COMMUNICATION:

Unit-3
TELEPHONE INSTRUMENTS AND SIGNALS
Telephone instruments and signals:
The Subscriber Loop, Standard Telephone Set, Basic Telephone Call Procedures, Call Progress
Tones and Signals, Cordless Telephones, Caller ID, Electronic Telephones, Paging systems.
Telecommunication is the transmission of messages, over significant distances, for the
purpose of communication. Telecommunications has typically involved the use of electric means
such as the telegraph, the telephone, and the teletype, the use of microwave communications, the
use of fiber optics and their associated electronics, and/or the use of the Internet.
The Subscriber Loop:

The simplest and most straight forward form of telephone service is called plain old telephone
services(POTS), which involves subscribers accessing the public telephone network through pair
of wires called the local subscriber loop (or simply local loop). A local loop is simply an
unshielded twisted - pair transmission line (cable pair), consisting of two insulated conductors
twisted together.
The subscriber loop provides the means to connect a telephone set at a subscriber location to the
closest telephone office, which is commonly called an end office, locale change office, or central
office. Once in the central office, the subscriber loop is connected to an electronic switching
system (ESS), which enables the subscriber to access the telephone network.
Standard Telephone Set:
A telephone is defined as an apparatus for reproducing sound, especially that of the human
voice (speech) at a great distance, by means of electricity, consisting of transmitting and
receiving instruments connected by a line or wire which conveys the electric current.
Functions of the telephone set:

1. Notify the subscriber when there is an incoming call with an audible signal, such as
a bell, or with a visible signal, such as a flashing light. This signal is analogous to an
interrupt signal on a microprocessor, as its intent is to interrupt what you are doing.
These signals are purposely made annoying enough to make people want to answer the telephone
as soon as possible.
2. Provide a signal to the telephone network verifying when the incoming call has been
acknowledged and answered (i.e., the receiver is lifted off hook).
3. Convert speech (acoustical) energy to electrical energy in the transmitter and vice versa in the
receiver. Actually, the microphone converts the acoustical energy to mechanical energy, which is
then converted to electrical energy. The speaker performs the opposite conversions.
4. Incorporate some method of inputting and sending desti nation telephone numbers (either
mechanically or electrically) from the telephone set to the central office switch over the local
loop. This is accomplished using either rotary diallers (pulses) or TouchTone pads (frequency
tones).
5. Regulate the amplitude of the speech signal the calling person outputs onto the telephone Line.
This prevents speakers from producing signals high enough in amplitude to interfere with other
peoples conversations taking place on nearby cable pairs (crosstalk).
6. Incorporate some means of notifying the telephone office when a subscriber wishes to place an
outgoing call (i.e., handset lifted off hook). Subscribers cannot dial out until they receive a dial
tone from the switching machine.
7. Ensure that a small amount of the transmit signal is fed back to the speaker enabling talkers to
hear themselves speaking. This feedback signal is sometimes called side tone or talkback. Side
tone helps prevent the speaker from talking too loudly.
8. Provide an open circuit (idle condition) to the local loop when the telephone is not in use (i.e.,
on hook) and a closed circuit (busy condition) to the local loop when the telephone is in use (off
hook).
9. Provide a means of transmitting and receiving call progress signals between the central office
switch and the subscriber, such as on and off hook, busy, ringing, dial pulses. Touch-Tone
signals, and dial tone.

Telephone Set, Local Loop and Central Office Switching machines:
A basic telephone set requires only two wires and these pair of wires connecting a subscriber to the
closest telephone office is called the local loop. One wire is called the tip, and the other is called the
ring. In case, a third wire is used, its called the sleeve. Since the 1960s, phone plugs and jacks have
gradually been replaced in the home with miniaturized plastic plug known as RJ-11 and a matching
plastic receptacle shown in figure. RJ stands for registered jacks and is sometimes described as RJ-XX. RJ
is a series of telephone connection interfaces (receptacle and plug) that ar e registered with the U.S.
Federal Communications Commission (FCC). The term jack sometimes describes both the receptacle and
the plug and sometimes specifies only the receptacle. RJ -11 is the most
common telephone jack in use today and can have up to six conductors.
Block Diagram of a Telephone Set:
A telephone set is an apparatus that creates an exact likeness of sound waves with an electric
current. The essential components of a telephone set are the ringer circuit, on/off hook circuit,
equalizer circuit, hybrid circuit, speaker, microphone, and a dialing circuit.
Functional block diagram of a standard telephone set:
Ringer circuit: The ringer circuit, which was originally an electromagnetic bell, is placed directly
across the tip and ring of the local loop and its sole purpose is to alert the destination party of
incoming calls. The tone of the ringer should be loud enough to be heard from a distance. In
modern telephones, the bell has been replaced with and electronic oscillator connected to the
speaker. Today, ringing signals can be of any imaginary sound.
On/off hook circuit: The on/off hook circuit (sometimes called a switch hook) is nothing more
than a simple single-throw double-pole (STDP) switch placed across the tip and ring. The switch
is mechanically connected to the telephone handset so that when the telephone is idle (on hook),
the switch is open. When the telephone is in use (off hook), the switch is closed, completing an
electrical path through the microphone between the tip and ring of the local loop.
Equalizer circuit: Equalizers are combinations of passive components (resistors, capacitors, and
so on) that are used to regulate the amplitude and frequency response of the voice signals. The
equalizer helps solve an important transmission problem in telephone set design, namely, the
interdependence of the transmitting and receiving efficiencies and the wide range of transmitter
currents caused by a variety of local loop cables with different dc resistances.
Speaker: The speaker is the receiver for the telephone. The speaker converts electrical signals
received from the local loop to acoustical signals (sound waves) that can be heard and
understood by a human being. The speaker is connected to the local loop through the hybrid
network. The speaker is typically enclosed in the handset of the telephone along with the
microphone.
Microphone: The microphone is the transmitter for the telephone and it converts acoustical
signals in the form of sound pressure waves from the caller to electrical signals that are
transmitted into the telephone network through the local subscriber loop. The microphone is also
connected to the local loop through the hybrid network. Both the microphone and the speaker are
transducers, as they convert one form of energy into another form of energy. A microphone
converts acoustical F energy first to mechanical
energy and then to electrical energy, while the speaker performs the exact opposite sequence of
conversions
Hybrid network: The hybrid network (sometimes called a hybrid coil or duplex coil) in a
telephone set is a special balanced transformer used to convert a two -wire circuit (the local loop)
into a four-wire circuit (the telephone set) and vice versa, thus enabling full-duplex operation
over a two-wire circuit. In essence, the hybrid network separates the transmitted signals from the
received signals. Another function of the hybrid network is to allow a small portion of the
transmit signal to be returned to the receiver in the form of a sidetone.Dialing circuit: The
dialling circuit enables the subscriber to output signals representing digits, and this enables the
caller to enter the destination telephone number. The dialling circuit could be a rotary dialer, or
most likely an electronic dial-pulsing circuit (or a Touch-Tone keypad) which sends various
combinations of tones representing the called digits.
Basic Telephone Call Procedures:
When the calling party takes the telephone set off the hook, the switch hook in the set is released,
completing a dc path between the tip and ring of the loop through the microphone. The ESS
(electronic switching system) senses a dc current in the loop and recognizes this as an off-hook
condition and this procedure is referred to as loop start operation. Completing a local telephone
call between two subscribers connected to the same telephone switch is accomplished through a
standard set of procedures including the following 10 steps. This manner of accessing the
telephone system is known as POTS.
Step 1:- Calling station goes off hook.
Step 2:- After detecting a dc current flow on the loop, the switching machine returns an audible
dial tone to the calling station, acknowledging that the caller has access to the switching
machine.
Step 3:- The caller dials the destination telephone number using one of two methods:
mechanical dial pulsing or, more likely, electronic dual-tone multifrequency (TouchTone)
signals.
Step 4:-When the switching machine detects the first dialed number, it removes the dial tone
from the loop.
Step 5:-The switch interprets the telephone number and then locates the local loop for the
destination telephone number.
Step 6:- Before ringing the destination telephone, the switching machine tests the destination
loop for dc current to see if it is idle (on hook) or in use (off hook). At the same time, the
switching machine locates a signal path through the switch between the two local loops.
Step 7a:- if the destination telephone is off hook, the switching machine sends a station busy
signal back to the calling station.
Step7b:- If the destination telephone is on hook, the switching machine sends a ringing signal
to the destination telephone on the local loop and at the same time sends a ring back signal to
the calling station to give the caller some assurance that something is happening.

Step 8:- When the destination answers the telephone, it completes the loop, causing dc current
to flow.
Step 9:- The switch recognizes the dc current as the station answering the telephone. At this
time, the switch removes the ringing and ring-back signals and completes the path through the
switch, allowing the calling and called parties begin their conversation.
Step 10:- When either end goes on hook, the switching machine detects an open circuit on that
loop and then drops the connections through the switch.
Call Progress Tones and Signals:
Call progress tones and call progress signals are acknowledgment and status signals that ensure
the processes necessary to set up and terminate a telephone call are completed in an orderly and
timely manner. Call progress tones and signals can be sent from machines to machines, machines
to people, and people to machines.Signalling can be broadly divided into two major
categoriesstation signalling and interoffice signalling. Station signalling is the exchange
of signalling messages over local loops between stations (telephones) and telephone company
switching machines. On the other hand, interoffice signalling is the exchange of signalling
messages between switching machines. Signaling messages can be subdivided further into one of
four categories: alerting, supervising, controlling, and addressing. Alerting signals indicate a
request for service, such as going off hook or ringing the destination telephone. Supervising
signals provide call status information, such as busy or ring-back signals. Controlling signals
provide information in the form of announcements, such as number changed to another number,
a number no longer in service, and so on. Addressing signals provide the routing information,
such as calling and called numbers.
Examples of essential call progress signals are dial tone, dual-tone multifrequency tones,
multifrequency tones, dial pulses, station busy, equipment busy, ringing, ring -back receiver on
hook, and receiver off hook.
Call Progress Tone Summary:
Dial Tone: Dial tone is an audible signal comprised of two frequencies: 350 Hz and 440 Hz. Dial
tone informs subscribers that they have acquired access to the electronic switching machine and
can now dial or use Touch-Tone in a destination telephone number. After the subscriber hears
the dial tone and starts dialling, it is removed and this condition is called breaking dial tone.
Sometimes, dial tone may not be heard even in off hook condition and this condition is called no
dial tone.
Dual-Tone Multifrequency (DTMF): DTMF was originally called Touch-Tone. DTMF is a more
efficient means than dial pulsing for transferring telephone numbers from a subscribers location
to the central office switching machine. DTMF is a simp le two-of-eight encoding scheme where
each digit IS represented by the linear addition of two frequencies.
DTMF is strictly for signaling between a Subscribers location and the nearest telephone office
or message switching center. The following figure shows the four-row-by-four column keypad
matrix used with DTMF keypad.The keypad is comprised of 16 keys and eight frequencies. The
four vertical frequencies (low group frequencies) are 697 Hz, 770 Hz, 852 Hz and 941 Hz, and
the four horizontal frequencies (high group frequencies) are 1209 Hz, 1336 Hz, 1447 Hz and
1633 Hz. The digits 2 through 9 can also be used to represent 24 of the 26
letters. When a digit is pressed, two of the eight frequencies (one from either group) are
transmitted. The major advantage of using Touch-Tone signalling over dial pulsing is speed
and control. Here, all digits take the same length of time to produce and transmit and
also it eliminates the impulse noise produced by mechanical switches used in dial pulses.
Multifrequency (MF): MF tones (codes) are similar to DTMF signals in that they
involve the simultaneous transmission of two tones. MF tones are used to transfer digits
and control signals between switching machines. MF tones are combinations of two frequencies
that fall within the normal speech bandwidth so that they can be propagated over the same
circuits as voice, which is called in-band signalling. The two-tone MF combinations and the
digits or control information they represent is shown below.
Multifrequency Codes MF codes are used to transmit the calling and called numbers from the
originating telephone office to the destination telephone office. MF tones involve transmission of
two of the six possible frequencies representing the 10 digits plus two control signals. The key
pulse (KP) signal is used to indicate the beginning of a sequence of dialled digits. The start (ST)
signal is used to indicate the end of a sequence of dialled digits.
Dial Pulses: Dial pulsing (sometimes called rotary dial Pulsing) is the method originally used to
transfer digits from a telephone set to the local switch.The process begins when the telephone set
is lifted off hook, completing a path for current through the local loop. When the switching
machine detects the 0ff -hook Condition, it responds with dial tone. After hearing the dial
tone, the subscriber begins dial pulsing digits by rotating a mechanical dialling
mechanism and then letting it return to its rest position. As the rotary switch returns to its
rest position, it outputs a series of dial pulses corresponding to the digit dialled. When a digit is
dialled, the loop circuit alternately opens (breaks) and closes (makes) a prescribed number of
times. The number of switch make/break sequences corresponds to the digit dialled (i.e.,
the digit 3 produces three switch openings and three switch closures). Dial pulses occur at 10
make/break cycles per second (i.e., a period of 100 ms per pulse cycle). All digits do not take
the same length of time to dial.
Station Busy: A station-busy signal is sent from the switching machine back to the calling station
whenever the called telephone number is off hook (i.e., the station is in use). The station-busy
signal is a two-tone signal comprised of 480 Hz and 620 Hz. The two tones are on for 0.5
seconds, then off for 0.5 seconds. Thus, a busy signal repeats at a 60 -pulse-per minute (ppm)
rate.
Equipment Busy: The equipment-busy signal is sometimes called a congestion tone or a no-
circuits-available tone. The equipment-busy signal is sent from the switching machine back to
the calling station whenever the system cannot complete the call because of equipment
unavailability. This condition is called blocking and occurs whenever the system is overloaded
and more calls are being placed than can be comp leted. The equipment-busy signal uses the
same two frequencies as the station-busy signal, signal except the equipment-busy signal is on
for 0.2 seconds and off for 0.3 seconds (120 ppm).
Ringing: The ringing signal is sent from a central office to a subs criber whenever there is an
incoming call and its main purpose is to alert the subscriber that there is an incoming call. The
ringing signal is nominally a 20-Hz, 90-vrms signal that is on for 2 seconds and then off for 4
seconds.
Ring-Back: It is sent to the calling party at the same time the ringing signal is sent to the caller
party. The purpose of the ring-back signal is to assure the calling party that the destination
telephone number has been accepted and processed and is being rung. It is an audible
combination of two tones at 440 Hz and 480 Hz that are on 2 seconds and then off for 4 seconds.
Receiver On/Off Hook: When the telephone is on hook, the circuit is in idle state and there is no
current flowing on the loop. An on -hook signal is also used to terminate a call and initiate a
disconnect. When the telephone set is off hook, switch closure occurs causing a dc current to
flow on the loop. The receiver off-hook condition is the first step in completing a
telephone call. It is also used at the destination end as an answer signal to indicate that
the called party has answered the telephone.
Other Nonessential Signals and Tones: Some of the others are call waiting tones, caller waiting
tones, calling card service tones, comfort tones, hold tones, in trusion tones, stutter dial tones
etc..
Cordless Telephones:
Cordless telephones are simply telephones that operate without cords attached to the handset. It
is a full duplex, battery operated, portable radio transceiver that communicates directly with a
stationary transceiver located somewhere in the subscribers home or office.Cordless Telephone
System.The base station is an ac-powered stationary radio ransceiver capable of transmitting and
receiving both supervisory and voice signals over the subscri ber loop in the same manner as a
standard telephone. It also must be capable of relaying voice and control signals to and from the
portable telephone set through the wireless transceiver. The portable telephone set is a battery-
powered, two-way radio capable of operating in the full duplex mode. As it uses a full duplex
mode, it must transmit and receive at
different frequencies. Base stations transmit on high-band frequencies and receive on low-
band frequencies, while the portable unit transmits on low-band frequencies and receives
on high-band frequencies.
Cordless telephones using the 2.4 GHz band offer excellent sound quality utilizing digital
modulation and twin-band transmission to extend their range. With twin-band transmission, base
stations transmit in the 2.4 GHz band, while portable units transmit in the 902 MHz to 928 MHz
band.
Electronic Telephone Set:
A typical electronic telephone comprised of one multifunctional integrated- circuit chip, a
microprocessor chip, a Touch-Tone keypad, a speaker, a microphone, and several discrete
devices is shown above. The major components included in the multifunctional integrated circuit
chip are DTMF tone generator, MPU (microprocessor unit) interface circuitry, random access
memory (RAM), tone ringer circuit, speech network and a line voltage regulator.
The Touch-Tone keyboard provides a means for the operator of the telephone to access
the DTMF tone generator inside the multifunction integrated-circuit chip. The external
crystal provides a stable and accurate frequency reference for producing the dualtone
multifrequency signalling tones. Once a ringing signal occurs, the tone ringer circuit
activates and drives a piezoelectric sound element that produces an electronic ring. The

voltage regulator converts the dc voltage received from the local loop to a constant- level dc
supply voltage to operate the electronic components in the phone. The internal speech network
contains several amplifiers and associated components as in standard telephone.
The microprocessor interface circuit interfaces the MPU to the multifunction chip. The
MPU, with its internal RAM, controls many of the functions of the telephone, such as number
storage, speed dialling, redialling and autodialing. The bridge rectifier protects the telephone
from the relatively high-voltage ac ringing signal, and the switch hook is a mechanical
switch that performs the same functions as the switch hook on a standard telephone set.
Paging Systems:
Paging transmitters relay radio signals and messages from wire-line and cellular telephones to
subscribers carrying portable receivers.
Standard Simplex Paging System
Standard paging systems are one-way, with signals transmitted from the paging system to
portable pager and never in reverse direction. There are narrow-, mid- and widearea pagers. To
contact a person carrying a pager, the telephone number assigned to that pager has to be dialed.
The paging company receives the call and requests the number the pager person has to call. After
the number is entered a terminating signal is appended to the number (#). Then paging system
converts it to a digital code and transmits it in the form of a digitally encoded signal over a
wireless communications system. If the paged person is within the range of a broadcast
transmitter, the targeted pager will receive the message and the number to be called will be
shown on an alphanumeric display.Early paging systems used FM, but modern systems are using
FSK or PSK. Each portable pager is assigned a special code called a cap code, which is a
sequence of digits or a combination of digits and letters. It is broadcast along with the paging
partys telephone number. Upon receiving the signal, the paging unit with demodulate and
recognize the cap code. Once its been recognized, then the call-back number and may be a
message will be displayed on the unit.
The most recent paging protocol developed is FLEX, which has been designed to minimize
power consumption in the portable pager by using a synchronous time-slotted protocol to
transmit messages in precise time slots. Each frame consists of 128 data frames, transmitted only
once during a 4-minute period. Each frame lasts for 1.875 seconds and includes two
synchronizing sequences, a header containing f rame information and pager identification
addresses, and 11 discrete data blocks.
Caller ID:
Caller ID (identification) enables the destination stations of a telephone call to display the name
and telephone number of the calling party before the telephone is answered. This allows
subscribers to screen incoming calls and decide whether they want to answer the telephone. The
caller ID message is a simplex transmission sent from the central office switch over the local
loop to a caller ID display unit at the destination station.
To ensure detection of caller ID signal, the telephone must ring at least twice before being
answered as a 3-second window is present in between in which the caller ID signal must be
transmitted The Local Subscriber Loop, Telephone Message- Channel Noise and Noise
Weighting, Units of Powers Measurement, Transmission Parameters and Private-Line
Circuits, VoiceFrequency Circuit Arrangements, Crosstalk.A telephone circuit is comprised
of two or more facilities, interconnected in tandem, to provide a transmission path
between a source and a destination. The facilities may be metallic cable pairs, optical
fibers, or wireless carrier systems. The information transferred is called the message, and the
circuit used is called the message channel.
The Local Subscriber Loop
The local subscriber loop is the only facility required by all voice-band circuits, as it is the means
by which subscriber locations are connected to the local telephone company. The sole
purpose of a local loop is to provide subscribers access to the public telephone network.
The local loop is the primary cause of attenuation and phase distortion occurs when two
or more frequencies undergo different amount of phase shift. There are seven main
component parts that make up a traditional local loop: Feeder cable (F1), Serving area interface
(SAI), Distribution cable (F2), Subscriber or standard network interface (SNI), Drop wire,
Aerial and Distribution cable and drop-wire cross-connect point. Two components often
found on local loops are loading coils and bridge taps.Loading Coils: Loading coils placed in
a cable decrease the attenuation, increase the line impedance and improve transmission levels
for circuits longer than 18,000 feet. Loading coils allowed local loops to extend three to four
times their previous length. A loading coil is simply a passive conductor wrapped around a core
and placed in series with a cable creating a small electromagnet. Loading coils can be placed on
telephone poles, in manholes, or on cross-connect boxes. Loading coils increase the effective
distance that a signal must travel between two locations and cancels the capacitance that
inherently builds up between wires with distance. Loading coils cause a sharp drop in frequency
response at approximately 3400 Hz, this is undesirable for high-speed data transmission.
Therefore, for highperformance data transmission, loading coils should be removed from the
cables.
Bridge Taps: A bridge tap is an irregularity frequently found in cables serving subscriber
locations. Bridge taps are unused sections of cable that are connected in shunt to a working
cable pair, such as a local loop. Bridge taps can be placed at any point along a cables length.
Bridge taps increase the flexibility of a cable by making it easier to reassign a cable to a different
subscriber without requiring a person working in the field to cross connect sections of cable.
Bridge taps introduce a loss called bridging loss. They also allow signals to split and propagate
down more than one wire. Bridge taps that are short and closer to the originating or terminating
ends often produce the most interference. Bridge taps and loading coils are not generally harmful
to voice transmissions, but if improperly used, they can literally destroy the integrity of a data
signal.Loop Resistance: The dc resistance of a subscriber local loop is called loop resistance. It
depends primarily on the type of wire and wire size. Most local loops use 18 - to 26-gauge,
twisted-pair copper wire. The dc loop resistance for copper conductors is approximated by Rd=
0.1095/d2
Where, Rd = dc loop resistance (ohms per mile)
d = wire diameter (inches)

Units of Power Measurement
The decibel (dB) is the basic yardstick used for making power measurements in communications.
The unit dB is simply a logarithmic expression representing the ratio of one power level to
another and expressed as: dB = 10 log (P1 / P2), where P1 and P2 are power levels at two
different points in a transmission system.
The unit dBm is often used to reference the power level at a given point to 1 milliwatt. One
milliwatt is the level from which all measurements are referenced. It is expressed mathematically
as: dBm = 10 log (P / 1 mW), where P is the power at any point in a transmission system.
Transmission level point (TLP) is defined as the optimum level of a test tone on a channel at
some point in a communications system. The transmission level (TL) at any point in a
transmission system is the ratio in dB of the power of a signal at that point to the power the same
signal would be at a 0-dBm transmission level point. Data level point (DLP) is a parameter
equivalent to TLP except TLP is used for voice circuits, whereas DLP is used as a reference for
data transmission. The DLP is always 13 dB below the voice level for the same point.
Transmission Parameters and Private -Line Circuits
Transmission parameters apply to dedicated private-line data circuits that utilize the private
sector of the public telephone network. Private-line data circuits have several advantages over
using the switched public telephone network:
Transmission characters are more consistent because the same facilities are used with every
transmission.
The facilities are less prone to noise produced in the telephone company switches.
Line conditioning is available only on private-line facilities
Higher transmission bit rates and better performance is appreciated with private-line data
circuits.
Private-line data circuits are more economical for high-volume circuits.
Transmission parameters are divided into three broad categories:
Bandwidth parameters:- attenuation distortion and envelope delay distortion
Interface parameters:- terminal impedance, in-band and out-of-band signal power, test signal
power, and ground isolation
Facility parameters:- noise measurements, frequency distortion, phase distortion,
amplitude distortion, and nonlinear distortion.
Bandwidth Parameters
Attenuation distortion is the difference in circuit gain experienced at a particular frequency with
respect to the circuit gain of a reference frequency. This characteristic is sometimes referred to
as frequency response, differential gain, and 1004-Hz deviation.
Envelope delay distortion is an indirect method of evaluating the phase delay characteristics of
a circuit. To reduce attenuation and envelope delay distortion and improve the
performance of data modems operating over standard message channels, it is often
necessary to improve the quality of the channel. The process used to improve a basic telephone
channel is called line conditioning. Line conditioning improves the high frequency response of a
message channel and reduces power losses.
Telephone companies offer two types of special line conditioning for subscriber loops:
C-type and D-type.
C-type Line conditioning:- C-type conditioning specifies the maximum limits for attenuation
distortion and envelope delay distortion. C-type conditioning pertains to line impairments for
which compensation can be made with filters and equalizers. There are five classifications or
levels of C-type conditioning available. The grade of conditioning a subscriber selects depends
on the bit rate, modulation technique and desired performance of the data modems used on the
line.
The five classifications of C-type conditioning are the following: C1
and C2 conditioning pertain to two-point and multipoint circuits.
C3 conditioning is for access lines and trunk circuits associated with private switched networks.
C4 conditioning pertains to two-point and multipoint circuits with a maximum of four stations.
C5 conditioning pertains only to two-point circuits.
D-type Line conditioning: - D-type conditioning neither reduces the noise on a circuit
nor improves the signal-to-noise ratio. It simply sets the minimum requirements for signal-
to noise (S/N) ratio and nonlinear distortion. D-type conditioning is sometimes referred to as
high performance conditioning and can be applied to private-line data circuits in addition to
either basic or C-conditioned requirements.
There are two categories for D-type conditioning: Dl and D2.
Limits imposed by Dl and D2 are virtually identical. The only difference between the two
categories is the circuit arrangement to which they apply. Dl conditioning specifies
requirements for two-point circuits and D2 conditioning specifies requirements for
multipoint circuits. D-type conditioning is mandatory when the data transmission rate is
9600 bps because without D-type conditioning, it is highly unlikely that the circuit can meet the
minimum performance requirements guaranteed by the telephone company.
D-type conditioned circuits must meet the following specifications:

Signal-to-C-notched noise ratio: 28 dB
Nonlinear distortion
Signal-to-second-order distortion: 35 dB
Signal-to-third-order distortion: 40 dB
The signal-to-notched noise ratio requirement for standard circuits is only 24 dB, and they have
no requirements for nonlinear distortion.
Interface Parameters
The two primary considerations of the interface parameters are electrical protection of the
telephone network and its personnel and standardization of design arrangements. The interface
parameters include the following:
Station equipment impedances should be 600 resistive over the usable voice band.
Station equipment should be isolated from ground by a minimum of 20M dc and 50 k ac
The basic voice-grade telephone circuit is a 3002 channel; it has an ideal bandwidth of 0 Hz to 4
kHz and a usable bandwidth of 300 Hz to 3000 Hz
The circuit gain at 3000 Hz is 3 dB below the specified in-band signal power.
The gain at 4 kHz must be at least 15 dB below the gain at 3 kHz
The maximum transmitted signal power for a private-line circuit is 0 dBm
Facility Parameters
Facility parameters represent potential impairments to a data signal. They include the following:
1OO4-Hz variation: - The telephone industry has established 1004 Hz as the standard testtone
frequency. The purpose of the 1004-Hz test tone is to simulate the combined signal power of a
standard voiceband data transmission. The 1004-Hz channel loss for a privateline data circuit is
typically 16 dB. A 1004-Hz test tone applied at the transmit end of a circuit should be received at
the output of the circuit at -16 dBm. Long-term variations in the gainof the transmission facility
are called 1004-Hz variation and should not exceed 4 dB. Thus, the received signal power must
be within the limits of -12 dBm to -20 dBm.C-message noise: - C-message noise measurements
determine the average weighted rms noise power. Unwanted electrical signals are produced from
the random movement of electrons in conductors. This type of noise is commonly called thermal
noise because its magnitude is directly
proportional to temperature. Because the electron movement is completely random and
travels in all directions, thermal noise is also called random noise, and because it contains all
frequencies, it is sometimes referred to as white noise. Thermal noise is inherently present in
a circuit because of its electrical makeup. Because thermal noise is additive, its magnitude is

dependent, in part, on the electrical length of the circuit.C-message noise measurements are the
terminated rms power readings at the receive end of a circuit with the transmit end terminated in
the characteristic impedance of the telephone line.
Impulse noise: Impulse noise is characterized by high-amplitude peaks (impulses) of short
duration having an approximately flat frequency spectrum. Impulse noise can saturate a message
channel. Impulse noise is the primary source of transmission errors in data circuits.
There are numerous sources of impulse noisesome are controllable, but most are not. The
primary cause of impulse noise is man-made sources, such as interference from ac power lines,
transients from switching machines, motors, solenoids, relays, electric trains, and so on. Impulse
noise can also result from lightning and other adverse atmospheric conditions.Gain hits and
dropouts: A gain hit is a sudden, random change in the gain of a circuit resulting in a temporary
change in the signal level. The primary cause of gain hits is noise transients (impulses) on
transmission facilities during the normal course of a day. A dropout is a decrease in circuit gain
(i.e., signal level) of more than 12 dB l asting longer than 4 ms. Dropouts are characteristics of
temporary open-circuit conditions and are generally caused by deep fades on radio facilities or
by switching delays.
Phase hits: Phase hits (slips) are sudden, random changes in the phase of a signal. Phase hits are
classified as temporary variations in the phase of a signal lasting longer than 4 ms. Phase hits,
like gain hits are caused by transients when transmission facilities are switched.
Phase jitter: Phase jitter is a form of incidental phase modulation-a continuous,
uncontrolled variation in the zero crossings of a signal. Generally, phase jitter occurs at
a 300-Hz rate or lower, and its primary cause is low-frequency ac ripple in power supplies.Single
frequency interference: Single-frequency interference is the presence of one or more
continuous, unwanted tones within a message channel. The tones are called spurious tones and
are often caused by crosstalk or cross modulation between adjacent channels in a transmission
system due to system nonlinearities. Spurious tones are measured by terminating the
transmit end of a circuit and then observing the channel frequency band. Spurious tones can
cause the same undesired circuit behaviour as thermal noise. Singlefrequency interference
is shown below:
Frequency Shift: Frequency shift is when the frequency of a signal changes during transmission.
Analog transmission systems used by telephone companies require coherent demodulation and
for this the receiver should be synchronous i.e. the frequency must be reproduced exactly in the
receiver. The longer a circuit, the more analog transmission systems and the more likely
frequency shift will occur.
Phase intercept distortion: Phase intercept distortion occurs in coherent SSBSC systems, such as
those using frequency division multiplexing when the received carrier is not reinserted with the
exact phase relationship to the received signal as the transmit carrier possessed.

Peak-to-average ratio:The difficulties encountered in measuring true phase distortion or envelope
delay distortion led to the development of peak-to-average ratio (PAR) tests. Low
peak-to-average ratios indicate the presence of differential delay distortion. PAR measurements
are less sensitive to attenuation distortion than EDD tests and are easier to perform.
Crosstalk
Crosstalk can be defined as any disturbance created in a communications channel by signals in
other communications channels (i.e., unwanted coupling from one signal path into another).
Crosstalk can originate in telephone offices, at a subscribers location, or on the facilities used to
interconnect subscriber locations to telephone offices. The nature of crosstalk is described as
intelligible or unintelligible. Intelligible crosstalk includes real loss of privacy. Unintelligible
crosstalk usually involves crosstalk between unlike channels, such as different types of carrier
facilities, frequency inversion and digital encoding.
There are three primary types of crosstalk: Nonlinear crosstalk, Transmittance crosstalk and
Coupling crosstalk.
Nonlinear crosstalk: - It is a direct result of nonlinear amplification in analogcommunications systems.
Nonlinear amplification produces harmonics and cross products (sum and difference frequencies). If
the nonlinear frequency components fall into the passband of another channel, they are considered
crosstalk. Nonlinear crosstalk can be distinguished from other types of crosstalk because the ratio of
the signal power in the disturbing channel to the interference power in the disturbed channel is a
function of the signal level in the disturbing channel.
Transmittance Crosstalk: - Crosstalk can also be caused by inadequate control of the frequency
response of a transmission system, poor filter design, or poor filter performance. This type of
crosstalk is most the prevalent when filters do not adequately reject undesired products from
other channels. As this type of interference is caused by inadequate control of the transfer
characteristics or transmittance of networks, it is called transmittance crosstalk.
Coupling Crosstalk: - Electromagnetic coupling between two or more physically isolated
transmission media is called coupling crosstalk. The most common coupling is due to the effects
of near-field mutual induction between cables from physically isolated circuits (i.e., when energy
radiates from a wire in one circuit to a wire in a different circuit). To reduce coupling crosstalk
due to mutual induction, wires are twisted together (hence the name twisted pair). Twisting the
wires causes a cancelling effect that helps eliminate crosstalk. The probability of coupling
crosstalk occurrence increases with cable length, signal power, and frequency.There are two
types of coupling crosstalk: near and far end. Near-end crosstalk
(NEXT) is crosstalk that occurs at the transmit end of a circuit and travels in the opposite
direction as the signal in the disturbing channel. Far-end crosstalk (FEXT) occurs at the far end
receiver and is the energy that travels in the same direction as the signal in the disturbing
channel.
Voice-Frequency Circuit Arrangements Two-Wire Voice-Frequency CircuitsTwo-wire
transmission involves two Wires (one for the signal and one for a reference ground) or a circuit
configuration that is equivalent to using only two wires. Two-wirecircuits are

ideally suited to simplex transmission, although they are often used for half - and full-duplex
transmission.
The above figure shows the simplest two-wire configuration, which is a passive circuit of
two copper wires connecting a telephone or voice-band modem at one station through a
telephone company interface to a telephone or voice -band modem at the destination
station.
The above figure shows an active two-wire transmission system (which provides gain). The one
difference between the above two circuits is the addition of an amplifier to compensate for
transmission line losses. The amplifier is unidirectional and thus, limits transmission to one
direction only (simplex).
The above circuit shows a two-wire circuit using a digital T carrier for the transmission medium.
This circuit requires a T carrier transmitter at one end and a T carrier receiver at the other end.
The digital T carrier transmission line is capable of two -way transmission but, the transmitter
and receiver are not. The digital transmission medium is a pair of copper wires.
The above circuit is an equivalent two-wire circuit as the transmission medium is earths
atmosphere and there are no copper wires between the two stations.
Four-Wire Voice-Frequency Circuits
Four-wire transmission involves four wires (two for each direction-a signal and areference) or a
circuit configuration that is equivalent to using four wires. Four-wire circuits are ideally suited to
full-duplex transmission, although they can (and very often do) operate in the half-duplex mode.
As with two-wire transmission, there are two forms of four-wire transmission systems: physical
four wires and equivalent four wire.Four wire circuits have several inherent advantages like they
are less noisy, have less crosstalk, and provide more isolation between the two directions of
transmission when operating in either half-duplex or full-duplex mode. But, two-wire circuits
require less wire, less circuitry, and thus less money than their four-wire counterparts. Four-wire
operation has a disadvantage of providing amplification.
The following figures show the block diagrams for four possible four-wire circuit configurations.
Hybrids, Echo Suppressors and Echo CancelersWhen a two-wire circuit is connected to a four-
wire circuit, an interface circuit is called a hybrid, or terminating, set. The hybrid circuit used to
convert two-wire circuits to four-wire circuits is similar to the hybrid coil found in
standard telephone sets.Hybrid (terminating) setsThe above figure shows the block diagram
for a two-wire to four-wire hybrid network. The hybrid coil compensates for impedance
variations in the two-wire portion of the circuit. The amplifiers and attenuators adjust the
signal power to required levels, and the equalizers compensate for impairments in the
transmission line that affect the frequency response of the transmitted signal, such as line
inductance, capacitance, and resistance. Signals travelling west to east (W-E) enter the
terminating set from the two-wire line, where they are inductively coupled into the west -to-east
transmitter section of the four-wire circuit. Signals received from the four-wire side of the hybrid
propagate through the receiver in the east-to-west (E-W) section of the four- wire circuit, where
they are applied to the center taps of the hybrid coils. If the impedances of the
two-wire line and the balancing network are properly matched, all currents produced in the

upper half of the hybrid by the E-W signals will be equal in magnitude but opposite in
polarity. Therefore, the voltages induced in the secondarys will be 1800 out of phase with
each other and, thus, cancel. This prevents any of the signals from being retransmitted to the
sender as an echo. If the impedances of the two-wire line and the balancing network are not
matched, voltages induced in the secondarys of the hybrid coil will not completely
cancel. This imbalance causes a portion of the received signal to be returned to the sender on
the W-E port ion of the four-wire circuit.Balancing networks can never completely match a
hybrid to the subscriber Loop because of long-term temperature variations and degradation
of transmission lines. The talker hears the returned portion of the signal as an echo, and
if the round-trip delay exceeds approximately 45 ms, the echo can become quite annoying. To
eliminate this echo, devices called echo suppressors are inserted at one end of the four-wire
circuit.
Echo Suppressor
The above figure shows a simplified block diagram of an echo suppressor. The speech
detector senses the presence and direction of the signal. It then enables the amplifier in
the appropriate direction and disables the amplifier in the opposite direction, thus
preventing the echo from returning to the speaker. A typical echo suppressor suppresses the
returned echo by as much as 60 dB. If the conversation is changing direction rapidly, the people
listening may be able to hear the echo suppressors turning on and off (every time an echo
suppressor detects speech and is activated, the first instant of sound isremoved from the message,
giving the speech a choppy sound). If both parties talk at the same time, neither person is heard
by the other. With an echo suppressor in the circuit, transmissions cannot occur in both
directions at the same time, thus limiting the circuit to half-duplex operation. Long-distance
carriers, such as AT&T, generally place echo suppressors in four-wire circuits that exceed 1500
electrical miles in length (the longer the circuit, the longer the round-trip delay time). Echo
suppressors are automatically disabled when they receive a tone between 2020 Hz and 2240 Hz,
thus allowing full-duplex data transmission over a circuit with an echo suppressor. Full- duplex
operation can also be achieved by replacing the echo suppressors with echo cancelers. Echo
cancelers eliminate the echo by electrically subtracting it from the original signal rather than
disabling the amplifier in the return circuit.
1.What is a telephone set? Describe in detail the various functional components of a standard
telephone set/
2. Discuss about the call progress tones and signals.
3. What are the steps involved in completing a local telephone call?
4. Expalin briefly how caller ID operates and when it is used.
5. What is crosstalk and what are the three types of crosstalk in telephone systems?
Compare near-end crosstalk and far-end crosstalk
6. What considerations are addressed by facility parameters? Compare phase hits and phase jitter
7. Compare the operation of a cordless telephone and a standard telephone
8. What is meant by transmission line encoding? Compare C-type and D-type line conditioning.
9. What is a paging system? Describe in detail with a neat block diagram how a paging system
work

UNIT-4
CELLULAR TELEPHONE SYSTEMS


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LECTURE NOTES

II B.Tech II semester (JNTUH-R15)

ELECTRONICS AND COMMUNICATION ENGINEERING

INSTITUTE OF AERONAUTICAL ENGINEERING

INTRODUCTION TO DATA COMMUNICATIONS AND NETWORKING

Figure 2.1 Basic concept of multiplexing

Multiplexing techniques can be categorized into the following three types:

Frequency-division multiplexing (FDM): It is most popular and is used extensively in

In frequency division multiplexing, the available bandwidth of a single physical medium

Figure 2.2 Basic concept of FDM

Data communication-(A40409) Page 46

Figure 2.4 Use of guard bands in FDM

Data communication-(A40409) Page 47

Figure 2.7 Time division multiplexing operation

Data communication-(A40409) Page 48

Data communication-(A40409) Page 49

2.9 Synchronous vs. Asynchronous TDM

Data communication-(A40409) Page 50

Data communication-(A40409) Page 51

Data communication-(A40409) Page 52

Data communication-(A40409) Page 62

The Subscriber Loop:

Functions of the telephone set:

Data communication-(A40409) Page 76

Data communication-(A40409) Page 78

Data communication-(A40409) Page 81

Data communication-(A40409) Page 83

D-type conditioned circuits must meet the following specifications:

Data communication-(A40409) Page 86

Data communication-(A40409) Page 87

Data communication-(A40409) Page 88

Data communication-(A40409) Page 89

Data communication-(A40409) Page 90

Data communication-(A40409) Page 91

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