UNIT-3

Part-1: 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.

Subscriber Loop:
A subscriber loop is that part of a telecommunication transmission system between a
subscriber's premises and the serving central office. Loops consist of twisted metallic cable
pairs, radio links, or optical fibres and serve as end-links in the communication channel
between users. The loop usually carries low-traffic volumes compared to interoffice facilities.

CO: Central Office; DLC: Digital Loop Carrier; LIU: Line Interface Unit; ONU: Optical
Network Unit; RF: Radio Frequency; RT: Remote Terminal; SIU: Subscriber Interface Unit
The twisted pair loop is used in the public network. It carries digital signals covering the range
from near dc to over 3 Mbps, with band widths extending upwards of 6 MHz. Digital loop
carrier (DLC) uses multiplexers with digital loops where, a number of individual analog and
digital channels can be combined into an aggregate data stream. In the places where it is not
possible to use twisted pair or fiber optic cables, like rural areas, digital or analog radio systems
are used.
Every subscriber in a telephone network is generally connected to a nearest switching
office called as Telephone Exchange by means of a dedicated pair of wires. These wires are
called as subscriber loop. The laying of lines to the subscriber premises from the exchange
office is called Cabling. It is difficult to run cables from each subscriber’s premises to the
exchange, large cables are used through which the drop wires (subscriber lines) are taken to a
distribution point. The drop wires are connected to wire pairs at the distribution point, in the
cables. Such distribution cables from nearby geographical area are connected at a same feeder
point where they connected to branch feeder cables which in turn, are connected to the main
feeder cable.

MDF: Main Distribution Frame, FP: Feeder Point, DP: Distribution Point, DW: Drop Wires,
DC: Distribution Cables, MF: Main Feeder, BF: Branch Feeder
The subscriber cable pairs from the exchange will also terminate at MDF through main feeder
cables that carry large number of wire pairs. These subscriber pairs and exchange pairs are
interconnected at the MDF using jumpers, which makes MDF to provide flexible mechanism
for reallocating cable pairs and subscriber numbers. This means a subscriber who shifts to a
different location though in the same exchange area, can be allowed to use the same number
using appropriate jumper, while his old drop wires can be used by another subscriber with a
new number.
Standard Telephone Set:
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 is called as Telephone. Sound
waves are acoustic waves and have no electrical component. The basic telephone set is a simple
analog transceiver designed with the primary purpose of converting speech or acoustical
signals to electrical signals.
The first telephone set that combined a transmitter and receiver into a single handheld unit was
introduced in 1878 and called the Butterstamp telephone.
In 1951, Western Electric Company introduced a telephone set that was the industry standard
for nearly four decades (the rotary dial telephone used by your grandparents). This telephone
set is called the Bell System 500-type telephone.
In modern-day telephone sets, the rotary dial mechanism is replaced with a Touch-Tone keypad.
The modern Touch-Tone telephone is called a 2500-type telephone set.

Functions of the Telephone Set

1. Notify the subscriber when there is an incoming call with an audible signal.
2. Provide a signal to the telephone network verifying when the incoming call has been
acknowledged and answered.
3. Convert speech (acoustical) energy to electrical energy in the transmitter and vice versa
in the receiver.
4. Incorporate some method of inputting and sending destination telephone numbers from
the telephone set to the central office switch over the local loop. This is accomplished
using either rotary diallers (pulses) or Touch-Tone pads (frequency tones).
5. Regulate the amplitude of the speech signal the calling person outputs onto the
telephone line.
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. 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).
Block Diagram of a Standard Telephone Set.
A standard telephone set is consisted of a transmitter, electrical network and a receiver for
equalization, connected circuitry to control side tone levels and for regulating signal power,
and required signalling circuitry. The necessary elements of a telephone set are the ringer
circuit, equalizer circuit, on/off hook circuit, hybrid circuit, microphone, speaker and a dialling
circuit.
Ringer Circuit: The reason of the ringer is to alert the destination party of incoming calls. The
audible tone by the ringer should be load adequate to be heard from a reasonable distance and
offensive sufficient to make a person need to answer the telephone immediately possible. In
modem telephones, the bell has been placed along with an electronic oscillator associated to
the speaker.
On/off hook circuit: The on/off hook circuit (sometimes termed as a switch hook) is nothing
more than an easy single-throw, double-pole (STDP) switch placed across the tip and ring. The
switch is mechanically linked to the telephone handset in order that when the telephone is idle
or on hook, the switch is open. While the telephone is in use or off hook, the switch is closed
completing and electrical path by the microphone in between the tip and ring of the local loop.
Equalizer circuit: Equalizers are combination of passive elements (capacitors, resistors etc.)
which are used to regulate the frequency and amplitude response of the voice signals.
Speaker: The speaker is the receiver, in essence for the telephone. The speaker changes
electrical signals received by the local loop to acoustical signals (as sound waves) which can
be heard and understood by human being. The speaker is connecting to the local loop by the
hybrid network. The speaker is classically enclosed in the handset of the telephone with the
microphone.
Microphone: For all practical reasons, the microphone is the transmitter for the telephone. The
microphone changes acoustical signals in the form of sound pressure waves by the caller to
electrical signals which are transmitted in telephone net-work by the hybrid network. The
microphone and the speaker both are transducers, as they convert one form of energy in other
form of energy. A microphone changes acoustical energy first to mechanical energy and after
that to electrical energy.
Hybrid network: The hybrid network (sometimes termed as a duplex coil or hybrid coil) in a
telephone set is a particular balanced transformer used to convert a two-wire circuit as the local
loop into a four-wire circuit as the telephone set and the vice-versa, therefore enabling full
duplex operation over a two-wire circuit. Fundamentally, the hybrid network separates the
transmitted signals by the received signals. Outgoing voice signals are classically in the 1-V to
2-V range, whereas incoming voice signals are normally half that value. The other function of
the hybrid network is to permit a small portion of the transmit signal to be returned to the
receiver into the form of a sidetone. In adequate sidetone causes the speaker to increase his
voice, making the telephone conversation appear unnatural. Excessively sidetone causes the
speaker to talk too softly, so reducing the volume which the listener receives.
Dialling Circuit: This circuit enables the subscriber to output signals showing digits, and it
enables the caller to enter the destination telephone number. This circuit could be a rotary
dialler that is nothing more than a switch connected with a mechanical rotating mechanism that
controls the number and interval of the on/off condition of a switch. Though, more than likely,
the dialling circuit is either a Touch-Tone keypad or an electronic dial-pulsing circuit that sends
various combinations of tones representing the termed as digits.
Basic Telephone Call Procedures:
Each subscriber is connected to the switch through a local loop. The switch is most likely some
sort of an electronic switching system (ESS machine). The local loops are terminated at the
calling and called stations in telephone sets and at the central office ends to switching machines.
When the calling party’s telephone set goes off hook, the switch hook in the telephone set is
released, completing a dc path between the tip and the ring of the loop through the microphone.
The ESS machine senses a dc current in the loop and recognizes this as an off-hook condition.
This procedure is referred to as loop start operation since the loop is completed through the
telephone set.
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 (Touch-Tone) signals.
Step 4: When the switching machine detects the first dialled 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.
Step 7b: 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 to 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.
 The people are the subscribers (i.e., the calling and the called party), and the machines
are the electronic switching systems in the telephone offices and the telephone sets
themselves.
 Signalling can be broadly divided into two major categories: station signalling and
interoffice signalling.
 Station signalling is the exchange of signalling messages over local loops between
stations (telephones) and telephone company switching machines.
 Interoffice signalling is the exchange of signalling messages between switching
machines.
 Signalling messages can be subdivided further into one of four categories:
 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.

Call Progress Tones:

 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 subscriber’s location to the central office switching machine. DTMF is a simple
two-of-eight encoding scheme where each digit IS represented by the linear addition of
two frequencies. DTMF is strictly for signalling between a Subscriber’s location and
the nearest telephone office or message switching centre. The following figure shows
the four-row-by-four column keypad matrix used with DTMF keypad.
1209 Hz 1336 Hz 1477 Hz 1633 Hz

2 3
697 Hz 1 A
ABC DEF

4 5 6
770 Hz B
GHI JKL MNO

7 8 9
852 Hz C
PRS TUV WXY

941 Hz * 0 # D
 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 signaling 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.
Digit or Frequencies Sum of
Control assigned (Hz) frequencies (Hz)

0 1300, 1500 2800

1 700, 900 1600
2 700, 1100 1800
3 900, 1100 2000
4 700, 1300 2000
5 900,1300 2200
6 1100, 1300 2400
7 700, 1500 2200
8 900, 1500 2400
9 1100, 1500 2600
Key Pulse (KP) 1100,1700 2800
Start (ST) 1500, 1700 3200
Idle 2600
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.

Call Progress Signals:

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 off-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 completed. 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 subscriber 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, intrusion tones, stutter dial
tones etc.
Cordless Telephones
 Cordless telephones are simply telephones that operate without cords attached to the
handset.
 Cordless telephones originated around 1980.
 They originally occupied a narrow band of frequencies near 1.7 MHz, used the 117 V
ac, 60-Hz household power line for an antenna.
 These early units used frequency modulation (FM) and susceptible to interference from
fluorescent lights and automobile ignition systems.
 In 1984, the FCC reallocated cordless telephone service to the 46-MHz to 49-MHz
band.
 In 1990, the FCC extended cordless telephone service to the 902-MHz to 928-MHz
band, FM and SST digital modulation are used in the 902-MHz to 928-MHz band.
 In 1998, the FCC expanded service again to the 2.4-GHz band.
 Adaptive differential pulse code modulation and spread spectrum technology (SST) are
used exclusively in the 2.4-GHz band

 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.
 The base station is an ac-powered stationary radio transceiver capable of transmitting
and receiving both supervisory and voice signals over the subscriber loop in the same
manner as a standard telephone.
 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.
 This technology uses twin band transmission: 2.4GHz and 902-928MHz.
 In 1984, the FCC allocated 10 full-duplex channels for 46-MHz to 49-MHz units. In
1995, the FCC added 15 additional full-duplex channels and extended the frequency
band to include frequencies in the 43-MHz to 44-MHz
Caller ID:
Caller ID enables the destination station 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 (no response is provided). The caller ID
information is transmitted and received using Bell System 202-compatible modems (ITU V.23
standard). This standard specifies a 1200-bps FSK (frequency shift keying) signal with a 1200-
Hz mark frequency (fc-fm) and a 2200-Hz space frequency (fc+fm). The FSK signal is
transmitted in a burst between the first and second 20-Hz, 90-Vrms ringing signals.

To ensure detection of the caller ID signal, the telephone must ring at least twice before being
answering. The caller ID signal does not begin until 500 ms after the end of the first ring and
must end 500 ms before the beginning of the second ring. Therefore, the caller ID signal has a
3-second window in which it must be transmitted.
Electronic Telephones

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 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 wide- area pagers.
To contact a person carrying a pager, the telephone number assigned to that pager has to be
dialled. 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.

Standard Simplex Paging System

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 party’s 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 frame 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
Part-2: CELLULAR TELEPHONE SYSTEMS
First- Generation Analog Cellular Telephone, Personal Communications system, Second-
Generation Cellular Telephone Systems, N-AMPS, Digital Cellular Telephone, Interim
Standard, Global system for Mobile Communications.
CELLULAR TELEPHONE SYSTEMS
 Mobile telephone services began in the 1940s and were called MTSs (mobile telephone
systems or manual telephone systems).
 MTS systems utilized frequency modulation and were generally assigned a single
carrier frequency in the range of 35-MHz to 45-MHz.
 The mobile unit used a push-to-talk (PTT) switch to activate the transceiver.
 Depressing the PTT button turned the transmitter on and the receiver off, whereas
releasing the PTT turned the receiver on and the transmitter off.
 When the PTT switch was depressed, the transmitter turned on and sent a carrier
frequency to the base station, illuminating a lamp on a switchboard.
 An operator answered the call by plugging a headset into a jack on the switchboard.
 The calling party verbally tells the operator the telephone number they wished to call.
 The operator connects the mobile unit to a trunk circuit of public telephone network of
destination with a patch cord.

• The conversation was limited to half-duplex operation, and only one conversation could
take place at a time due to single carrier frequency.
• Once the destination mobile unit answered, the operator disconnected from the
conversation, and the two mobile units will communicate directly with one another
through the airways.
• MTS numbers were generally five digits long and could not be accessed directly
through the public switched telephone network (PSTN).
• Two-way mobile radio systems operate half-duplex and use PTT transceivers.
 • Ex: Citizens Band, Walke-Talke
• In 1964, the Improved Mobile Telephone System (IMTS) was introduced, which used
several carrier frequencies and could handle several simultaneous mobile conversations
at the same time.
• Callers could reach an IMTS mobile phone by dialing the PSTN directly.
• Because of their high cost, limited availability, and narrow frequency allocation, early
mobile telephone systems were not widely used.
• The term mobile suggested any radio transmitter, receiver, or transceiver that could be
moved while in operation.
• The term portable described a relatively small radio unit that was handheld, battery
powered, and easily carried by a person.
• Definition of mobile telephone is any wireless telephone capable of operating while
moving at any speed, battery powered, and small enough to be easily carried by a
person.
• Cellular telephone is similar to two-way mobile radio in that most communications
occurs between base stations and mobile units.
• Base stations are fixed-position transceivers with relatively high-power transmitters
and sensitive receivers.
• Cellular telephone offers full-duplex transmissions and each cellular telephone is
assigned a unique telephone number.

CELLULAR TERMINOLOGY:
• Mobile Stations (MS): Mobile handsets, which is used by an user to communicate with
another user.
• Cell: Each cellular service area is divided into small regions called cell (5 to 20 Km).
The lesser is the cell radius, the more is the bandwidth.
 Base Stations (BS): Each cell contains an antenna, which is controlled by a small office.
Base stations are fixed-position transceivers with relatively high-power transmitters
and sensitive receivers
 Mobile Switching Center (MSC): Each base station is controlled by a switching office,
called mobile switching center.
 Frequency Reuse: Frequency reuse is the process in which the same set of frequencies
(channels) can be allocated to more than one cell, provided the cells are separated by
sufficient distance. The figure shows a geographic cellular radio coverage area
containing three groups of cells called clusters. Each cluster has seven cells in it, and
all cells are assigned the same number of full-duplex cellular telephone channels. Cells
with the same letter use the same set of channel frequencies. A, B, C, D, E, F and G
denote the seven sets of frequencies.
• Handoff: At any instant, each mobile station is logically in a cell and under the control
of the cell’s base station.
• When a mobile station moves out of a cell, the base station notices the MS’s signal
fading away and requests all the neighboring BSs to report the strength they are
receiving.
• The BS then transfers ownership to the cell getting the strongest signal and the MSC
changes the channel carrying the call. The process is called handoff. There are two types
of handoff; Hard Handoff and Soft Handoff.
• In a hard handoff, which was used in the early systems, a MS communicates with one
BS. As a MS moves from cell A to cell B, the communication between the MS and base
station of cell A is first broken before communication is started between the MS and
the base station of B. As a consequence, the transition is not smooth.
• For smooth transition from one cell (say A) to another (say B), an MS continues to talk
to both A and B. As the MS moves from cell A to cell B, at some point the
communication is broken with the old base station of cell A. This is known as soft
handoff (also called as make before break). A soft handoff may involve using
connections to more than two cells, e.g. connections to three, four or more cells can be
maintained by one phone at the same time. Softer handoffs are possible when the cells
involved in the handoff have a single cell site.
Example 1: Determine the number of channels per cluster and the total channel
capacity for a cellular telephone area comprised of 10 clusters with seven cells in each
cluster and 10 channels in each cell.
Sol: The total number of full-duplex channels is F = (10)(7) = 70 channels per cluster.
The total channel capacity is C = (10)(7)(10) = 700 channels.

Example 2: Determine
a. The channel capacity for a cellular telephone area comprised of seven macro cells
with 10 channels per cell.
b. Channel capacity if each macro cell is split into four minicells.
c. Channel capacity if each minicell is further split into four microcells.
Sol:
(a) The channel capacity = × = 70 𝑐ℎ𝑎𝑛𝑛𝑒𝑙𝑠/𝑎𝑟𝑒𝑎
(b) Each macro cell is split into 4 mini cells. So, total cells per area = 7x4 = 28 cells / area.
The channel capacity = × = 280 𝑐ℎ𝑎𝑛𝑛𝑒𝑙𝑠/𝑎𝑟𝑒𝑎
(c) Each mini cell is split into 4 micro cells. So, total cells per area = 7x4x4 = 112 cells /
area. The channel capacity = × = 1120 𝑐ℎ𝑎𝑛𝑛𝑒𝑙𝑠/𝑎𝑟𝑒𝑎

First Generation Analog Cellular Telephone:

• AMTS (Advanced Mobile Telephone System) was invented at Bell Lab in 1971 and
was used in 1983.
• It uses many low power cell site transceivers linked through a central computer-
controlled switching and control center.
• The original system used omnidirectional antennas to minimize initial equipment costs
and employed low-power (7-watt) transmitters in both base stations and mobile units.
• Voice-channel radio transceivers with AMPS cellular telephones use narrowband
frequency modulation (NBFM) with a usable audio-frequency band from 300 Hz to 3
kHz and a maximum frequency deviation of 12 kHz for 100% modulation.
• The AMPS system uses a seven-cell reuse pattern with provisions for cell splitting and
sectoring to increase channel capacity when needed.

AMPS Frequency Allocation:

• In 1980, the Federal Communications Commission decided to license two common
carriers per cellular service area.
• Two frequency allocation plans emerged—system A and system B—each with its own
group of channels that shared the allocated frequency spectrum.
• System A is defined for the non-wireline companies (i.e., cellular telephone companies)
and system B for existing wireline companies (i.e., local telephone companies).
• FCC initially assigned a 40-MHz frequency band consisting of 666 two-way channels
per service area with 30-kHz spacing between adjacent channels.
• The frequencies allocated to AMPS by the Federal Communications Commission
(FCC) range between 824 to 849 MHz in reverse channels (mobile to base) and 869 to
894 MHz in forward channels (base to mobile).
• Simultaneous transmission in both directions in a transmission mode is called full
duplex (FDX) or simply duplexing.
• Frequency-division duplexing (FDD) is used with AMPS and occurs when two distinct
frequency bands are provided to each user.
• A special device called duplexer is used in each mobile unit and base station to allow
simultaneous transmission and reception on duplex channels.
• In 1989, the Federal Communications Commission added an additional 10-MHz
frequency spectrum to the original 40-MHz band.
• The additional frequencies are called the expanded spectrum and include channels 667
to 799 and 991 to 1023.
• The mobile unit’s transmit carrier frequency in MHz for any channel is calculated as
follows:

where ft = transmit carrier frequency (MHz); N = channel number

• The mobile unit’s receive carrier frequency is obtained by simply adding 45 MHz to
the transmit frequency:
 AMPS Identification Codes:
• The AMPS system uses several identification codes for each mobile unit.
• Mobile Identification Number (MIN): It is a 34-bit binary code, which in the United
States represents the standard 10-digit telephone number. The MIN is comprised of a
three-digit area code, a three-digit prefix (exchange number), and a four-digit subscriber
(extension) number. The exchange number is assigned to the cellular operating
company. If a subscriber changes service from one cellular company to another, the
subscriber must be assigned a new cellular telephone number.
• Electronic Serial Number (ESN): It is a manufactured characteristic of the mobile
unit. This identifier is permanent and associated with the physical equipment. It is 32
bits in length, with the first 8 bits identifying the manufacturer.
• Station Class Mark (SCM): It is a four-bit wide, which indicates whether the terminal
has access to all 832 channels or only for 666 channels.
• System Identifier (SID): It is a 15-bit binary code issued by the FCC to an operating
company when it issues a license to provide AMPS cellular service to an area. The SID
is stored in all base stations and all mobile units to identify the operating company.
Every mobile unit knows the SID of the system it is subscribed to, which is the mobile
unit’s home system. Whenever a mobile unit initializes, it compares its SID to the SID
broadcast by the local base station. If the SIDs are the same, the mobile unit is
communicating with its home system. If the SIDs are different, the mobile unit is
roaming.
• Local operating companies assign a two-bit digital color code (DCC) and a supervisory
audio tone (SAT) to each of their base stations to help the mobile units distinguish one
base station from a neighboring base station.
 AMPS Control Channels:
 The AMPS channel spectrums are divided into two basic sets or groups: voice or user
channels and control channels.
 Voice or User Channels: Used for propagating actual voice conversations or subscriber
data.
 Control Channels: Dedicated for exchanging control information between mobile
units and base stations. They are used exclusively to carry service information.
 Control channels are used in cellular telephone systems to enable mobile units to
communicate with the cellular network through base stations without interfering with
normal voice traffic occurring on the normal voice or user channels.
 Control channels are used for call origination, for call termination, and to obtain system
information.
 Control channels are digital and uses the digital modulation technique called as
Frequency Shift Keying (FSK).
 In AMPS, base stations broadcast on the forward control channel (FCC) and listen on
the reverse control channel (RCC). The control channels are sometimes called setup or
paging channels.
 Forward Control Channel

o
 Reverse Control Channel

o
 Both the channels transmit the data at a rate of 10Kbps.
 The control channel message is preceded by a 10-bit dotting scheme, which is a
sequence of alternating 1s and 0s.
 The dotting scheme is followed by an 11-bit synchronization word with a unique
sequence of 1s and 0s that enables a receiver to instantly acquire synchronization.
 The sync word is immediately followed by the message repeated five times. The
redundancy helps compensate for the ill effects of fading. If three of the five words are
identical, the receiver assumes that as the message.
 Forward control channel data formats consist of three discrete information streams:
stream A, stream B, and the busy-idle stream.
 Messages to the mobile unit with the least-significant bit of their 32-bit mobile
identification number (MIN) equal to 0 are transmitted on stream A, and MINs with the
least significant bit equal to 1 are transmitted on stream B.
  The busy-idle data stream contains busy-idle bits, which are used to indicate the current
status of the reverse control channel (0 busy and 1 idle).
 The types of messages transmitted over the FCC are the mobile station control message
and the overhead message train.
 RCC messages consists of a 30-bit dotting sequence, an 11-bit synchronization word,
and the coded digital color code (DCC), which is added so that the control channel is
not confused with a control channel from a nonadjacent cell that is reusing the same
frequency.

Second Generation Cellular Telephone System:

The drawbacks of 1st generation SCTS are:
1. They are designed for limited customers
2. The battery performance is poor.
3. Channel bandwidth is also less.
4. Size of the device is large.
5. Not cost effective.
6. Weak signal strength.
7. False handoffs.
 To overcome the above drawbacks, digital technologies are to be employed in the SCTS.
 Due to the uncertainty about practical implementation and cost effectiveness of digital
technologies, Motorola developed 2nd generation CTS called as Narrow-band AMPS (N-
AMPS) to increase channel capacity of standard AMPS.
Narrow-band AMPS (N-AMPS):
 N-AMPS uses Interim Standards like IS-54, IS-136, IS—95.
 During this phase, Global System for Mobile Communication (GSM) was emerged in
Europe.
 N-AMPS had provided short term solution to traffic congestion problem.
 It reduced the frequency deviation and divided the 30KHz band into three parts, so that,
it allows 3 mobile units to use 30 KHz channel at a time i.e., a 10KHz channel is allocated
to each user and this channel can handle its own calls.
 Here one NAMPS uses the original carrier frequency and the remaining two NAMPS
uses the same carrier frequency with an offset of ±10 KHz. For example, if one NAMPS
uses 825MHz carrier frequency, then the second NAMPS uses 825MHz+10KHz =
825.01MHz and the third NAMPS uses 825MHz-10KHz = 824.99MHz as the carrier
frequencies or vice versa.
 NAMPS can be operated in dual mode: 30KHz carrier mode and 10KHz carrier mode.
So, it can handle 4 different handoffs.
 They are wide to wide handoff (30KHz to 30KHz), wide to narrow handoff (30KHz to
10KHz), narrow to wide handoff (10KHz to 30KHz) and narrow to narrow handoff
(10KHz to 10KHz).
Drawbacks of 2nd Generation CTS:
Reducing the channel bandwidth to increase the channel capacity leads to the following
disadvantages:
1. Reduced speech quality.
2. Reduced Signal to Interference Ratio.
3. Increased frequency reuse factor.
These drawbacks are compensated by using Interference Avoiding Scheme called as Mobile
Reported Interference (MRI).

Personal Communication System (PCS):

 To overcome the drawbacks of 2nd generation AMPS, PCS was introduced.
 PCS is a combination of AMPS and intelligent network. So, it is termed as 3rd
generation Cellular Telephone System.
 FCC defined PCS as “A family of mobile or portable radio communication services
which provide services to individuals and business and is integrated with a variety of
competing networks”.
 PCS provides data services along with voice-based services. So, it had extended its
frequency band to 1850MHz to 2000MHz.
 It uses protocols such as IS-54 and IS-95.
 It assigns a personal telephone number (PTN) to each subscriber and it is stored in the
database of Signalling System (SS7) network.
 The database in SS7 tracks the mobile unit.
 When a call is placed, the AI NETWORK in SS7 determines where the call is to be
directed and how it should be directed.
 Each PCS maintains 3 databases: HLR, VLR and EIR.
 HLR (Home Location Register): It stores home subscription information, and
provides additional services such as call waiting, call hold and call conference. Each
HLR is allocated to only one personal mobile.
 VLR (Visitor Location Register): It stores the information about subscribers in
particular MTSO area. One VLR is allocated per mobile switch. It stores permanent
data, such as that found in the HLR, plus temporary data, such as the subscriber’s
current location.
 EIR (Equipment Identification Register): It stores the equipment ID and the type of
equipment present in it. It is used to identify the stolen or fraudulent mobile units.
 PCS offers the following voice-based services in addition to standard CTS.
 Available Mode: It allows all calls except the numbers that are blocked. Only a few
numbers can be blocked. Initially, it was checked in the database whether it is blocked
or not. The subscribers can update the database.
 Screen Mode: In this mode, the name of the calling party appears on the mobile display.
The unanswered calls are automatically forwarded to a forwarding destination specified
by the subscriber.
  Private Mode: In this mode, all the calls will be automatically forwarded to the
forwarding destination without ringing except the calls specified by the subscriber. The
subscriber can modify the list of numbers.
 Unavailable Mode: In this mode, no calls are allowed to pass to the subscriber. All the
incoming calls are automatically forwarded to a forwarding destination.

Advantages of PCS:
1. These are very small and portable.
2. These use digital technology. Hence, it requires only less power.
3. They use smaller cells and can accommodate more base stations in a service area.
Disadvantages of PCS:
1. Cost of the network is more: PCS uses more base stations due to smaller size and
hence more transceivers, antennas and trunk circuits are needed.
2. Antenna placement is difficult.

DIGITAL CELLULAR TELEPHONE:

 Day-by-day, the number of subscribers using cellular phones has been increasing
rapidly. On the other hand, the frequency spectrum allocated for cellular
communication system remained unchanged.
 Even though, band splitting, cell splitting and sectoring concepts are used in N-AMPS
to increase the user capacity, AMPS is not able to handle high traffic density.
 To overcome these disadvantages, digital cellular telephone systems have been
introduced.
 United States Digital Cellular System (USDCS) was designed and developed to support
high user density within a fixed bandwidth.
 This system supports IS-54 protocol with dual mode operation.
 It uses the same carrier frequencies, frequency reuse plan, and base stations.
 USDC system maintains compatibility with AMPS systems in several ways, it is also
known as Digital AMPS (D-AMPS or sometimes DAMPS).
 The USDC cellular telephone system has an additional frequency band in the 1.9- GHz
range that is not compatible with the AMPS frequency allocation.
The total usable spectrum is subdivided into sub bands (A through F); however, the individual
channel bandwidth is limited to 30 kHz (the same as AMPS).
Time-Division Multiple Accessing (TDMA):
 USDC uses time-division multiple accessing (TDMA) as well as frequency-division
multiple accessing (FDMA).
 It divides the total available radio-frequency spectrum into individual 30-kHz cellular
channels (i.e., FDMA).
 TDMA allows more than one mobile unit to use a channel at the same time by further
dividing transmissions within each cellular channel into time slots.
 A USDC TDMA transmission frame consists of six equal-duration time slots enabling
each 30-kHz AMPS channel to support three full-rate or six half-rate users. Hence, it
offers as much as six times the channel capacity as AMPS.
 The original USDC standard also utilizes the same 50-MHz frequency spectrum and
frequency-division duplexing scheme as AMPS.
 The advantages of digital TDMA multiple-accessing systems over analog AMPS
FDMA systems are given below:
1. Time domain multiple accessing allows for a threefold to sixfold increase in the number
of mobile subscribers using a single cellular channel.
2. Digital signals are much easier to process than analog signals as most of the modern
modulation techniques are developed to be used in a digital environment.
3. Digital signals (bits) can be easily encrypted and decrypted, safeguarding against
eavesdropping.
4. The entire telephone system is compatible with other digital formats, such as those used
in computers and computer networks.
5. Digital systems inherently provide a quieter (less noisy) environment than their analog
counterparts.

INTERIM STANDARD(IS):
IS-54:
 In 1990, the Electronics Industries Association and Telecommunications Industry
Association (EIA/TIA) standardized the dual-mode USDC/AMPS system as Interim
Standard 54 (IS-54), Cellular Dual Mode Subscriber Equipment.
 Using IS-54, a cellular telephone carrier could convert any or all of its existing analog
channels to digital.
 To achieve dual-mode operation, IS-54 provides digital control channels and both
analog and digital voice channels.
 Dual mode mobile units can operate in either the digital or the analog mode for voice
and access the system with the standard AMPS digital control channel.
 IS-54 specifies a 48.6 kbps rate per 30-kHz voice channel divided among three
simultaneous users.
 Each user is allocated 13 kbps, and the remaining 9.6 kbps is used for timing and control
overhead.
USDC Control Channels and IS-136.2
 The IS-54 USDC standard specifies 42 primary control channels as AMPS and 42
additional control channels called secondary control channels.
 So, USDC offers twice as many control channels as AMPS and is therefore capable of
providing twice the capacity of control traffic within a given market area.
 To maintain compatibility with existing AMPS cellular telephone systems, the primary
forward and reverse control channels in USDC cellular systems use the same signalling
techniques and modulation scheme (FSK) as AMPS.
 A new standard IS-136.2 replaces FSK with π/4 DQPSK modulation for the 42
dedicated USDC secondary control channels, allowing digital mobile units to operate
entirely in the digital domain.
 The IS-136.2 standard is called North American- TDMA. IS 136 was developed to
provide a host of new features and services.
 An additional “sleep mode” which conserves power is also provided.
 The IS-54 standard specifies three types of channels: analog control channels, analog
voice channels, and a 10-kbps binary FSK digital control channel (DCCH).
 The IS-136 standard provides the above three channels and an additional one: a digital
control channel with a signalling rate of 48.6 kbps on USDC-only control channels.
 The new digital control channel includes several logical channels with different
functions, including the random-access channel (RACH), the SMS point-to-point,
paging, and access response channel (SPACH); the broadcast control channel (BCCH)
and the shared channel feedback (SCF) channel.
 RACH: It is a unidirectional channel used by mobile units to request access to the
cellular telephone system.
 SPACH: It is used to transmit information from base to specific mobile station and
information transmitted on SPACH channel include 3 separate logical subchannels:
SMS point-to-point messages, paging messages, and access response messages.
 BCCH: It is an acronym referring to the F-BCCH, E-BCCH and S-BCCH. The fast
broadcast channel (F-BCCH) broadcasts digital control channel (DCCH) structure
parameters. Mobile units use F-BCCH information when initially accessing the system
to determine the beginning and ending of each logical channel in the DCCH frame. The
extended broadcast control channel (E-BCCH) carries information about neighbouring
analog and TDMA cells and optional messages, such as emergency information, time
and date messaging etc. The SMS broadcast channel (S-BCCH) is a logical channel
used for sending short messages to individual mobile units.
 SCF: It is used to support random access channel operation by providing information
about which time slots the mobile unit can use for access attempts and also if a mobile
unit’s previous RACH transmission was successfully received.
USDC Digital Voice Channel:
Like AMPS, each USDC voice channel is assigned a 30-kHz bandwidth on both the forward
and the reverse link. With USDC, each channel can support as many as three full-rate mobile
users simultaneously by using digital modulation and a TDMA format called North American
Digital Cellular (NADC). Each radio-frequency voice channel in the total AMPS FDMA
frequency band consists of one 40-ms TDMA frame comprised of six time slots containing 324
bits each. The average cost per subscriber per base station equipment is lower with TDMA
since each base station transceiver can be shared by upto six users at a time.

(a) D-AMPS channel with 3 users (b) D-AMPS channel with 6 users
E-TDMA:
General Motors Corporation implemented a TDMA scheme called E-TDMA {Extended or
Enhanced TDMA}, which incorporates six half-rate users transmitting at half the bit rate of
standard USDC TDMA systems. E-TDMA systems also incorporate digital speech
interpolation (DSI) to dynamically assign more than one user to a time slot, deleting silence on
the calls.
Consequently E-TDMA can handle approximately 12 times the user traffic as standard AMPS
systems and four times that of systems complying with IS-54.
Each time slot in every USDC voice-channel frame contains four data channels-three for
control and one for digitized voice and user data.
The full-duplex digital traffic channel (DTC) carries digitized voice information and consists
of a reverse digital traffic channel (RDTC) and a forward digital traffic channel (FDTC) that
carry digitized speech information or user data.
The three supervisory channels are given below:
 Coded digital verification colour code (CDVCC): Its purpose is to provide co-channel
identification similar to the SAT signal transmitted in the AMPS system. It is a 12-bit
message transmitted in every time slot.
 Slow associated control channel (SACCH): It is a signalling channel for transmission
of control and supervision messages between the digital mobile unit and the base station
while the mobile unit is involved with a call. It is also used by the mobile unit to report
signal strength measurements of neighbouring base stations, so when needed the base
station can initiate a mobile-assisted handoff (MAHO).
 Fast associated control channel (FACCH): It is a second signalling channel for
transmission of control and specialized supervision and traffic messages between the
base station and the mobile units. It is a blank-and-burst type of transmission than when
transmitted replaces digitized speech information with control and supervision
messages within a subscriber’s time slot.
USDC Digital modulation scheme
 To achieve a transmission bit rate of 48.6 kbps in a 30-kHz AMPS voice channel, a
bandwidth (spectral) efficiency of 1.62 bps/Hz is required, binary FSK is incapable.
  USDC voice and control channels use a symmetrical differential, phase-shift keying
technique known as π/4 DQPSK or π/4 differential quadrature phase shift keying, which
offers several advantages such as improved co-channel rejection and bandwidth
efficiency.
 In π/4 DQPSK modulator, data bits are split into two parallel channels that produce a
specific phase shift in the analog carrier, and since there are four possible bit pairs, there
are four possible phase shifts using a quadrature I/Q modulator and the four phase
changes are π/4, - π/4, 3 π/4 and -3 π/4, which define eight possible carrier phases.
 Using pulse shaping with π/4 DQPSK allows for the simultaneous transmission of three
separate 48.6-kbps speech signals in a 30- kHz bandwidth.

Global System for Mobile Communications:

 Throughout the evolution of cellular telecommunications, various systems have been
developed without the benefit of standardized specifications. This presented many
problems directly related to compatibility.
 GSM standard is intended to address these problems. GSM was the world’s first totally
digital cellular telephone system designed to use the services of SS7 signalling and an
all-digital data network called integrated services digital network (ISDN) to provide a
wide range of network services.
 GSM is now the world’s most popular standard for new cellular telephone and personal
communications equipment.
Advantages of GSM
 Communication: mobile, wireless communication, support for voice and data services
 Total mobility: international access, chip-card enables use of access points of different
 providers.
 Worldwide connectivity: one number, the network handles every location.
 High capacity: better frequency efficiency, smaller cells, more customers per cell.
 High transmission quality: high audio quality and reliability for wireless, uninterrupted
phone calls at higher speeds (e.g., from cars, trains).
GSM Services:
 GSM telephone services are broadly classified into three categories: bearer services,
teleservices, and supplementary services.
 Teleservices are mainly voice services that provide subscribers with the complete
capability to communicate with other subscribers.
 Data services provide the capacity necessary to transmit appropriate data signals
between two access points creating an interface to the network. Some of the subscriber
services are given below:
 Dual tone Multifrequency (DTMF): DTMF is a tone signalling scheme used for
various control purposes via the telephone network, such as remote control of an
answering machine.
 Facsimile Group III: GSM supports CCITT group 3 facsimile. This enables a GSM
connected fax to communicate with any analog fax in the network.
 Short Message Service: A message consisting of 160 alphanumeric characters can
be sent to or from a mobile station. If the mobile station is off or not in the coverage
 area, the message is stored and then offered back ensuring that the message will be
received.
 Cell Broadcast: a message of maximum of 93 characters can be broadcast to all
mobile subscribers in a given geographic area. Typical applications include traffic
congestion warnings and reports on accidents.
 Voice mail: This service is actually an answering machine within the network
controlled by the subscriber. Calls can be forwarded to the subscriber’s voice mail
box, which can be checked later by the subscriber via a personal code.
 Fax mail: with this service, the subscriber can receive fax at any fax machine.

 GSM supports a set of supplementary services that can complement and support both
telephony and data services. These are defined by GSM and are termed as revenue
generating services. Some of them are listed below:
 Call forwarding: It gives the subscriber the ability to forward incoming calls to
another number if the called unit is not reachable, not answering, or busy.
 Barring of outgoing calls: this service makes it possible for a subscriber to prevent
all outgoing calls.
 Barring of incoming calls: It allows the subscriber to prevent incoming calls either
completely or if in roaming.
 Advise of Charge: The AoC service provides the mobile subscriber with an estimate
of the call charges.
 Call hold: This service enables the subscriber to interrupt an ongoing call and then
subsequently re-establish the call.
 Call waiting: It allows the mobile subscriber to be notified of an incoming call
during a conversation. The subscriber then can answer, reject or ignore the incoming
call.
GSM architecture:
The GSM network is divided into three major sub systems: the Network Switching Subsystem
(NSS), the Base Station Subsystem (BSS) and the Operation and Support System (OSS). The
basic GSM elements are shown below:
Network Switching Subsystem: The NSS is responsible for performing call processing and
subscriber related functions. The switching system includes the following functional units:
 Home Location Register (HLR): It is a database used for storage and management of
subscriptions. HLR stores permanent data about subscribers, including a subscriber’s
service profile, location information and activity status. When an individual buys a
subscription from the PCS provider, he or she is registered in the HLR of that operator.
 Visitor Location Register (VLR): It is a database that contains temporary information
about subscribers that is needed by the MSC in order to service visiting subscribers.
VLR is always integrated with the MSC. When a MS roams into a new MSC area, the
VLR connected to that MSC will request data about the mobile station from the HLR.
Later if the mobile station needs to make a call, VLR will be having all the information
needed for call setup.
 Authentication Centre (AUC): A unit called the AUC provides authentication and
encryption parameters that verify the user’s identity and ensure the confidentiality of
each call.
 Equipment Identity Register (EIR): It is a database that contains information about the
identity of mobile equipment that prevents calls from stolen, unauthorized or defective
mobile stations.
 Mobile Switching Centre (MSC): The MSC performs the telephony switching functions
of the system. It controls calls to and from other telephone and data systems.
Base Station Subsystem (BSS): All radio related functions are performed in the BSS, which
is also known as radio subsystem. It provides and manages radio-frequency transmission paths
between mobile units and MSC. It consists of many base station controllers (BSC) and base
transceiver stations (BTS).
 Base station controllers (BSC): The BSC provides all the control functions and physical
links between the MSC and BTS. It is a high-capacity switch that provides functions
such as handover, cell configuration data, and control of radio frequency (RF) power
levels in BTS. A number of BSC’s are served by and MSC.
 Base transceiver station (BTS): The BTS handles the radio interface to the mobile
station. The BTS is the radio equipment (transceivers and antennas) needed to service
each cell in the network. A group of BTS’s are controlled by an BSC.
Operation and Support System:
The operations and maintenance centre (OMC) is connected to all equipment in the
switching system and to the BSC. Implementation of OMC is called operation and support
system (OSS). The OSS is the functional entity from which the network operator monitors and
controls the system. The purpose of OSS is to offer the customer cost-effective support for
centralized, regional and local operational and maintenance activities that are required for a
GSM network. OSS provides a network overview and allows engineers to monitor, diagnose
and troubleshoot every aspect of the GSM network.

The mobile station (MS) consists of the mobile equipment (the terminal) and a smart
card called the Subscriber Identity Module (SIM). The SIM provides personal mobility, so that
the user can have access to subscribed services irrespective of a specific terminal. By inserting
the SIM card into another GSM terminal, the user is able to receive calls at that terminal, make
calls from that terminal, and receive other subscribed services.
 The mobile equipment is uniquely identified by the International Mobile Equipment
Identity (IMEI). The SIM card contains the International Mobile Subscriber Identity (IMSI)
used to identify the subscriber to the system, a secret key for authentication, and other
information. The IMEI and the IMSI are independent, thereby allowing personal mobility. The
SIM card may be protected against unauthorized use by a password or personal identity
number.

GSM Radio Subsystem:

GSM uses two 25-MHz frequency bands that have been set aside for system use in all
member companies. The 890 MHz to 915 MHz band is used for mobile unit-to-base station
transmissions (reverse link transmissions), and the 935-MHz to 960-MHz frequency band is
used for bas station-to-mobile unit transmission (forward link transmission). GSM uses
frequency-division duplexing and a combination of TDMA and FDMA techniques to provide
base stations simultaneous access to multiple mobile units. The available forward and reverse
frequency bands are subdivided into 200-kHz wide voice channels called absolute radio-
frequency channel numbers (ARFCN). The ARFCN number designates a forward reverse
channel pair with 45-MHz separation between them. Each voice channel is shared among as
many as eight mobile units using TDMA.
Each of the ARFCN channel subscribers occupies a unique time slot within the TDMA
frame. Radio transmissions in both directions is at a 270.833-kbps rate using binary Gaussian
minimum shift keying (GMSK) modulation with an effective channel transmission rate of
33.833 kbps per user.