KURUKSHETRA

INDUSTRIAL TRAINING REPORT

Under the guidance of:

Submitted By:
Mr. Rajesh Kumar

Yogesh Verma

S.D.E. (OCB-283)

111360

Telephone Exchange

7th SEM, ECE Dept.

Thanaser, Kurukshetra

NIT Kurukshetra
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ACKNOWLEDGEMENT
I acknowledge my gratitude and thanks to all the well-knowledge person
for giving me opportunity to provide all the best facilities available at
this telecom center. Success of every project depends largely on the self
& encouragement and guidance of many others. I take this opportunity
to express my gratitude to the people who has been involved in the
successful completion of this study project.
First of all, I would like to thank the management at BSNL for
giving me opportunity do my FOUR weeks project training in their
organization. I was with valuable advice and endless supply of new ideas
and support for this project.
I would like to thank Mr. Rajesh Kumar, SDE (OCB) for
providing practical exposure for the project and his valuable guidance
during the project work.
-

Yogesh Verma

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CONTENTS:

Abstract..5
1. History of Telephone Exchange.6
2. Introduction to OCB-2837
2.1 Salient Features of OCB-2837
2.2 Typical Telephone exchange using OCB-283..9
3. Signaling in Telecommunications.....10
3.1 Common Associated Signaling...10
3.2 Common Channel Signaling...10
4. PCM Principles.12
4.1 Introduction.12
4.2 Multiplexing Techniques.12
4.3 Pulse Code Modulation14
5. Optical-Fibre Communications.20
5.1 Brief History20
5.2 Fibre-Optics Applications21
5.2 Fibre-optics System..21
5.3 Principle of Operation..21
5.4 Fibre Types..23
5.5 OFC Splicing...24

6. Digital Hierarchies.27
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6.1 Overview of PDH.....27

6.2 SDH/ SONET.....29
6.3 SDH Rates...31
7. MOBILE COMMUNICATION- GSM, CDMA......32
7.1 Introduction.32
7.2 Subsystems and Network Elements in GSM...32
7.3 Operation & Maintenance Centre35
7.4 Evolution from GSM to 3G.38
7.5 Introduction to CDMA 2000...38
7.6 Overview of WCMA...39
7.7 Spread Spectrum Principle..41

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ABSTRACT:

This report takes a pedagogical stance in demonstrating how results from theoretical
communication may be applied to yield significant insight into the behavior of the
telecommunication systems, and that this is immediately attainable with the present state of the
art. The focus for this detailed study is provided by various COMMUNICATION systems like
Mobile Communication currently being deployed throughout the world. Accuracy and system
reliability concerns dominate in this domain. Thus, the training has been carried out with the idea
of understanding the telecommunication systems.

A GLANCE AT
BHARAT SANSHAR NIGAM LIMITTED
(A Govt. of India Enterprise)

1) HISTORY OF TELEPHONE EXCHANGE

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The Telephone was invented by Mr. Graham Bell. During early stage of development of
telephone exchange, the connections are established with help of human operator. Those type of
exchange were called Manual Telephone Exchange. The technology of telephony was going on
progress with the introduction of automatic exchange. Manual telephone is replaced & automatic
exchange became in use there were lot of advantage of automatic exchange over manual.
In manual telephony, the type of exchange used is Central Battery (C.B.). In certain case
local battery exchange (L.B.) is also used. The local battery exchange is also called magnet
exchange because the set has a magneto generator which the subscriber is required to rotate, to
generate the A.C. necessary to operate the indicator at the exchange.
In automatic telephony connections between two subscribers are established with the help
of human operator. Obviously the junction of human operator is carried out by the machine
known as switching or selector stages. After the development of automatic telephone exchange
technology as a subscriber directly & it has many advantages over manual telephone exchange.
Now a day electronic Automatic exchange is widely used due to their advantages.

1876

Invention of Telephone

1915

First transcontinental telephone(NY-SF)

1920

First automatic switches

1956

TAT-1 transatlantic cables(35 lines)

1962

Digital transmission(T1)

1965

1ESS analog switches

1974

Internet packet voice

1977

4ESS digital switches

1980s

Signaling System

1990s

Advanced Intelligent Network (AIN)

(out-of-band)

2) INTRODUCTION TO OCB-283
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OCB-283 is digital switching system which supports a variety of communication needs

like basic telephony, ISDN, interface to mobile communication, data communication etc. This
system has been developed by CIT ALCATEL of France and therefore has many similarities to
its predecessors E-10B also known as OCB-181 in France.
The first OCB-283 exchange if R-11 version was commissioned in Brest, France and
Beijing, china in 1911, the first OCB-283 exchange came to India in 1993 subsequently the
system has been upgraded and current version R-20 was fully validated in January 1994. The
exchanges, which are being supplied to India, belong to R-20 version. Thereafter time up
gradation to this OCB-283 system was upgraded to R-25 version.

2.1 SALIENT FEATURE OF OCB-283:

1. It is a digital switching system with an angle T stage switch a maximum of 2048 PCM can
be connected.
2. It supports both analog & digital subscribers.
3. The system supports all the existing signaling system like decadic, M1 (R2), CAS & also
CCITT#7 signaling system.
4. The system has automatic recovery feature. When a serious fault occurs in control unit, it
gives the message to SMM. The SMM pulls this unit out of service, loads software of this unit in
a backup unit & brings it in to service diagnostic programs are running on the faulty unit & a
diagnosis is printed on terminal.
5. OCB-283 has a double remoting facility. Subscriber access unit CSNL can be placed at a
remote place & connected to main exchange through PCM links. Further line connectors can also
be placed at remote location & connected to CSNL or CSND through PCM this special feature
can meet entire range of necessity viz urban, semi-urban & rural.
6. Various units of OCB-283 system are connected over Token ring. This enables fast exchange
of information & avoid complicated links & wiring between various units.
7. The charge account of subscriber are automatically saved in a disk once in a day. This avoid
loss of revenue in case of total power supply failure.
8. The exchange can be managed either locally or from an NMC through 64 KBps links.
9. All the control units are implemented on same type of hardware. This is called a station.
Depending on the requirement of processing capacity, software of either one or several control
units can be located on the same station. For all these control units only one backup station is
provided enabling Automatic Recovery in case of fault.
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10. The OCB-283 system is made up of only 35 type of cards. This excludes the cards required
for CSN. Because of this number of space card to be kept for maintenance are drastically
reduced.
11. The system has modular structure. The expansion can be very easily carried out by adding
necessary hardware or software.
12. The SMMs (O&M Units) are duplicated with one active & other hot standby. In case of
faults, switch over takes place automatically. More over as disk are connected SMMs, there is no
necessity of changing cables from one system to another.
13. The hard disk of memory capacity 9.2 GB is very compact & maintenance free. The detail
billing data regularly saved in the disk itself from where they can be transferred to magnetic
tapes for processing.

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2.2 A Typical Telephone Exchange Using OCB-283:

LR

CSNNNNL

CSNND
CSED

STS
1x3
SMX

LR

SMT
( 1 TO 28) X 2

LR

SMA
( 2 TO 37)

1 TO 4 MAS
SMC
2 TO 14

1 MIS
SMM
1x2
TMN

CSN

Digital satellite center

SMC :

Main Control Station

SMA :

Auxiliary Equipment Control Station

SMT :

Truck Control Station

SMX :

Matrix Control Station

SMM :

Maintenance Station

STS

Synchronization and Time Base Station

3)

SIGNALLING IN TELECOMMUNICATIONS

The term signaling, when used in telephony, refers to the exchange of control information
associated with the establishment of a telephone call on a telecommunications circuit. An
example of this control information is the digits dialed by the caller, the caller's billing number,
and other call-related information.
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When the signaling is performed on the same circuit that will ultimately carry the
conversation of the call, it is termed Channel Associated Signaling (CAS).
In contrast, SS7 signaling is termed as Common Channel Signaling (CCS) in that the path
and facility used by the signaling is separate and distinct from the telecommunications channels
that will ultimately carry the telephone conversation. With CCS, it becomes possible to exchange
signaling without first seizing a facility, leading to significant savings and performance increases
in both signaling and facility usage.

3.1 CHANNEL ASSOCIATED SIGNALING:

Channel Associated Signaling (CAS), also known as per-trunk signaling (PTS), is a form of
digital communication signaling. As with most telecommunication signaling methods, it uses
routing information to direct the payload of voice or data to its destination. With CAS signaling,
this routing information is encoded and transmitted in the same channel as the payload itself.
This information can be transmitted in the same band or a separate band to the payload.
CAS potentially results in lower available bandwidth for the payload. For example, in the PSTN
the use of out-of-band signaling within a fixed bandwidth reduces a 64 Kbit/s DS0 to 56 Kbit/s.
Because of this, and the inherent security benefits of separating the control lines from the
payload, most current telephone systems rely more on Common Channel Signaling (CCS).

3.2 COMMON CHANNEL SIGNALING

In telephony, Common Channel Signaling (CCS) is the transmission of signaling information
(control information) on a separate channel from the data, and, more specifically, where that
signaling channel controls multiple data channels.
For example, in the public switched telephone network (PSTN) one channel of a
communications link is typically used for the sole purpose of carrying signaling for
establishment and Tear down of telephone calls. The remaining channels are used entirely for the
transmission of voice data. In most cases, a single 64kbit/s channel is sufficient to handle the call
setup and call clear-down traffic for numerous voice and data channels.

CCS offers the following advantages over CAS, in the context of the PSTN:

Faster call setup.

No falsing interference between signaling tones by network and speech frequencies.

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Greater trunking efficiency due to the quicker set up and clear down, thereby reducing
traffic on the network.

No security issues related to the use of in-band signaling with CAS.

CCS allows the transfer of additional information along with the signaling traffic
providing features such as caller ID.

The most common CCS signaling methods in use today are Integrated Services Digital Network
(ISDN) and Signaling System 7 (SS7). ISDN signaling is used primarily on trunks connecting
end-user private branch exchange (PBX) systems to a central office.
Access to an ISDN is provided in two forms:

Basic-Rate Access (BRA)

The customers line carries two 64 Kbit/s B channels plus a 16 Kbit/s D channel (a
common signaling channel) in each direction.

Primary Rate Access (PRA)

The line carries a complete PCM frame at 2 Mbit/s in each direction. This gives the
customer 30 circuits at 64 Kbit/s plus a common signaling channel, also at 64 Kbit/s.

4) PCM PRINCIPLES:
4.1 INTRODUCTION:
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A long distance or local telephone conversation between two persons could be provided
by using a pair of open wire lines or underground cable as early as early as mid of 19th
century. However, due to fast industrial development and increased telephone awareness,
demand for trunk and local traffic went on increasing at a rapid rate. To cater to the increased
demand of traffic between two stations or between two subscribers at the same station we
resorted to the use of an increased number of pairs on either the open wire alignment, or in
underground cable. This could solve the problem for some time only as there is a limit to the
number of open wire pairs that can be installed on one alignment due to headway
consideration and maintenance problems. Similarly increasing the number of open wire pairs
that can be installed on one alignment due to headway consideration and maintenance
problems. Similarly increasing the number of pairs to the underground cable is uneconomical
and leads to maintenance problems.
It, therefore, became imperative to think of new technical innovations which could
exploit the available bandwidth of transmission media such as open wire lines or underground
cables to provide more number of circuits on one pair. The technique used to provide a number of
circuits using a single transmission link is called Multiplexing.
4.2 MULTIPLEXING TECHNIQUES:
There are basically two types of multiplexing techniques
i.

Frequency Division Multiplexing (FDM)

ii.

Time Division Multiplexing (TDM)

4.2.1 Frequency Division Multiplexing Techniques (FDM):

The FDM techniques is the process of translating individual speech circuits (300-3400
Hz) into pre-assigned frequency slots within the bandwidth of the transmission medium. The
frequency translation is done by amplitude modulation of the audio frequency with an
appropriate carrier frequency. At the output of the modulator a filter network is connected to
select either a lower or an upper side band. Since the intelligence is carried in either side band,
single side band suppressed carrier mode of AM is used. This results in substantial saving of
bandwidth mid also permits the use of low power amplifiers. Refer Fig. 1.
FDM techniques usually find their application in analogue transmission systems. An
analogue transmission system is one which is used for transmitting continuously varying signals.

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Fig. 1 FDM Principle

4.2.2 Time Division Multiplexing (TDM):

Basically, time division multiplexing involves nothing more than sharing
a transmission medium by a number of circuits in time domain by establishing a sequence of
time slots during which individual channels (circuits) can be transmitted. Thus the entire
bandwidth is periodically available to each channel. Normally all-time slots are equal in length.
Each channel is assigned a time slot with a specific common repetition period called a frame
interval. This is illustrated in Fig. 2.

Fig. 2 Time Division Multiplexing

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Each channel is sampled at a specified rate and transmitted for a fixed duration. All channels
are sampled one by, the cycle is repeated again and again. The channels are connected to individual
gates which are opened one by one in a fixed sequence. At the receiving end also similar gates
are opened in unison with the gates at the transmitting end.
The signal received at the receiving end will be in the form of discrete
samples and these are combined to reproduce the original signal. This staggering of channels
in time sequence for transmission over a common medium is called Time Division
Multiplexing (TDM).
4.3 Pulse Code Modulation:
It was only in 1938, Mr. A.M. Reaves (USA) developed a Pulse Code Modulation
(PCM) system to transmit the spoken word in digital form. Since then digital speech
transmission has become an alternative to the analogue systems.
PCM systems use TDM technique to provide a number of circuits on the same
transmission medium viz open wire or underground cable pair or a channel provided by carrier,
coaxial, microwave or satellite system.
Basic Requirements for PCM System:
To develop a PCM signal from several analogue signals, the following processing
steps are required

Filtering

Sampling

Quantization

Encoding

Line Coding

4.3.1 Filtering:
Filters are used to limit the speech signal to the frequency band 300-3400 Hz.
4.3.2 Sampling:
It is the most basic requirement for TDM. Suppose we have an analogue signal Fig.
3 (b), which is applied across a resistor R through a switch S as shown in Fig. 3 (a). Whenever
switch S is closed, an output appears across R. The rate at which S is closed is called the sampling
frequency because during the make periods of S, the samples of the analogue modulating
signal appear across R. Fig. 3(d) is a stream of samples of the input signal which appear across R.
The amplitude of the sample is depend upon the amplitude of the input signal at the instant of
sampling. The duration of these sampled pulses is equal to the duration for which the switch S is
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closed. Minimum number of samples are to be sent for any band limited signal to get a good
approximation of the original analogue signal and the same is defined by the sampling Theorem.

FIG. 3: SAMPLING PROCESS

Sampling Theorem:
Sampling Theorem States"If a band limited signal is sampled at regular intervals of time and at a rate equal to or
more than twice the highest signal frequency in the band, then the sample contains all the
information of the original signal." Mathematically, if fH is the highest frequency in the signal to
be sampled then the sampling frequency Fs needs to be greater than 2 fH.
i.e. Fs>2fH
Let us say our voice signals are band limited to 4 KHz and let sampling frequency be 8 KHz.
Time period of sampling Ts =

1 sec
8000

Or Ts = 125 micro seconds

If we have just one channel, then this can be sampled every 125 microseconds and the
resultant samples will represent the original signal. But, if we are to sample N channels one by
one at the rate specified by the sampling theorem, then the time available for sampling each
channel would be equal to Ts/N microseconds.
In a 30 channel PCM system. TS i.e. 125 microseconds are divided into 32 parts. That is 30
time slots are used for 30 speech signals, one time slot for signaling of all the 30 channels,
and one time slot for synchronization between Transmitter & Receiver.
The time available per channel would be Ts/N = 125/32 = 3.9 microseconds. Thus in a 30
channel PCM system, time slot is 3.9 microseconds and time period of sampling i.e. the interval
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between 2 consecutive samples of a channel is 125 microseconds. This duration i.e. 125
microseconds is called Time Frame.
The signals on the common medium (also called the common highway)
of a TDM system will consist of a series of pulses, the amplitudes of which are proportional to
the amplitudes of the individual channels at their respective sampling instants. This is illustrated
in Fig. 5

Fig 5: PAM Output Signals

The original signal for each channel can be recovered at the receive end by applying
gate pulses at appropriate instants and passing the signals through low pass filters. (Refer Fig. 6).

FIG. 6: RECONSTRUCTION OF ORIGINAL SIGNAL

Quantization:
The output of the sampler is a PAM signal as shown in Fig. 3; the transmission of PAM
signal will require linear amplifiers at Trans and receive ends to recover distortion less signals. This
type of transmission is susceptible to all the disadvantages of AM signal transmission. Therefore, in
PCM systems, PAM signals are converted into digital form by using Quantization Principles.
The discrete level of each sampled signal is quantized with reference to a certain specified level on
an amplitude scale.
The process of measuring the numerical values of the samples and giving them a table
value in a suitable scale is called "Quantizing". Of course, the scales and the number of points
should be so chosen that the signal could be effectively reconstructed after demodulation.
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Quantizing, in other words, can be defined as a process of breaking down a continuous

amplitude
range
into
a
finite
number
of
amplitude
values or steps.
For example, a 1 volt signal can be divided into 10mV ranges like 10-20mV, 30-40mV and
so on. The interval 10-20 mV, may be designated as level 1, 20-30 mV as level 2 etc. For the
purpose of transmission, these levels are given a binary code. This is called encoding. In practical
systems-quantizing and encoding are a combined process. For the sake of understanding, these
are treated separately.
Quantizing Process:
Suppose we have a signal as shown in Fig. 7 which is sampled at instants a, b, c, d and
e. For the sake of explanation, let us suppose that the signal has maximum amplitude of 7 volts.
In order to quantize these five samples taken of the signal, let us say the total amplitude is
divided into eight ranges or intervals as shown in Fig. 7. Sample (a) lies in the 5th range.
Accordingly, the quantizing process will assign a binary code corresponding to this i.e. 101,
similarly codes are assigned for other samples also. Here the quantizing intervals are of the
same size. This is called Linear Quantizing.

FIG. 7: QUANTIZING-POSITIVE SIGNAL

Assigning an interval of 5 for sample 1, 7 for 2 etc. is the quantizing
process. Giving, the assigned levels of samples, the binary code are
called coding of the quantized samples.
Quantizing is done for both positive and negative swings. As shown in
Fig.6, eight quantizing levels are used for each direction of the
analogue
signal.
To indicate
whether
a
sample
is
negative
with
reference to zero or is positive with reference zero, an extra digit is
added to the binary code. This extra digit is called the "sign bit". In Fig.
8. Positive values have a sign bit of ' 1 ' and negative values have sign
bit of'0'.
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Encoding:
Conversion of quantized analogue levels to binary signal is called encoding. To represent
256 steps, 8 level code is required. The eight bit code is also called an eight bit "word".
The 8 bit word appears in the form:
P

ABC

Polarity bit 0 for + ve '1' for - ve.

Segment Code

WXYZ
Linear encoding
in the segment

The first bit gives the sign of the voltage to be coded. Next 3 bits gives the segment
number. There are 8 segments for the positive voltages and 8 for negative voltages. Last 4 bits
give the position in the segment. Each segment contains 16 positions. Referring to Fig. 9(b),
voltage Vc will be encoded as 1 1 1 1 0101.

1.1.1.1.1 FIG. 9 (b): Encoding Curve with Compression 8 Bit Code

The quantization and encoding are done by a circuit called coder. The coder converts
PAM signals (i.e. after sampling) into an 8 bit binary signal. The coding is done as per Fig. 9
which shows a relationship between voltage V to be coded and equivalent binary number N. The
function N = f (v) is not linear.
The curve has the following characteristics.
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It is symmetrical about the origins. Zero level corresponds to zero voltage to be

encoded.
It is logarithmic function approximated by 13 straight segments numbered 0 to 7 in positive
direction and 'O' to 7 in the negative direction. However 4 segments 0, 1, 0, 1 lying between
levels + vm/64 -vm/64 being collinear are taken as one segment.
The voltage to be encoded corresponding to 2 ends of successive segments are in the ratio
of 2. That is vm, vm/2, vm/4, vm/8, vm/16, vm/32, vm/64, vm/128 (vm being the maximum
voltage).

5) OPTICAL-FIBRE COMMUNICATIONS
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5.1 Brief History:

Optical communication systems date back to the 1790s, to the optical semaphore
telegraph invented by French inventor Claude Chappe. In 1880, Alexander Graham Bell patented
an optical telephone system, which he called the Photo phone. However, his earlier invention, the
telephone, was more practical and took tangible shape.
By 1964, a critical and theoretical specification was identified by Dr. Charles K. Kao for
long-range communication devices, the 10 or 20 dB of light loss per kilometer standard. Dr. Kao
also illustrated the need for a purer form of glass to help reduce light loss. By 1970 Corning
Glass invented fibre-optic wire or "optical waveguide fibres" which was capable of carrying
65,000 times more information than copper wire, through which information carried by a pattern
of light waves could be decoded at a destination even a thousand miles away. Corning Glass
developed an SMF with loss of 17 dB/km at 633 nm by doping titanium into the fibre core. By
June of 1972, multimode germanium-doped fibre had developed with a loss of 4 dB per
kilometer and much greater strength than titanium-doped fibre.
In April 1977, General Telephone and Electronics tested and deployed the world's first
live telephone traffic through a fibre-optic system running at 6 Mbps, in Long Beach, California.
They were soon followed by Bell in May 1977, with an optical telephone communication system
installed in the downtown Chicago area, covering a distance of 1.5 miles (2.4 kilometers). Each
optical-fibre pair carried the equivalent of 672 voice channels and was equivalent to a DS3
circuit. Today more than 80 percent of the world's long-distance voice and data traffic is carried
over optical-fibre cables.
5.2 Fibre-Optic Applications:
The use and demand for optical fibre has grown tremendously and optical-fibre
applications are numerous. Telecommunication applications are widespread, ranging from global
networks to desktop computers. These involve the transmission of voice, data, or video over
distances of less than a meter to hundreds of kilometers, using one of a few standard fibre
designs in one of several cable designs.
Optical fibre is also used extensively for transmission of data. Multinational firms need
secure, reliable systems to transfer data and financial information between buildings to the
desktop terminals or computers and to transfer data around the world.
Cable television companies also use fibre for delivery of digital video and data services.
The high bandwidth provided by fibre makes it the perfect choice for transmitting broadband
signals, such as high-definition television (HDTV) telecasts. Intelligent transportation systems,
such as smart highways with intelligent traffic lights, automated tollbooths, and changeable
message signs, also use fibre-optic-based telemetry systems.
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Another important application for optical fibre is the biomedical industry. Fibre-optic systems
are used in most modern telemedicine devices for transmission of digital diagnostic images.
Other applications for optical fibre include space, military, automotive, and the industrial sector.

5.3 Fibre Optic System:

Optical Fibre is new medium, in which information (voice, Data or Video) is transmitted through
a glass or plastic fibre, in the form of light, following the transmission sequence give below:
(1)

Information is encoded into Electrical Signals.

(2)

Electrical Signals are converted into light Signals.

(3)

Light Travels down the Fibre.

(4)

A Detector Changes the Light Signals into Electrical Signals.

(5)

Electrical Signals are decoded into Information.

Fig. Principle of Fibre optic transmission system

5.4 Principle of Operation Theory:

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Total
Internal
Reflection: The Reflection that Occurs when a Light Ray Travelling in One Material
Hits a Different Material and Reflects Back into the Original Material without any
Loss of Light.
Fig. 2

Speed of light is actually the velocity of electromagnetic energy in vacuum such as space. Light
travels at slower velocities in other materials such as glass. Light travelling from one material to
another changes speed, which results in light changing its direction of travel. This deflection of
light is called Refraction.
The amount that a ray of light passing from a lower refractive index to a higher one is bent
towards the normal. But light going from a higher index to a lower one refracting away from the
normal, as shown in the figures.
Angle of incidence

1
n1
n2
2

Light is bent away

from normal

n1
n2

Angle of
reflection

n1
n2

Light does not enter

second material

As the angle of incidence increases, the angle of refraction approaches 90o to the normal. The
angle of incidence that yields an angle of refraction of 90o is the critical angle. If the angle of
incidence increases amore than the critical angle, the light is totally reflected back into the first
material so that it does not enter the second material. The angle of incidence and reflection are
equal and it is called Total Internal Reflection.
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PROPAGATION OF LIGHT THROUGH FIBRE:

The optical fibre has two concentric layers called the core and the cladding. The inner core is the
light carrying part. The surrounding cladding provides the difference refractive index that allows
total internal reflection of light through the core. The index of the cladding is less than 1%, lower
than that of the core. Typical values for example are a core refractive index of 1.47 and a
cladding index of 1.46. Fibre manufacturers control this difference to obtain desired optical fibre
characteristics. Most fibres have an additional coating around the cladding. This buffer coating
is a shock absorber and has no optical properties affecting the propagation of light within the
fibre. Figure shows the idea of light travelling through a fibre. Light injected into the fibre and
striking core to cladding interface at greater than the critical angle, reflects back into core, since
the angle of incidence and reflection are equal, the reflected light will again be reflected. The
light will continue zigzagging down the length of the fibre. Light striking the interface at less
than the critical angle passes into the cladding, where it is lost over distance. The cladding is
usually inefficient as a light carrier, and light in the cladding becomes attenuated fairly.
Propagation of light through fibre is governed by the indices of the core and cladding by Snell's
law.
Such total internal reflection forms the basis of light propagation through an optical fibre. This
analysis consider only meridional rays- those that pass through the fibre axis each time, they are
reflected. Other rays called Skew rays travel down the fibre without passing through the axis.
The path of a skew ray is typically helical wrapping around and around the central axis.
Fortunately skew rays are ignored in most fibre optics analysis.

Jacket

Jacket
Cladding
Core

Cladding (n2)
Core (n2)

Cladding
Jacket
Light at less than Angle of Angle of
critical angle is
incidence reflection
absorbed in jacket
Light is propagated by
total internal reflection
Fig. Total Internal Reflection in an optical Fibre

FIG. PROPAGATION OF LIGHT THROUGH FIBRE

Fibre sizes are usually expressed by first giving the core size followed by the cladding size. Thus
50/125 means a core diameter of 50m and a cladding diameter of 125m.
5.5 FIBRE TYPES:
The refractive Index profile describes the relation between the indices of the core and cladding.
Two main relationship exists:
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(I)

Step Index

(II)

Graded Index

The step index fibre has a core with uniform index throughout. The profile shows a sharp step at
the junction of the core and cladding. In contrast, the graded index has a non-uniform core. The
Index is highest at the center and gradually decreases until it matches with that of the cladding.
There is no sharp break in indices between the core and the cladding.
By this classification there are three types of fibres:
(I)

Multimode Step Index fibre (Step Index fibre)

(II)

Multimode graded Index fibre (Graded Index fibre)

(III)

Single- Mode Step Index fibre (Single Mode Fibre)

5.6.1 STEP-INDEX MULTIMODE FIBRE: It has a large core, up to 100 microns in diameter.
As a result, some of the light rays that make up the digital pulse may travel a direct route,
whereas others zigzag as they bounce off the cladding. These alternative pathways cause the
different groupings of light rays, referred to as modes, to arrive separately at a receiving point.
The pulse, an aggregate of different modes, begins to spread out, losing its well-defined shape.
The need to leave spacing between pulses to prevent overlapping limits bandwidth that is, the
amount of information that can be sent. Consequently, this type of fibre is best suited for
transmission over short distances, in an endoscope, for instance.

Fig. 6 STEP-INDEX MULTIMODE FIBRE

5.6.2 GRADED-INDEX MULTIMODE FIBRE: It contains a core in which the refractive
index diminishes gradually from the center axis out toward the cladding. The higher refractive
index at the center makes the light rays moving down the axis advance more slowly than those
near the cladding.

Fig.7 GRADED-INDEX MULTIMODE FIBRE

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Also, rather than zigzagging off the cladding, light in the core curves helically because of
the graded index, reducing its travel distance. The shortened path and the higher speed allow
light at the periphery to arrive at a receiver at about the same time as the slow but straight rays in
the core axis. The result: a digital pulse suffers less dispersion.
5.6.3 SINGLE-MODE FIBRE: It has a narrow core (eight microns or less), and the index of
refraction between the core and the cladding changes less than it does for multimode fibres.
Light thus travels parallel to the axis, creating little pulse dispersion. Telephone and cable
television networks install millions of kilometers of this fibre every year.

Fig. 8 SINGLE-MODE FIBRE

5.6 OFC Splicing:

Splices are permanent connection between two fibres. The splicing involves cutting of the
edges of the two fibres to be spliced.
Splicing Methods
The following three types are widely used:
1.

Adhesive bonding or Glue splicing.

2.

Mechanical splicing.

3.

Fusion splicing.

5.9.1 Adhesive Bonding or Glue Splicing:

This is the oldest splicing technique used in fibre splicing. After fibre end preparation, it
is axially aligned in a precision Vgroove. Cylindrical rods or another kind of reference surfaces
are used for alignment. During the alignment of fibre end, a small amount of adhesive or glue of
same refractive index as the core material is set between and around the fibre ends. A two
component epoxy or an UV curable adhesive is used as the bonding agent. The splice loss of this
type of joint is same or less than fusion splices. But fusion splicing technique is more reliable, so
at present this technique is very rarely used.
5.9.2 Mechanical Splicing:

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This technique is mainly used for temporary splicing in case of emergency repairing. This
method is also convenient to connect measuring instruments to bare fibres for taking various
measurements.
The mechanical splices consist of 4 basic components:
(i)

An alignment surface for mating fibre ends.

(ii)

A retainer

(iii)

An index matching material.

(iv)

A protective housing

5.9.3 Fusion Splicing:

The fusion splicing technique is the most popular technique used for achieving very low
splice losses. The fusion can be achieved either through electrical arc or through gas flame.
The process involves cutting of the fibres and fixing them in micropositioners on the
fusion splicing machine. The fibres are then aligned either manually or automatically core
aligning (in case of S.M. Fibre) process. Afterwards the operation that takes place involve
withdrawal of the fibres to a specified distance, preheating of the fibre ends through electric arc
and bringing together of the fibre ends in a position and splicing through high temperature
fusion.
If proper care taken and splicing is done strictly as per schedule, then the splicing loss
can be minimized as low as 0.01 dB/joint.
The splice loss indicated by the splicing machine should not be taken as a final value as it
is only an estimated loss and so after every splicing is over, the splice loss measurement is to be
taken by an OTDR (Optical Time Domain Reflectometer). The manual part of the splicing is
cleaning and cleaving the fibres. For cleaning the fibres, Dichlorine Methyl or Acetone or
Alcohol is used to remove primary coating.
It is also desirable to limit the average splice loss to be less than 0.1 dB.

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6) DIGITAL HIERARCHIES:
The term digital hierarchy has been created when developing digital transmission
systems. Consequently, a digital hierarchy comprises a number of levels. Each level is assigned a
specific bit rate which is formed by multiplexing digital signals, each having the bit rate of the
next lower level. In CCITT Rec. G.702, the term digital multiplex hierarchy is defined as
follows:
A series of digital multiplexes graded according to capability so that multiplexing at one
level combines a defined number of digital signals, each having the digit rate prescribed for the
next lower order, into a digital signal having a prescribed digit rate which is then available for
further combination with other digital signals of the same rate in a digital multiplex of the next
higher order.
6.1 OVERVIEW OF PDH:
With the introduction of PCM technology in the 1960s, communications networks were
gradually converted to digital technology over the next few years. To cope with the demand for
even higher bit rates, a multiplex hierarchy called the plesiochronous digital hierarchy (PDH)
evolved. The bit rates start with the basic multiplex rate of 2 Mbit/s with further stages of 8, 34
and 140 Mbit/s. In North America and Japan, the primary rate is 1.5 Mbit/s. Hierarchy stages of
6 and 44 Mbit/s developed from this. Because of these very different developments, gateways
between one network and another were very difficult and expensive to realize. PCM allows
multiple use of a single line by means of digital time-domain multiplexing. The analog telephone
signal is sampled at a bandwidth of 3.1 kHz, quantized and encoded and then transmitted at a bit
rate of 64 Kbit/s. A transmission rate of 2048 Kbit/s results when 30 such coded channels are
collected together into a frame along with the necessary signaling information. This so-called
primary rate is used throughout the world. Only the USA, Canada and Japan use a primary rate
of 1544 Kbit/s, formed by combining 24 channels instead of 30. The growing demand for more
bandwidth meant that more stages of multiplexing were needed throughout the world. A
practically synchronous (or, to give it its proper name: plesiochronous) digital hierarchy is the
result. Slight differences in timing signals mean that justification or stuffing is necessary when
forming the multiplexed signals. Inserting or dropping an individual 64 Kbit/s channel to or from
a higher digital hierarchy requires a considerable amount of complex multiplexer equipment.

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Fig 1. - Plesiochronous Digital Hierarchies (PDH)

Traditionally, digital transmission systems and hierarchies have been based on multiplexing
signals which are plesiochronous (running at almost the same speed). PDH requires steps (14034, 34-8, 8-2 demultiplex; 2-8, 8-34, 34-140 multiplex) to drop out or add an individual speech
or data channel (see Fig 1).
6.1.2 The main problems of PDH systems are:
1. Homogeneity of equipment.
2. Problem of Channel segregation.
3. The problem cross connection of channels.
4. Theres no standardized definition of PDH bit rates greater than 140 Mbit/s.
5. There are different hierarchies in use around the world. Specialized interface equipment
is required to interwork the two hierarchies.
6. Each multiplexing section has to add overhead bits for justification.
7. add-drop-multiplexers are hard to build.
8. The management and monitoring functions were not sufficient in PDH.
9. PDH did not define a standard format on the transmission link.

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6.2 SDH/SONET Introduction:

Started by Bell core in 1985 as standardization effort for the US
telephone carriers (after AT&T was broken up in 1984),
Later joined by CCITT (later: ITU), which formed SDH in 1987
Three major goals:
Avoid the problems of PDH
Achieve higher bit rates (Gbit/s)
Better means for Operation, Administration, and Maintenance
(OA&M)
SDH is THE standard in telecommunication networks now
It is designed to transport voice rather than data
It covers the lower 2-3 OSI layers
SONET/SDH defines only a point-to-point connection in the network

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SDH is an ITU-T standard for a high capacity telecom network. SDH is a synchronous digital
transport system, aim to provide a simple, economical and flexible telecom infrastructure. The
basis of Synchronous Digital Hierarchy (SDH) is synchronous multiplexing - data from multiple
tributary sources is byte interleaved
6.2.1 Features of SDH:
SDH brings the following advantages to network providers:
High transmission rates:
Transmission rates of up to 40 Gbit/s can be achieved in modern SDH systems. SDH is therefore
the most suitable technology for backbones, which can be considered as being the super
highways in today's telecommunications networks.
Simplified add & drop function:
Compared with the older PDH system, it is much easier to extract and insert low-bit rate
channels from or into the high-speed bit streams in SDH. It is no longer necessary to demultiplex
and then remultiplex the plesiochronous structure.

High availability and capacity matching:

With SDH, network providers can react quickly and easily to the requirements of their
customers. For example, leased lines can be switched in a matter of minutes. The network
provider can use standardized network elements that can be controlled and monitored from a
central location by means of a telecommunications network management (TMN) system.
Reliability:
Modern SDH networks include various automatic back-up and repair mechanisms to cope with
system faults. Failure of a link or a network element does not lead to failure of the entire network
which could be a financial disaster for the network provider. These back-up circuits are also
monitored by a management system.
Future-proof platform for new services:
Right now, SDH is the ideal platform for services ranging from POTS, ISDN and mobile radio
through to data communications (LAN, WAN, etc.), and it is able to handle the very latest
services, such as video on demand and digital video broadcasting via ATM that are gradually
becoming established.
Interconnection:
SDH makes it much easier to set up gateways between different network providers and to
SONET systems. The SDH interfaces are globally standardized, making it possible to combine
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network elements from different manufacturers into a network. The result is a reduction in
equipment costs as compared with PDH.

6.3 SDH Rates:

SDH is a transport hierarchy based on multiples of 155.52 Mbit/s. The basic unit of SDH is
STM-1. Different SDH rates are given below:
STM-1 = 155.52 Mbit/s
STM-4 = 622.08 Mbit/s
STM-16 = 2588.32 Mbit/s
STM-64 = 9953.28 Mbit/s
Each rate is an exact multiple of the lower rate therefore the hierarchy is synchronous.

6.4 Merits of SDH:

(i)

Simplified multiplexing/ demultiplexing techniques.

(ii)

Direct access to lower speed tributaries, without need to multiplex/demultiplex

the entire high speed signal.

(iii)

Enhanced operations, Administration, Maintenance and provisioning capabilities.

(iv)

Easy growth to higher bit rates in step with evolution of transmission technology.

(v)

Capable of transporting existing PDH signals.

(vi)

Capable of transporting future broadband (ATM) channel bit rates.

(vii)

Capable of operating in a multi-vendor and multi-operator environment.

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7) MOBILE COMMUNICATION-GSM & CDMA

7.1 Introduction:
A GSM system is basically designed as a combination of three major subsystems: the network
subsystem, the radio subsystem, and the operation support subsystem. In order to ensure that
network operators will have several sources of cellular infrastructure equipment, GSM decided
to specify not only the air interface, but also the main interfaces that identify different parts.
There are three dominant interfaces, namely, an interface between MSC and the Base Transceiver
Station (BTS), and an Um interface between the BTS and MS.

7.2 Subsystems and network elements in GSM:

The GSM network is called Public Land Mobile Network (PLMN). It is organized in three
subsystems:

Network Switching Subsystem (NSS)

Base Station Subsystem (BSS)

Network Management Subsystem (NMS)

The three subsystems, different network elements, and their respective tasks are presented in the
following.

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Network Switching Subsystem (NSS)

The Network Switching Subsystem (NSS) contains the network elements MSC, GMSC, VLR,

HLR, AC and EIR.

The Network Switching Subsystem (NSS)

The main functions of NSS are:

Call control
This identifies the subscriber, establishes a call, and clears the connection after the conversation
is over.
Charging
This collects the charging information about a call (the numbers of the caller and the called
subscriber, the time and type of the transaction, etc.) and transfers it to the Billing Centre.
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Mobility management
This maintains information about the subscriber's location.
Signaling
This applies to interfaces with the BSS and PSTN.
Subscriber data handling
This is the permanent data storage in the HLR and temporary storage of relevant data in the
VLR.

Mobile services Switching Centre (MSC):

Mobile-services Switching Centre (MSC) performs the switching functions for all mobile
stations located in the geographic area covered by its assigned BSSs. Functions performed
include interfacing with the Public Switched Telephone Network (PSTN) as well as with the
other MSCs and other system entities, such as the HLR, in the PLMN.
Functions of the MSC include:

Call handling that copes with mobile nature of subscribers (e.g., paging)

Management of required logical radio-link channel during calls

Management of MSC-BSS signalling protocol

Handling location registration and ensuring interworking between Mobile Station and VLR

Control of inter-BSS and inter-MSC handovers

Acting as a gateway MSC to interrogate the HLR

Exchange of signalling information with other system entities

Standard functions of a local exchange switch in the fixed network (example: charging)

Base Station Subsystem (BSS)

The Base Station Subsystem is responsible for managing the radio network, and it is controlled
by an MSC. Typically, one MSC contains several BSSs. A BSS itself may cover a considerably
large geographical area consisting of many cells (a cell refers to an area covered by one or more
frequency resources). The BSS consists of the following elements:
BSC Base Station Controller
BTS Base Transceiver Station
TRAU Transcoder and Rate Adaptation Unit (often referred to as TC (Transcoder))

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Radio path control

In the GSM network, the Base Station Subsystem (BSS) is the part of the network taking care of
radio resources, that is, radio channel allocation and quality of the radio connection.
7.3 OPERATION AND MAINTENANCE CENTER (OMC)
The Operations and Maintenance Center (OMC) is the centralized maintenance and diagnostic
heart of the Base Station System (BSS). It allows the network provider to operate, administer,
and monitor the functioning of the BSS.

MOBILE STATION:
The MS includes radio equipment and the man machine interface (MMI) that a subscribe needs
in order to access the services provided by the GSM PLMN. MS can be installed in Vehicles or
can be portable or handheld stations. The MS may include provisions for data communication as
well as voice. A mobile transmits and receives message to and from the GSM system over the air
interface to establish and continue connections through the system.
Different type of MSs can provide different type of data interfaces. To provide a common model
for describing these different MS configuration, reference configuration for MS, similar to
those defined for ISDN land stations, has been defined.

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Functions of MS:
The primary functions of MS are to transmit and receive voice and data over the air interface of
the GSM system. MS performs the signal processing function of digitizing, encoding, error
protecting, encrypting, and modulating the transmitted signals. It also performs the inverse
functions on the received signals from the BS.
In order to transmit voice and data signals, the mobile must be in synchronization with the
system so that the messages are the transmitted and received by the mobile at the correct instant.
To achieve this, the MS automatically tunes and synchronizes to the frequency and TDMA
timeslot specified by the BSC. This message is received over a dedicated timeslot several times
within a multiframe period of 51 frames. We shall discuss the details of this in the next chapter.
The exact synchronization will also include adjusting the timing advance to compensate for
varying distance of the mobile from the BTS.
The MS monitors the power level and signal quality, determined by the BER for known receiver
bit sequences (synchronization sequence), from both its current BTS and up to six surrounding
BTSs. This data is received on the downlink broadcast control channel. The MS determines and
send to the current BTS a list of the six best-received BTS signals. The measurement results from
MS on downlink quality and surrounding BTS signal levels are sent to BSC and processed
within the BSC. The system then uses this list for best cell handover decisions.
MS keeps the GSM network informed of its location during both national and international
roaming, even when it is inactive. This enables the System to page in its present LA.
The MS includes an equalizer that compensates for multi-path distortion on the received signal.
This reduces inter-symbol interface that would otherwise degrade the BER.
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BSC:
The BSC, as discussed, is connected to the MSC on one side and to the BTS on the other. The
BSC performs the Radio Resource (RR) management for the cells under its control. It assigns
and release frequencies and timeslots for all MSs in its own area. The BSC performs the intercell
handover for MSs moving between BTS in its control. It also reallocates frequencies to the BTSs
in its area to meet locally heavy demands during peak hours or on special events. The BSC
controls the power transmission of both BSSs and MSs in its area. The minimum power level for
a mobile unit is broadcast over the BCCH. The BSC provides the time and frequency
synchronization reference signals broadcast by its BTSs. The BSC also measures the time delay
of received MS signals relative to the BTS clock. If the received MS signal is not centered in its
assigned timeslot at the BTS, The BSC can direct the BTS to notify the MS to advance the
timing such that proper synchronization takes place. The functions of BSC are as follows.
The BSC may also perform traffic concentration to reduce the number of transmission
lines from the BSC to its BTSs, as discussed in the last section.

7.4 EVOLUTION FROM GSM TO 3G:

2G

2.5G

GSM

GPRS

GSM
GPRS
200 KHz carrier
200 KHz carrier
8 full-rate time slots 115 Kbps peak data rates
16 half-rate time slots

3G

EDGE

UMTS

UMTS
EDGE
5 MHz carrier
200 KHz carrier
Data rates up to 384 Kbps 2 Mbps peak data rates
New IMT-2000 2 GHz spectrum
8-PSK modulation
Higher symbol rate

HSCSD
HSCSD
Circuit-switched data
64 Kbps peak data rates

EDGE (ENHANCED DATA FOR GSM EVOLUTION):

Increased data rated up to 384 Kbps by bundling up to 8 channels of 48 Kbps/channel

GPRS is based on modulation technique known as GMSK

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EDGE is based on a new modulation scheme that allows a much higher bit rate across the airinterface called 8PSK modulation.

Since 8PSK will be used for UMTS, network operators will be required to introduce this at
some stage before migration to 3G.

7.5 INTRODUCTION TO CDMA2000:

The first operational cellular communication system was deployed in the Norway in 1981 and
was followed by similar systems in the US and UK. These first generation systems provided
voice transmissions by using frequencies around 900 MHz and analogue modulation.
The second generation (2G) of the wireless mobile network was based on low-band digital data
signaling. While GSM and other TDMA-based systems have become the dominant 2G
wirelesses technologies, CDMA technology IS 95A is recognized as providing clearer voice
quality with less background noise, fewer dropped calls, enhanced security, greater reliability
and greater network capacity.
The Second Generation (2G) wireless networks mentioned above are also mostly based on
circuit-switched technology, are digital and expand the range of applications to more advanced
voice services. 2G wireless technologies can handle some data capabilities such as fax and short
message service at the data rate of up to 9.6 kbps, but it is not suitable for web browsing and
multimedia applications. In the world of 2G, voice remains king while data is already dominant
in wire line communications. And, fixed or wireless, all are affected by the rapid growth of the
Internet.
Hence in mobile world also the aim was to achieve higher data speed. ITU also proposed the
conceptual 3G.
3G OR IMT-2000:
International Mobile Telecommunications-2000 (IMT-2000) is the official International
Telecommunication Union name for 3G and is an initiative intended to provide wireless access to
global telecommunication infrastructure through both satellite and terrestrial systems, serving
fixed and mobile phone users via both public and private telephone networks. Today's 3G
specifications call for 144 Kb/s while the user is on the move in an automobile or train, 384 Kb/s
for pedestrians, and ups to 2 Mb/s for stationary users. That is a big step up from 2G bandwidth
using 8 to 13 Kb/s per channel to transport speech signals. But no single technology could be
evolved as 3G.

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7.6 OVERVIEW OF WCDMA:

BACKGROUND:
There has been a tremendous growth in wireless communication technology over the past
decade. The significant increase in subscribers and traffic, new bandwidth consuming
applications such as gaming, music down loading and video streaming will place new demands
on capacity. The answer to the capacity demand is the provision of new spectrum and the
development of a new technology Wideband CDMA or hereinafter referred to as WCDMA.
WCDMA was developed in order to create a global standard for real time multimedia services
that ensured international roaming. With the support of ITU (International Telecommunication
Union) a specific spectrum was allocated 2GHz for 3G telecom systems. The work was later
taken over by the 3GPP (3rd Generation Partnership Project), which is now the WCDMA
specification body with delegates from all over the world. Ericsson has for a long time played a
very active role in both ITU and 3GPP and is a major contributor to WCDMA and the fulfillment
of the vision of a global mobile telecommunication system.
WCDMA A DEVELOPMENT FROM GSM AND CDMA:
Naturally there are a lot of differences between WCDMA and GSM systems, but there are many
similarities as well. The GSM Base Station Subsystem (BSS) and the WCDMA Radio Access
Network (RAN) are both connected to the GSM core network for providing a radio connection to
the handset. Hence, the technologies can share the same core network. Furthermore, both GSM
BSS and WCDMA RAN systems are based on the principles of a cellular radio system. The
GSM Base Station Controller (BSC) corresponds to the WCDMA Radio Network Controller
(RNC). The GSM Radio Base Station (RBS) corresponds to the WCDMA RBS, and the Ainterface of GSM was the basis of the development of the Iu-interface of WCDMA, which
mainly differs in the inclusion of the new services offered by WCDMA. The significant
differences, apart from the lack of interface between the GSM BSCs and an insufficiently
specified GSM Abis-interface to provide multi-vendor operability, are more of a systemic matter.
The GSM system uses TDMA (Time Division Multiple Access) technology with a lot of radio
functionality based on managing the timeslots. The WCDMA system on the other hand uses
CDMA, as described below, which means that both the hardware and the control functions are
different. Examples of WCDMA-specific functions are fast power control and soft handover.
Code Division Multiple Access and WCDMA:
Code Division Multiple Access (CDMA) is a multiple access technology where the users are
separated by unique codes, which means that all users can use the same frequency and transmit
at the same time. With the fast development in signal processing, it has become feasible to use
the technology for wireless communication, also referred to as WCDMA and CDMA2000. In
cdmaOne and CDMA2000, a 1.25 MHz wide radio signal is multiplied by a spreading signal
(which is a pseudo-noise code sequence) with a higher rate than the data rate of the message. The
resultant signal appears as seemingly random, but if the intended recipient has the right code, this
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process is reversed and the original signal is extracted. Use of unique codes means that the same
frequency is repeated in all cells, which is commonly referred to as a frequency re-use of 1.
WCDMA is a step further in the CDMA technology. It uses a 5 MHz wide radio signal and a chip
rate of 3.84 Mcps, which is about three times higher than the chip rate of CDMA2000 (1.22
Mcps). The main benefits of a wideband carrier with a higher chip rate are:
Support for higher bit rates
Higher spectrum efficiency thanks to improved trunking efficiency (i.e. a better statistical
averaging)
Higher QoS
Further, experience from second-generation systems like GSM and cdmaOne has enabled
improvements to be incorporated in WCDMA. Focus has also been put on ensuring that as much
as possible of WCDMA operators investments in GSM equipment can be reused. Examples are
the re-use and evolution of the core network, the focus on co-siting and the support of GSM
handover. In order to use GSM handover the subscribers need dual mode handsets.

CDMA TECHNOLOGY:
Access Network:
Access network, the network between local exchange and subscriber, in the Telecom
Network accounts for a major portion of resources both in terms of capital and manpower. So far,
the subscriber loop has remained in the domain of the copper cable providing cost effective
solution in past. Quick deployment of subscriber loop, coverage of inaccessible and remote
locations coupled with modern technology have led to the emergence of new Access
Technologies. The various technological options available are as follows i:
1.

Multi Access Radio Relay

2.

Wireless in Local Loop

3.

Fibre in the Local Loop

WIRELESS IN LOCAL LOOP (WLL):

Fixed Wireless telephony in the subscriber access network also known as Wireless in Local Loop
(WLL) is one of the hottest emerging market segments in global telecommunications today.
WLL is generally used as the last mile solution to deliver basic phone service expeditiously
where none has existed before. Flexibility and expediency are becoming the key driving factors
behind the deployment of WILL.
WLL shall facilitate cordless telephony for residential as well as commercial complexes where
people are highly mobile. It is also used in remote areas where it is uneconomical to lay cables
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and for rapid development of telephone services. The technology employed shall depend upon
various radio access techniques, like FDMA, TDMA and CDMA.
Different technologies have been developed by the different countries like CT2 from France,
PHS from Japan, DECT from Europe and DAMPS & CDMA from USA. Let us discuss CDMA
technology in WLL application as it has a potential ability to tolerate a fair amount of
interference as compared to other conventional radios. This leads to a considerable advantage
from a system point of view.
7.7 SPREAD SPECTRUM PRINCIPLE:
Originally Spread spectrum radio technology was developed for military use to counter the
interference by hostile jamming. The broad spectrum of the transmitted signal gives rise to
Spread Spectrum. A Spread Spectrum signal is generated by modulating the radio frequency
(RF) signal with a code consisting of different pseudo random binary sequences, which is
inherently resistant to noisy signal environment.
A number of Spread spectrum RF signals thus generated share the same frequency spectrum and
thus the entire bandwidth available in the band is used by each of the users using same frequency
at the same time.

Fig-1 CDMA ACCESS A CONCEPT

On the receive side only the signal energy with the selected binary sequence code is accepted and
original information content (data) is recovered. The other users signals, whose codes do not
match contribute only to the noise and are not despread back in bandwidth (Ref Fig-1) This
transmission and reception of signals differentiated by codes using the same frequency
simultaneously by a number of users is known as Code Division Multiple Access (CDMA)
Technique as opposed to conventional method of Frequency Division Multiple Access and Time
Division Multiple Access.
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In the above figure, it has been tried to explain that how the base band signal of 9.6 Kbps is
spread using a Pseudo-random Noise (PN) source to occupy entire bandwidth of 1.25 Mhz. At
the receiving end this signal will have interference from signals of other users of the same cell,
users of different cells and interference from other noise sources. All these signals get combined
with the desired signal but using a correct PN code the original data can be reproduced back.
CDMA channel in the Trans and receive direction is a FDD (Frequency Division Duplexing)
channel. The salient features of a typical CDMA system are as follows:
Frequency of operation:

824-849MHz and 869-894 MHz

Duplexing Method:

Frequency Division Duplexing (FDD)

Access Channel per carrier: Maximum 61 Channels

RF Spacing:

1.25 MHz

Coverage:

5 Km with hand held telephones and approx.

20 Km with fixed units.

The different types of codes used for identification of traffic channels and users identification etc
as follows:

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