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“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-283………………………………………………………7

2.1 Salient Features of OCB-283………………………………………………7

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 Modulation……………………………………………………14

5. Optical-Fibre Communications……………………………………………….20

5.1 Brief History………………………………………………………………20

5.2 Fibre-Optics Applications…………………………………………………21

5.2 Fibre-optics System………………………………………………………..21

5.3 Principle of Operation……………………………………………………..21

5.4 Fibre Types………………………………………………………………..23

5.5 OFC Splicing……………………………………………………………...24

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6. Digital Hierarchies…………………………………………………………….27

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 Centre…………………………………………35

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)

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1) HISTORY OF TELEPHONE EXCHANGE

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 (out-of-band)

1990s Advanced Intelligent Network (AIN)

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2) INTRODUCTION TO OCB-283

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.

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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.

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:

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

1.1.1.1.1 S

M

X

1.1.1.1.2 S

SMT

( 1 TO 28) X 2

SMA

( 2 TO 37)

SMC

2 TO 14

1 TO 4 MAS

SMM

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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.

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.

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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.

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 customer‟s 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.

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4) PCM PRINCIPLES:

4.1 INTRODUCTION:

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 WXYZ

Polarity bit „0‟ for + ve '1' for - ve. Segment Code 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.

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The curve has the following characteristics.

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).

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5) OPTICAL-FIBRE COMMUNICATIONS

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

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5.4 Principle of Operation – Theory:

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.

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

ø1

Angle of incidence

n1

n2

ø2

n1

n2

ø1

ø2

n1

n2

ø1 ø2

Angle of

reflection

Light is bent away

from normal

Light does not enter

second material

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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.

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.

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.

Jacket

Cladding

Core

Cladding

Angle of

reflection

Angle of

incidence

Light at less than

critical angle is

absorbed in jacket

Jacket

Light is propagated by

total internal reflection

Jacket

Cladding

Core

(n2)

(n2)

Fig. Total Internal Reflection in an optical Fibre

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5.5 FIBRE TYPES:

The refractive Index profile describes the relation between the indices of the core and cladding.

Two main relationship exists:

(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

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

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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 V–groove. 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:

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.

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(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 micro–positioners 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” (140-

34, 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. There‟s 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:

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

air-interface 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 A-

interface 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.

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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

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

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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.

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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