BHARAT SANCHA RNIGAM LIMITED

(A Govt. of India Enterprise)

SUMMER TRAINING REPORT
SUBMITTED BY:
GAURAV (194)
IV SEM
BIT PATNA
ACKNOWLEDGEMENT
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 have been instrumental in the successful completion of this study
project. First of fall I would like to thank the Management at BSNL for giving me
the opportunity to do my one-month project training in their esteemed
organization.
GAURAV
IV SEM
BIT Patna.
COMPANY PROFILE
Bharat Sanchar Nigam Ltd. formed in October, 2000, is World's 7th largest Telecommunications Company
providing comprehensive range of telecom services in India: Wire line, CDMA mobile, GSM Mobile, Internet,
Broadband, Carrier service, MPLS-VPN, VSAT, VoIP services, IN Services etc.
Bharat Sanchar Nigam Limited (known as BSNL, India Communications Corporation Limited) is a public sector
communications company in India. It is the India's largest telecommunication company with 24% market
share. Its headquarters are at Bharat Sanchar Bhawan, New Delhi. It has the status of Mini-ratna - a status
assigned to reputed Public Sector companies in India.BSNL has installed Quality Telecom Network in the
country and now focusing on improving it, expanding the network, introducing new telecom services with
ICT applications in villages and wining customer's confidence. Today, it has about 47.3 million line basic
telephone capacity, 4 million WLL capacity, 20.1 Million GSM Capacity, more than 37382 fixed exchanges,
18000 BTS, 287 Satellite Stations, 480196 Rkm of OFC Cable, 63730 Rkm of Microwave Network connecting
602 Districts, 7330 cities/towns and 5.5 Lakhs villages. BSNL is the only service provider, making focused
efforts and planned initiatives to bridge the Rural-Urban Digital Divide ICT sector. In fact there is no telecom
operator in the country to beat its reach with its wide network giving services in every nook & corner of
country and operates across India except Delhi & Mumbai including inaccessible areas of Siachen glacier and
North-eastern region of the country.
BSNL cellular service, CellOne, has more than 17.8 million cellular customers, garnering 24 percent of all
mobile users as its subscribers. That means that almost every fourth mobile user in the country has a BSNL
connection. In basic services, BSNL is miles ahead of its rivals, with 35.1 million Basic Phone subscribers i.e.
85 per cent share of the subscriber base and 92 percent share in revenue terms. BSNL has more than 2.5
million WLL subscribers and 2.5 million Internet Customers who access Internet through various modes viz.
Dial-up, Leased Line, DIAS, and Account Less Internet (CLI). BSNL has been adjudged as the NUMBER ONE
ISP in the country.
BSNL has set up a world class multi-gigabit, multi-protocol convergent IP infrastructure that provides
convergent services like voice, data and video through the same Backbone and Broadband Access Network.
At present there are 0.6 million DataOne broadband customers. The company has vast experience in
Planning, Installation, network integration and Maintenance of Switching & Transmission Networks and
also has a world class ISO 9000 certified Telecom Training Institute. BSNL plans to expand its customer base
from present 73 millions lines to 125 million lines and infrastructure investment plan to the tune of Rs. 733
crores (US$ 16.67 million) in the next three years. Today, BSNL is India's largest Telco and one of the largest
Public Sector Undertaking with estimated market value of $ 100 Billion.
INTRODUCTION
The telephone is a telecommunication device that is used to transmit and receive electronically or
digitally encoded speech between two or more people conversing. It is one of the most common household
appliances in the world today. Most telephones operate through transmission of electric signals over a
complex telephone network which allows almost any phone user to communicate with almost any other
user.
Telecommunication networks carry information signals among entities, which are geographically far
apart. An entity may be a computer or human being, a facsimile machine, a teleprinter, a data terminal and
so on. The entities are involved in the process of information transfer that may be in the form of a telephone
conversation (telephony) or a file transfer between two computers or message transfer between two
terminals etc. With the rapidly growing traffic and untargeted growth of cyberspace, telecommunication
becomes a fabric of our life. The future challenges are enormous as we anticipate rapid growth items of new
services and number of users. What comes with the challenge is a genuine need for more advanced
methodology supporting analysis and design of telecommunication architectures. Telecommunication has
evaluated and growth at an explosive rate in recent years and will undoubtedly continue to do so. The
communication switching system enables the universal connectivity. The universal connectivity is realized
when any entity in one part of the world can communicate with any other entity in another part of the world.
In many ways telecommunication will acts as a substitute for the increasingly expensive physical
transportation.
The telecommunication links and switching were mainly designed for voice communication. With the
appropriate attachments/equipments, they can be used to transmit data. A modern society, therefore needs
new facilities including very high bandwidth switched data networks, and large communication satellites with
small, cheap earth antennas.
SECTIONS VISITED
BROADBAND
SWITCHING
TELEPHONE EXCHANGE
TRANSMISSION SECTION
PULSE CODE MODULATION
MOBILE COMMUNICATION
BROADBAND
Broadband in data communications can refer to broadband networks or broadband Internet and may have the
same meaning as above, so that data transmission over a fiber optic cable would be referred to as broadband
as compared to a telephone modem operating at 56,000 bits per second.
However, broadband in data communications is frequently used in a more technical sense to refer to data
transmission where multiple pieces of data are sent simultaneously to increase the effective rate of
transmission, regardless of actual data rate. In network engineering this term is used for methods where two or
more signals share a medium.
Broadband in analog video distribution is traditionally used to refer to systems such as cable television, where
the individual channels are modulated on carriers at fixed frequencies. In this context, baseband is the term's
antonym, referring to a single channel of analog video, typically in composite form with an audio subcarrier.
The act of demodulating converts broadband video to baseband video.However, broadband video in the context
of streaming Internet video has come to mean video files that have bitrates high enough to require broadband
Internet access in order to view them.The various forms of Digital Subscriber Line (DSL) services are
broadband in the sense that digital information is sent over a high-bandwidth channel above the baseband voice
channel on a single pair of wires. A baseband transmission sends one type of signal using a medium's full
bandwidth, as in 100BASE-T Ethernet. Ethernet, however, is the common interface to broadband modems
such as DSL data links, and has a high data rate itself, so is sometimes referred to as broadband. Ethernet
provisioned over cable modem is a common alternative to DSL
BSNL is in the process of commissioning of a world class, multi-gigabit, multi-protocol, convergent IP
infrastructure through National Internet Backbone-II (NIB-II), that will provide convergent services through
the same backbone and broadband access network. The Broadband service will be available on DSL
technology (on the same copper cable that is used for connecting telephone), on a countrywide basis spanning
198 cities.
In terms of infrastructure for broadband services NIB-II would put India at par with more advanced nations.
The services that would be supported includes always-on broadband access to the Internet for residential and
business customers, Content based services, Video multicasting, Video-on-demand and Interactive gaming,
Audio and Video conferencing, IP Telephony, Distance learning, Messaging: plain and feature rich, Multi-site
MPLS VPNs with Quality of Service (QoS) guarantees. The subscribe will be able to access the above services
through subscribers service selection system (SSSS) portal.
OBJECTIVES
To provide high speed Internet connectivity (upto 8 Mbps)
To provide Virtual Private Network (VPN) service to the broadband customers
To provide dial VPN service to MPLS VPN customers.
To provide multicast video services, video-on-demand, etc. through the Broadband Remote Access
Server (BRAS).
To provide a means to bill for the aforesaid services by either time-based or volume-based billing. It
shall provide the customer with the option to select the services through web server
To provide both pre-paid and post paid broadband services.
TECHNICAL CAPABILITY
The Broadband Service will be given through the state of the art Multi Protocol Label Switching (MPLS)
based IP Infrastructure, which is designed to provide reliable routes to cover all possible destinations within
and outside the country. Layer 1 of the network will consist of a high speed Backbone comprising of 24
powerful Core Routers connected with high speed 2.5 Gbps (STM-16) links. The routers are located on the
national DWDM network interfacing at STM-16 optical level to provide for high transmission speeds.
Advantage of MPLS over other Technologies
MPLS VPN is a technology that allows a service provider like BSNL to have complete control over
parameters that are critical to offering its customers service guarantees with regard to bandwidth
throughputs, latencies and availability.
Services available through Broadband
High speed Internet Access: This is the always-on Internet access service with speed ranging from
256 kbps to 8 Mbps.
Bandwidth on Demand: This will facilitate customer to change bandwidth as per his / her
requirement. For example a customer with 256 kbps can change to 1 Mbps during the video
Conferencing session.
Multicasting: This is to provide video multicast services for application in distance education,
telemedicine etc
Dial VPN Service: This service allows remote users to access their private network securely over the
NIB-II infrastructure.
Video and Audio Conferencing:
Content based Services: Like Video on Demand, Interactive Gaming, Live and time shifted TV
SWITCHING
Switching system fundamentals :
Telecommunications switching systems generally perform three basic functions: they transmit
signals over the connection or over separate channels to convey the identity of the called (and sometimes
the calling) address (for example, the telephone number), and alert (ring) the called station; they establish
connections through a switching network for conversational use during the entire call; and they process
the signal information to control and supervise the establishment and disconnection of the switching
network connection. In some data or message switching when real-time communication is not needed, the
switching network is replaced by a temporary memory for the storage of messages. This type of switching
is known as store-and-forward switching.
Signaling and control :
The control of circuit switching systems is accomplished remotely by a specific form of data
communication known as signaling. Switching systems are connected with one another by
telecommunication channels known as trunks. They are connected with the served stations or terminals by
lines. In some switching systems the signals for a call directly control the switching devices over the same
path for which transmission is established. For most modern switching systems the signals for identifying or
addressing the called station are received by a central control that processes calls on a time-shared basis.
Central controls receive and interpret signals, select and establish communication paths, and prepare signals
for transmission. These signals include addresses for use at succeeding nodes or for alerting (ringing) the
called station.
Most electronic controls are designed to process calls not only by complex logic but also by logic
tables or a program of instructions stored in bulk electronic memory. The tabular technique is known as
translator. The electronic memory is now the most accepted technique and is known as stored program
control (SPC). Either type of control may be distributed among the switching devices rather than residing
centrally. Microprocessors on integrated circuit chips are a popular form of distributed stored program
control.
Switching fabrics :
Space and time division are the two basic techniques used in establishing connections. When an
individual conductor path is established through a switch for the duration of a call, the system is known as
space division. When the transmitted speech signals are sampled and the samples multiplexed in time so that
high-speed electronic devices may be used simultaneously by several calls, the switch is known as time
division.
In the early stages of development in telecommunication, manual switching methods were deployed.
But later on to overcome the limitations of manual switching; automatic exchanges, having Electro-
mechanical components, were developed. Strowger exchange, the first automatic exchange having direct
control feature, appeared in 1892 in La Porte (Indiana). Though it improved upon the performance of a
manual exchange it still had a number of disadvantages, viz., a large number of mechanical parts, limited
availability, inflexibility, bulky in size etc. As a result of further research and development, Crossbar
exchanges, having an indirect control system, appeared in 1926 in Sweden.
The Crossbar exchange improved upon many short- comings of the Strowger system. However, much
more improvement was expected and the revolutionary change in field of electronics provided it. A large
number of moving parts in Register, marker, Translator, etc., were replaced en-block by a single computer.
This made the exchange smaller in size, volume and weight, faster and reliable, highly flexible, noise-free,
easily manageable with no preventive maintenance etc.
Implementation of Switching Network.
In an electronic exchange, the switching network is one of the largest sub-system in terms of size of
the equipment. Its main functions are Switching (setting up temporary connection between two or more
exchange terminations), Transmission of speech and signals between these terminations, with reliable
accuracy.
There are two types of electronic switching system. viz. Space division and Time Division.
Space Division switching System
In a space Division Switching system, a continuous physical path is set up between input and output
terminations. This path is separate for each connection and is held for the entire duration of the call. Path
for different connections is independent of each other. Once a continuous path has been established., Signals
are interchanged between the two terminations. Such a switching network can employ either metallic or
electronic cross points. Previously, usage of metallic cross-points using reed relays and all were favored. They
have the advantage of compatibility with the existing line and trunk signaling conditions in the network.
Time Division Switching System
In Time Division Switching, a number of calls share the same path on time division sharing basis. The
path is not separate for each connection, rather, is shared sequentially for a fraction of a time by different
calls. This process is repeated periodically at a suitable high rate. The repetition rate is 8 KHz, i.e. once every
125 microseconds for transmitting speech on telephone network, without any appreciable distortion. These
samples are time multiplexed with staggered samples of other speech channels, to enable sharing of one
path by many calls. The Time Division Switching was initially accomplished by Pulse Amplitude
Modulation (PAM) Switching. However, it still could not overcome the performance limitations of
signal distortion noise, cross-talk etc. With the advent of Pulse Code Modulation (PCM), the PAM signals were
converted into a digital format overcoming the limitations of analog and PAM signals. PCM signals are
suitable for both transmission and switching. The PCM switching is popularly called Digital Switching.
Digital Switching Systems
A Digital switching system, in general, is one in which signals are switched in digital form. These
signals may represent speech or data. The digital signals of several speech samples are time multiplexed on
a common media before being switched through the system.
To connect any two subscribers, it is necessary to interconnect the time-slots of the two speech
samples, which may be on same or different PCM highways. The digitalized speech samples are switched in
two modes, viz., Time Switching and Space Switching. This Time Division Multiplex Digital Switching System
is popularly known as Digital Switching System.
The ESS No.1 system was the first fully electronic switching system but not digital. But later came ESS
No.4 system which was digital for trunk portion only. When designed, the cost of A/D conversion (CODEC) on
each subscriber line was seen as prohibitive. So the ESS No.4 system was acting as a Trunk/Tandem exchange
but not as a local exchange. So the main difficulty for implementing a digital local exchange was the
implementation of the subscriber line interface. This was solved by the introduction of Integrated Circuits,
which made the digital local exchange economically feasible.
General architecture of Digital Switching System
The next evolutionary step was to move the PCM codec from the exchange end of the
customers line to the customers end. This provides digital transmission over the customers line, which can
have a number of advantages. Consider data transmission. If there is an analog customers line, a modem
must be added and data can only be transmitted at relatively slow speeds. If the line is digital, data can be
transmitted by removing the codec (instead of adding a modem). Moreover, data can be transmitted at 64
kbit/s instead of at, say, 2.4 kbit/s. Indeed, any form of digital signal can be transmitted whose rate does not
exceed 64 kbit/s. This can include high-speed fax, in addition to speech and data.
This concept had led to the evolution of Integrated services digital network (ISDN), in which
the customers terminal equipment and the local digital exchange can be used to provide many different
services, all using 64 kbit/s digital streams. In simple terms, we can say ISDN provides end-to-end digital
connectivity.
DEVELOPMENT OF EXCHANGES
Subs interface
Other
exchanges
N x 2
Mbps
links
CONTROL
PROCESSOR
Operation &
Maintenance
Trunks interface
Other auxiliary inter faces
Such as,
(a) Tone generator
(b) Frequency receives
(c) Conference call facility
(d) CCS# 7 Protocol Manager
(e) V 5.2 access manager
Digital Switch
Local and trunk Network
Trunk Lines :
The termTrunk Line in telecommunications refers to the high-speed connection between telephone
central offices in the Public Switched Telephone Network (PSTN). Trunk lines are always digital. The wiring
between central offices was originally just pairs of twisted copper wire (the twists in the wiring prevented
things known as crosstalk and noise). Because it is expensive to string up (or lay trenches for buried cables),
the phone company researched ways in which to carry more data over the existing copper lines. This was
achieved by using time-division multiplexing. Later, when fiber-optic technology became available, phone
companies upgraded their trunk lines to fiber optics and used statistical time-division multiplexing,
synchronous digital heirarchy, coarse or dense wave division multiplexing and optical switching to further
improve transmission speeds.
The signaling information exchanged between different exchanges via inter exchange trunks for the
routing of calls is termed as Inter exchange Signaling. Earlier in band /out of band frequencies were used for
transmitting signaling information. Later on, with the emergence of PCM systems, it was possible to
segregate the signaling from the speech channel. A trunk line is a circuit connecting telephone switchboards
(or other switching equipment), as distinguished from local loop circuit which extends from telephone
exchange switching equipment to individual telephones or information origination/termination equipment.
A Typical Telephone Exchange -OCB-283
FUNCTIONAL ARCHITURE
The Alcatel E10 system is located at the heart of the telecommunication networks concerned. It is made up
of three independent functional units:
- The Subscriber Access Subsystem which carries out connection of analogue and digital subscriber lines,
- Connection and Control which carries out connections and processing of calls,
- Operation and Maintenance which is responsible for all functions needed by the network operating
authority.
Each functional unit is equipped with softwares which are appropriate for handling the functions for which it
is responsible.
Time base (BT)
The BT ensures times distribution for LR and PCM to provide the synchronization, and also for working out
the exchange clock.Time distribution is tripled.
Time generation can be either autonomous or slaved to an external rhythm with a view to synchronise the
system with the network.
Auxiliary Equipment Control Station SMA
Auxiliary equipment manager (ETA)
The ETA Supports:
- The tone generators (GT).
- The frequency receiving and generation (RGF) devices,
- Conference circuits (CCF),
- The exchange clock
CCS7 protocol handler (PUPE) and CCS7 controller (PC): CCITT No. 7 protocol processing
For connection of 64 kbit/s signaling channels, semi- permanent connections are established via the
connection matrix, to the PUPE which processes the CCITT No. 7 protocol.
More precisely, the PUPE function carries out the following:
signaling channel Level 2 processing,
the message routing function
(Part of Level 3). The PC carries out:
the network management function (part of Level 3),
PUPE defence,
Various observation tasks which are not directly linked to CCITT No. 7.
TRANSMISSION SECTION
The call is transmitted from CALLER to RECEIVER via following pathway:
SUB
CRIB
ER
CONN
ECTIO
N
OPERAT
ION
DATA
TELEPH
VALUE
ADDED
CCITT N
0
7
N
P

OPERAT
ION
Caller
/
D.P
/
Pillar
/
MDF
/
Exchange
/
Tax
/
Exchange
/
MDF
\
Pillar
\
D.P
\
Receiver
There are two types of media to carry out above functionality:
Guided Media (OFC)
Un-Guided Media.
OPTICAL FIBRE
An optical fiber (or optical fibre) is a flexible, transparent fiber made of glass (silica) or plastic, slightly
thicker than a human hair. It can function as a waveguide, or light pipe, to transmit light between
the two ends of the fiber. The field of applied science and engineering concerned with the design and
application of optical fibers is known as fiber optics. Optical fibers are widely used in fiber-optic
communications, which permits transmission over longer distances and at higher bandwidths (data
rates) than other forms of communication.
Fig: Parts of OF
Fiber 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 Coverted into light Signals.
(3) Light Travels Down the Fiber.
(4) A Detector Changes the Light Signals into Electrical Signals.
(5) Electrical Signals are Decoded into Information.
- Inexpensive light sources available.
- Repeater spacing increases along with operating speeds because low loss fibres are used at high data rates.
Optical fibres come in two types: Single-mode fibres and Multi-mode fibres.
Single-mode fibres have small cores (about 9 microns in diameter) and transmit infrared laser light.
Multi-mode fibres have larger cores (62.5 microns in diameter) and transmit infrared light from LEDs.
Optical fibres are made up of three components:
Core - Thin glass centre of the fibre where the light travels
Cladding - Outer optical material surrounding the core that reflects the light back into the core.
Buffer coating - Plastic coating that protects the fibre from damage and moisture.
The advantages of optical fibres over conventional copper wires are:
Less expensive - Several miles of optical cable can be made cheaper than equivalent lengths of copper wire.
Higher carrying capacity - Because optical fibres are thinner than copper wires, more fibres can be bundled
into a given-diameter cable than copper wires. This allows more phone lines to go over the same cable or
more channels to come through the cable into your cable TV box.
Less signal degradation - The loss of signal in optical fibre is less than in copper wire.
Light signals - Unlike electrical signals in copper wires, light signals from one fibre do not interfere with those
of other fibres in the same cable. This means clearer phone conversations or TV reception.
Digital signals - Optical fibres are ideally suited for carrying digital information, which is especially useful in
computer networks.
Non-flammable - Because no electricity is passed through optical fibres, there is no fire hazard.
Lightweight and thin - An optical cable weighs less than a comparable copper wire cable. Optical fibres can be drawn to
smaller diameters than copper wire. Fibre-optic cables take up less space in the ground.
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
Quantisation
Encoding
Line Coding
FILTERING
Filters are used to limit the speech signal to the frequency band 300-3400 Hz.
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 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. : Sampling Process
Sampling Theorem
A complex signal such as human speech has a wide range of frequency components with the
amplitude of the signal being different at different frequencies. To put it in a different way, a complex signal
will have certain amplitudes for all frequency components of which the signal is made. Let us say that these
frequency components occupy a certain bandwidth B. If a signal does not have any value beyond this
bandwidth B, then it is said to be band limited. The extent of B is determined by the highest frequency
components of the signal.
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 or Ts = 125 micro seconds
8000
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.
FIG. 4: Sampling and combining Channels
Fig. 4 shows how a number of channels can be sampled and combined. The channel gates (a, b
... n) correspond to the switch S in Fig. 3. These gates are opened by a series of pulses called "Clock pulses".
These are called gates because, when closed these actually connect the channels to the transmission medium
during the clock period and isolate them during the OFF periods of the clock pulses. The clock pulses are
staggered so that only one pair of gates is open at any given instant and, therefore, only one channel is
connected to the transmission medium. The time intervals during which the common transmission medium
is allocated to a particular channel is called the Time Slot for that channel. The width of.this time slot will
depend, as stated above, upon the number of channels to be combined and the clock pulse frequency i.e. the
sampling frequency.
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 signalling of all the 30 chls, 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 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
i
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
In FDM systems we convey the speech signals in their analogue electrical form. But in PCM, we
convey the speech in discrete form. The sampler selects a number of points on the analogue speech signal (by
sampling process) and measures their instant values. The output of the sampler is a PAM signal as shown in Fig.
3; The transmission of PAMsignal will require linear amplifiers at trans and receive ends to recover distortion
less signals. This type of transmission is susceptible to all thedisadvantages 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 quantified 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 "Quantising". Of course, the scales and the number of points should be so chosen that
the signal could be effectively reconstructed after demodulation.
Quantising, 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.
A sampled signal exists only at discrete times but its amplitude is drawn from a continuous range of
amplitudes of an analogue signal. On this basis, an infinite number of amplitude values is possible. A
suitable finite number of discrete values can be used to get an. approximation of the infinite set. The
discrete value of a sample is measured by comparing it with a scale having a finite number of
intervals and identifying the interval in which the sample falls. The finite number of amplitude intervals is
called the "quantizing interval". Thus, quantizing means to divide the analogue signal's total amplitude
range into a number of quantizing intervals and assigning a level to each. intervals.
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 cal led Linear Quantizing.
FIG. 7: QUANTIZING-POSITIVE SIGNAL
FIG. 8: QUANTIZING - SIGNAL WITH + Ve & - Ve VALUES
Relation between Binary Codes and Number of levels.
Because the quantized samples are coded in binary form, the quantization intervals will be in powers
of 2. If we have a 4 bit code, then we can have 2" = 16 levels. Practical PCM systems use an eight bit code with
the first bit as sign bit. It means we can have 2" = 256 (128 levels in the positive direction and 128 levels in
the negative direction) intervals for quantizing.
Encoding
Conversion of quantised 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 1 Segment Code Linear encoding
for + ve 'O' for - ve. 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.
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.
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
colinear 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).
There are 128 quantification levels in the positive part of the curve and 128 in the negative part of the
curve. In a PCM system the channels are sampled one by one by applying the sampling pulses to the sampling
gates. Refer Fig. 10. The gates open only when a pulse is applied to them and pass the analogue signals through
them for the duration for which the gates remain open. Since only one gate will be activated at a given instant,
a common encoding circuit is used for all channels. Here the samples are quantized and encoded. The encoded
samples of all the channels and signals etc are combined in the digital combiner and transmitted.
MOBILE COMMUNICATION
1946- 1960s 1980s 1990s 2000s
Appeared 1G 2G 3G
Analog Digital Digital
Multi Multi Unified
Standard Standard Standard
Terrestrial Terrestrial Terr. & Sat
WIRELESS GENERATIONS
1 G -analog (cellular revolution) only mobile voice services
2 G - digital (breaking digital barrier) -mostly for voice services & data delivery possible
3 G - Voice & data (breaking data barrier) Mainly for data services where voice services will also be possible
LIMITATIONS OF 2nd GENERATION SYSTEMS
No Global standards
No common frequency band
Low information bit rates
Low voice quality
No support of Video
Various categories of systems to meet specific requirements
THIRD GENERATION (3 G ) STANDARD
International mobile telecom 2000. imt-2000
ITUs vision for third generation mobile system
a future standard in which a single inexpensive mobile terminal can truly provide communications
any time and any where
Provisioning of these services over wide range of user densities and coverage areas.(in-building , urban
, sub-urban, global)
Efficient use of radio spectrum consistent with providing service at acceptable costly.
IMT-2000 shall cover application areas presently provided by seperately systems i.e cellular, cordless and
paging etc.
A high degree of commonality of design worldwide.
A modular structure which will allow the system to grow in size and complexity.
Single unified standard (data & multimedia services)
Anywhere, anytime communication
Across networks, across technologies, seamless operation using a small pocket terminal worldwide.
High speed access 144kb/s, 384 kb/s & 2mb/s fast wireless access to internet
Full motion videophone
Terrestrial & satellite components
Enhanced voice quality, ubiquitous coverage and enable operators to provide service at reasonable cost
Increased network efficiency and capacity
New voice and data services and capabilities
An orderly evolution path from 2G to 3G systems to protect investments.
IMT TECHNOLOGIES
ITU has finally narrowed down technology options to the following five:
1. IMT -DS (Direct Spread) : W-CDMA UTRA FDD
2. IMT -MC (Multi Carrier) : CDMA 2000
3. IMT-TC ( Time Code) : TD -SCDMA UTRA TDD
4. IMT -SC ( Single Carrier ) : UWC - 136
5. IMT-FT (Frequency Time) : DECT
IMT-DS IMT-MC IMT-TC IMT-SC IMT-FT
WCDMA CDMA2000
1X/3X
TDMA CDMA FDMA
CDMA-TDD UWC-136 FDMA/TDMA
DECT
IMT-2000 TERRESTRIAL
RADIO INTERFACES
Migration Path
While a multiplicity of 2G standards have been developed and deployed, the ITU wanted to avoid a similar
situation to develop for 3G.
Hence, the ITU Radio communication Sector (ITU-R) has elaborated on a framework for a global set of 3G
standards, which will facilitate global roaming by operating in a common core spectrum and providing
migration path from all the major existing 2G technologies.
The major 2G Radio access networks are based on either CDMA One or GSM technologies and different
migration path is proposed for each of these technologies.
Evolution from GSM to 3G
GSM Evolution
GSM
2G
HSCSD
GPRS
2.5G
EDGE
3G
GPRS
200 KHz carrier
115 Kbps peak data rates
EDGE
200 KHz carrier
Data rates up to 384 Kbps
8-PSK modulation
Higher symbol rate
UMTS
5 MHz carrier
2 Mbps peak data rates
New IMT-2000 2 GHz spectrum
GSM
200 KHz carrier
8 full-rate time slots
16 half-rate time slots
GSM GPRS EDGE UMTS
3G 2.5G 2G
HSCSD
HSCSD
Circuit-switched data
64 Kbps peak data rates
Architecture of the GSM Network
The GSM technical specifications define the different entities that form the GSM network by defining their
functions and interface requirements. The GSM network can be divided into four main parts:
Fig : 1 Architecture of the GSM network
The Base Station Subsystem
The BSS connects the Mobile Station and the NSS. It is in charge of the transmission and reception. The
BSS can be divided into two parts:
The Base Transceiver Station
The BTS corresponds to the transceivers and antennas used in each cell of the network. A BTS is usually
placed in the center of a cell. Its transmitting power defines the size of a cell. Each BTS has between one and
sixteen transceivers depending on the density of users in the cell.
Functions of BSC
1. It controls a group of BTS and manages their radio resources.
2. It perform all process of handovers.
3. It also control of the radio frequency power levels of the BTSs.
OMC
MSC
BSC HLR
A
MS
Other
MSCs
BTS AUC
Other
Networks
EIR
Other
MSCs
VLRs
VLR
BSS
B
C
D
E
F
G
Un
Abis
BTS Structural Dimensions and Arrangement
EDGE (Enhanced Data for GSM Evolution)
Next step towards 3G for GSM/GPRS Networks
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
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.
CDMA
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 :
1. Multi Access Radio Relay
2. Wireless In Local Loop
3. Fibre In the Local Loop
Wireless in Local Loop (WILL)
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 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.
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.
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 Mehtod: 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.
Different Codes used in CDMA
Walsh Code:
In CDMA the traffic channels are separated by unique Walsh code. All such codes are orthogonal to each
other. The individual subscriber can start communication using one of these codes. These codes are traffic
channel codes and are used for orthogonal spreading of the information in the entire bandwidth.
Orthogonality provides nearly perfect isolation between the multiple signals transmitted by the base station.
The basic concept behind creation of the code is as follows:
(a) Repeat the function right
(b) Repeat the function below
(c) Invert function (diagonally)
Long Code:
The long pseudo random noise (PN) sequence is based on 2
42
characteristic polynomial. With this long code
the data in the forward direction (Base to Mobile) is scrambled. The PN codes are generated using linear shift
registers. The long code is unique for the subscribers and is known as users address mask.
Short Code:
The short pseudo random noise (PN) sequence is based on 2
15
characteristic polynomial. This short code
differentiates the cells & the sectors in a cell. It also consists of codes for I & Q channel feeding the modulator.
Advantages: CDMA wireless access provides the following unique advantages.
Larger Capacity:
Let us discuss this issue with the help of Shannons Theorem. It states that the channel capacity is related to
product of available band width and S/N ratio.
C = W log 2 (1+S/N)
Where C = channel capacity
W = Band width available
S/N = Signal to noise ratio.
It is clear that even if we improve S/N to a great extent the advantage that we are expected to get in terms
of channel capacity will not be proportionally increased. But instead if we increase the bandwidth (W), we
can achieve more channel capacity even at a lower S/N. That forms the basis of CDMA approach, wherein
increased channel capacity is obtained by increasing both W & S/N. The S/N can be increased by devising
proper power control methods.
Introduction to CDMA 2000-1X
Base station subsystem (BSS) Base station subsystem is the general term for the wireless devices
and wireless channel control devices that serve one or several cells. Generally, a BSS contains
one more base station controllers (BSC) and base transmitter stations (BTS).
RGMTTC Presentation
Um E
Abis A
Q C
B
N
H
MS
BTS BSC
PSTN MSC
MSC/SSP VLR
HLR
AUC
MC
D
MC
M
MSS
BSS
Ai
Architecture of CDMA MSC Based WLL system
Mobile switch center (MSC)
MSC is a functional entity that performs control and switching to the mobile stations within the
area that it serves, and an automatic connecting device for the subscriber traffic between the
CDMA network and other public networks or other MSCs. MSC is the kernel of the CDMA cellular
mobile communication system, and it is different from a wired switch in that an MSC must
consider the allocation of the wireless resources and the mobility of subscribers, and at least it
must implement the follows processing activities:
1. Location Registration processing;
2. Handoff.
Gateway MSC (GMSC)
When a non-CDMA subscriber calls a CDMA subscriber, the call will first be routed to an MSC,
which will inquires the corresponding HLR and further route the call to the called partys MSC.
This kind of MSC is called Gateway MSC (GMSC). It is up to the network operator to select
which MSCs as GMSCs.
Visitor location register (VLR)
VLR is responsible for the storage and updating of the subscriber data of mobile stations that
roamed to the service area of this VLR. The VLR is generally configured together with the MSC.
When the mobile station enters a new location area, the MSC will notice the VLR, which will
initiate registration processing to the HLR to update the subscriber location information. The
VLR also stores necessary information for the establishment of calls in the database for the
MSC to search. One VLR can cover one or more MSC areas.
Home location register (HLR)
The HLR provides subscriber information storage and management functions for the mobile
network, including mobile subscriber subscription and cancellation and service authorization
and cancellation. At the same time, it helps in the implementation of subscribers call and
service operations. A CDMA can contain one or more HLRs based on the number of
subscribers, equipment capacity and network organization mode, with multi-HLR mode
realized in the form of virtual HLRs. The subscriber information stored in the HLR includes the
following two types in information:
1. Subscription information
2. Subscriber-related information stored in the HLR
Authentication center (AUC)
Authentication center is a function entity for the management of authentication information
related to the mobile station. It implement mobile subscriber authentication, stores the
mobile subscriber authentication parameters, and is able to generate and transmit the
corresponding authentication parameters based on the request from MSC/VLR. The
authentication parameters in the AUC can be stored in the encrypted form. The
authentication center is generally configured together with the HLR. The authentication
parameter stored in the AUC include:
1. Authentication key (A_KEY);
2. Share secret data (SSD);
3. Mobile identification number/international mobile subscriber identity (MIN/IMSI);
4. Authentication algorithm (AAV);
5. Accounting (COUNT).
Short message center (MC or SC)
As an independent entity in the CDMA cellular mobile communication system, the short
message center works in coordination with other entities such as MSC, HLR to implement the
reception, storing and transfer of the short messages from CDMA cellular mobile
communication system subscribers, and store subscriber-related short message data.
Short message entity (SME)
SME is a function entity for synthesis and analysis of short messages.
Operation and maintenance Center (OMC)
The OMC provides the network operator with network operation and maintenance services,
manages the subscriber information and implements network planning, to enhance the
overall working efficiency and service quality of the system. There two type of operation and
maintenance centers: OMC-S and OMC-R. An OMC-S is mainly used for the maintenance work
at the mobile switching subsystem (MSS) side; an OMC-R is mainly used for the maintenance
work at the base station subsystem (BSS) side.