Network Planning To Students

Menoufia University
Faculty of Electronic Engineering

Dept. of Electronics & Electrical Communications
Telecommunication Network Planning
4th Year
Prof. Sami A. El-Dolil

‫جامعة المنوفية‬
‫كلية الهندسة اإللكترونية‬
‫قسم هندسة االلكترونيات واالتصاالت الكهربية‬
Department offering the program: Electronics and Electrical Communications Engineering

Department offering the course: Electronics and Electrical Communications Engineering
Course Specification
1- Course basic information :
Course Code: ECE 424 Course Title: Academic Year: 2022-2023

nd
Network Planning Level (4) – Semester : 2
Department requirement Teaching hours: Lecture [2] Tutorial [1] Lab [0]
1. Describe, explain, analyze, and evaluate the performance of common

telecommunication networks including PSTN.
2. Identify, explain, and analyze different forecasting techniques using
both linear and nonlinear forecasting model.
3. Demonstrate different network topologies and their properties and
features.
4. Illustrate the applications of mathematical background to develop and
2- Aim of the course evaluate different network switching techniques including circuit
switching, message switching, and packet switching.
5. Define, formulate, solve different teletraffic models as well as identify
their suitability for different queuing and network applications.
6. Use software tools to design, implement, and evaluate teleraffic
networks and their performance.
7. Develop the skills related to creative thinking, problem solving, and
teamwork in different fields of networking.
3- Course related program competencies
A.1 Identify, formulate, and solve complex engineering problems by applying
Level A - General
engineering fundamentals, basic science and mathematics.

A.3 Apply engineering design processes to produce cost-effective solutions that meet
specified needs with consideration for global, cultural, social, economic, environmental,
ethical and other aspects as appropriate to the discipline and within the principles and
contexts of sustainable design and development.
A.10 Acquire and apply new knowledge, and practice self, lifelong and other learning
strategies.
B.2 Design, model and analyze an electrical/electronic/digital system or component for
a specific application; and identify the tools required to optimize this design.
Speciality
Level B -
B.3 Design and implement elements, modules, sub-systems, or systems in

electrical/electronic/digital engineering using technological and professional tools.
B.5 Adopt suitable national and international standards and codes to design, build,
operate, inspect and maintain electrical/electronic/digital equipment, systems and
services.
Electronics and Electrical Communication Engineering Program 4th Year - Course Specification
1
C.1 Design, analyze and measure the performance of communication systems: wireless
and wired networks, encoding and decoding systems; microwave systems with
Level C - Sub-
identifying software tools and mathematical methods required to optimize their

Speciality
performance.
C.2 Demonstrate the knowledge about state-of-the-art designs of components and
systems; and develop innovative solutions for practical industrial problems in
electronics and electrical communications engineering.
C.4 Implement and integrate electronic and electrical communication systems for
different engineering applications.
4- Course Goal of network planning - Fundamental plans: Financial plan - Forecasting plan -
Contents Provision timing plan - Technical plans: Switching plan - Numbering plan - Signaling
plan - Capacity plan - Routing plan - Charging plan - Evaluation and development plan -
Transmission plan - Quality of service - Grade of service - Mobile radio network -
Satellite network.
# Topic Lecture Tutorial/Practical No of hours
1. Introductory Topics 4 2 6
2. Fundamental plans 4 2 6
3. Technical plans 4 2 6
4. Capacity plan 4 2 6
5. Fundamentals of Teletraffic
4 2 6
Theory
6. Some Basic Teletraffic
4 2 6
Models
7. Mini-project using
4 2 6
MATLAB
Total sum 28 14 42
5- Teaching and learning 1. Lecture (in class)
methods 2. Discussion
3. Tutorial
4. Problem solving
5. Brain storming
6. Self-learning
7. Research and Reporting
6- Teaching and learning 1. Additional Tutorials
methods for disable students 2. Online lectures and assignments
3. Using as many audio/visual aids as possible
4. Providing extra opportunities/time for practice
7- Teaching and learning 1. Assign a portion of the office hours for those students and
methods for low capacity provide them with specific tailored tasks
students 2. Repeat the explanation of some of the material and tutorials
3. Assign a teaching assistance to follow up their performance
8- Teaching and learning 1. Assign course project tasks to those students
methods for outstanding 2. Give them advanced extra-curriculum topics
students 3. Encourage them to take part in a pilot research and case
studies
2
9- Students assessment
a- Assessment 1. Mid Term Examination (written)
methods 2. Formative (quizzes- presentation -reports)
3. Final Term Examination (written)
b- Assessment - Exercise sheet/ Lab assignment : Weekly
schedule - Quizz-1: Week no 5
- Mid-Term exam: Week no 8
- Quizz-2: Week no 12
- Final – term examination: Week no 16
c- Weighting of - Class tutorial and quizzes : 10 %
assessment - Mid-term examination: 20 %
- Final – term examination: 70 %
Total 100 %
10- List of text books and references:

a- Course notes There are lectures notes prepared.
b- Text books [1] Roger L. Freeman, Fundamentals of Telecommunications, John
nd
Wiley & Sons, Inc., 2 ed. 2005.
c- Recommended [1] Annabel Z. Dodd, The Essential Guide to Telecommunications,
th
books 5 ed., 2012.
d- Periodicals, Web  https://www.comsoc.org/publications/magazines/ieee-communications-
sites ………etc magazine
 https://www.comsoc.org/
11-Course contents – Course related program competencies
Level A Level B Level C

A.1 A.3 A.10 B.2 B.3 B.5 C.1 C.2 C.4
Introductory Topics   
Fundamental plans
       
Technical plans 
       
Capacity plan
       
Fundamentals of Teletraffic Theory
       
Some Basic Teletraffic Models
       
Mini-project using MATLAB  
    
12-Teaching and learning methods - Course related program competencies
Level A Level B Level C
3
A.1 A.3 A.10 B.2 B.3 B.5 C.1 C.2 C.4
Lecture (in class)      
Discussion    
Tutorial     
Problem solving     
Brain storming      
Self-learning    
Research and Reporting     
13- Assessment methods - Course related program competencies
Course related program comptencies

Assessment methods Level A Level B Level C
A.1 A.3 A.10 B.2 B.3 B.5 C.1 C.2 C.4
1. Mid Term Examination (written)     
2. Formative (quizzes- presentation -
   
reports)
3. Final Term Examination (written)       
Authorized from department board at:

Authorized from college board at:
Course coordinator: Head of Department:
Prof. Sami A. Eldolil Prof. Osama Zahran
4
Contents
Preface
Chapter One Introduction
Chapter Two Fundamental Plan
2.1 Financial Plan
2.1-1 Main Financial Plan
2.1-2 Medium Financial Plan
2.1-3 Micro Financial Plan
2.2 Forecasting Plan

2.2-1 Forecasting of Subscriber Density
2.2-1.1 Linear Model
2.2-1.2 Exponential Model
2.2-2 Forecasting of Traffic per Line
2.3 Provision Timing Plan

Chapter Three Technical Plan
3.1 Switching Plan
3.1-1 Network Topologies
3.1-2 Hierarchical Network
3.1-3 Switching Techniques
3.1-3.1 Circuit Switching
3.1-3.2 Message Switching
3.1-3.3 Packet Switching
3.2 Numbering Plan

3.2-1 Constraints on Numbering Plan
3.2-2 International Number
3.3 Signaling Plan

3.3-1 Functions of the Signaling System
3.3-1.1 Supervisory Signaling
3.3-1.2 Address signaling
3.3-1.3 Call Forwarding
3.3-2 Signaling Techniques
3.3-2.1 In-Channel Signaling
3.3-2.2 Common Channel Signaling
3.4 Capacity Plan

3.4-1 Loss System with Infinite Number of Users
3.4-2 Loss System with Finite Number of Users
3.4-3 Delay System
3.4-3.1 Multi-channels delay system
3.4-3.2 Single-channels delay System
Contents
3.5 Routing Plan

3.5-1 Routing Structure
3.5-2 Routing Schemes
3.5-2.1 Fixed Routing Scheme
3.5-2.2 Dynamic Routing Scheme
3.5-3 Route Selection
3.5-4 Routing Applications
3.5-4.1 Automatic Alternative Routing
3.5-4.2 Automatic Rerouting
3.5-4.3 Load Sharing
3.6 Charging Plan

3.6-1 Local Call Charging
3.6-1.1 Message Rate Charging Scheme
3.6-1.2 Flat-rate Tariff Scheme
3.6-1.3 Hybrid Scheme
3.6-2 Trunk Call Charging
3.6-3 International Charging
ii
Preface
Telecommunications (long distance communications) technology is paving the way to

a world in which distance is no longer a barrier.
It enables us to interact with one another irrespective of our physical locations, stationary or in
motion, bringing people of different countries, cultures, and races closer to one another. It fulfills
the quest of people for obtaining, processing and transferring information as quickly as possible
wired or wireless.
It is one of the fastest growing business sectors of modern information technologies. A couple of
decades ago, to have a basic understanding of telecommunications, it was enough to know how
the telephone network operated. Today, the field of telecommunications encompasses a vast
variety of modern technologies and services. Some services, such as the fixed telephone service
in developed countries, have become mature, and some have been exploding (e.g., cellular mobile
communications and the Internet). The deregulation of the telecommunications industry has
increased business growth, even though, maybe because, tariffs have decreased.
The present telecommunications environment, in which each of us has to make choices, has
become complicated. In the past, there was only one local telephone network operator that we
chose to use or not use. Currently, many operators offer us ADSL or cable modem for Internet
access and we have many options for telephone service as well.
Special attention has to be paid to the telecommunications security aspects and costs of services.
The first purpose of this book is to provide an overall view of telecommunications networks
planning, management and development. This kind of general knowledge is useful to the users of
telecommunications services, the personnel of operators, and the employees of
telecommunications system manufacturers. The distinct separation of the networks that provide
data, speech, and mobile services is required to be clearly investigated.
Network planning is a process that is defined and depends mainly on measurements, and data
collection from the service area. Moreover, the planning process has to be defined carefully and
carried out in different phases in order to manage the strong influences between:
• coverage
• capacity
• quality (interference probability).
These three areas must all be optimized in order to achieve a cost-efficient and overall high
Quality and grade of Service (good speech quality, minimum network congestion).
The aim of this course is to help the student to understand the basis of the structure and operation
of the telecommunication networks, and differentiate between its different types, build his skills,
and put him on the track to be familiar with the different terminologies used in the subject. This is
an essential condition for being able to communicate with specialists, understand the technical
literature, and to be able to dig deeper in the subject later.
Imagine what technical understanding is needed to plan, design, implement and optimize the
service and the network.
ii
At the same time as the telecommunication network has been expanding, the huge progress and
developments in technology have enabled communications to be provided at lower cost so that
expansion has taken place without absorbing a larger share of the world’s resources.
Technological advances have also improved the services that the network can provide. This has
led to increased demands from users and more growth of the network. Further scientific and
technological advances are needed to facilitate the future growth and new services on which
economic and social progress will increasingly depend.
Hopefully this text including something useful, but as all things in our life still something missed.
This book is categorized into four chapters:
Chapter 1: Introduction, introduces an introduction about communication systems and
communication networks. It is terminated by the main requirements that a good planned network
must achieve.
Chapter 2: Fundamental Plan, investigates three sub plans: financial plan, forecasting plan and
provision timing plan with description to their importance to gain the required performance.
Chapter 3: Technical Plan, provides the main technical issues concerned with the
communication network operation to be considered and analyzed to acquire the required
knowledge to assist for network evaluation and development.
Chapter 4: Evaluation and Development Plan,
4.1 Detailed plan
4.1-1 Parameter Planning
4.1-2 Transmission Plan
4.1-3 Quality of Service
4.1-4 Signal Attenuation
4.1-5 Network Noise
4.1-6 Non-linear Distortion
4.1-7 Crosstalk
4.1-8 Echo
iii
Chapter 1
Introduction
The concept of telecommunications (long distance communications) is generally

used to denote the totality of techniques and circuits necessary for optimum transfer of
information (speech, data, graphics, music, image, video,……etc.) from one location
wherever it is to another instantly and efficiently. This must be carried on demand and in
economical manner via a given transmission media (wired or wireless), in the presence of
noise and other forms of transmission impairments. More specifically, it represents the
whole of the technology and procedures used to realize optimum information transfer via
public facilities such as; Public Switched Telephone Network, (PSTN), and the Integrated
Services Digital Network, (ISDN), or private facilities such as Local Area Networks,
(LANs). These networks all have standard interfaces. Traditionally different information
categories have been approached completely differently from a communication point of
view with their own networks, procedures, and switching principles. Fortunately, recently,
there is an increasing tendency towards full integration in many directions such as; several
traffic types to facilitate multimedia communication, and various networks to achieve
highly efficient with less cost communication network, ….etc., on the price of complexity.
Investment in communication equipment in the networks is so large, and the increase in
demand for communications services is so massive, that it is not possible to scrap older
equipment in short term. Thus, although new equipment is being introduced, only rarely
do older systems disappear.
New innovations in radio communication have brought about the use of this technology in
new application areas. The majority of the traffic carried by telecommunication networks
was voice traffic, but other forms of messages such as data, video, or images have been
integrated and transmitted over a single speech channel using suitable terminal equipment.
Network planning is an evolving process, and there is no standard telecommunication
1
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network planning process. Though some of the steps may be common, the process is
determined by criteria and targets, and the type of projects. The process has to be applied
case by case.
Network planning is a complicated process consisting of several phases takes input from
the dimensioning, initial network configuration. The final target for the network planning
process is to define the network design. The network design can be an extension of an
existing network or a new network to be launched. The difficulty in network planning is to
combine all of the requirements in an optimal way and to design a cost-effective network.
Before the actual planning is started for a new network, some key information are defined:
market share, network service areas, services, tariffs, etc. Questions arise about the nature
of the targeted user group, how large is the coverage provided at the beginning and how it
will grow in the future. It is also decided in the beginning what kind of services will be
offered and which is connected to the target user group.
The basic telecommunication system consists of three main parts that are always present
in some form or another as shown in figure 1.1.
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rheIimtppah reIIaC raoatcah
Figure 1.1 Model of a communication system
1- A transmitter that takes information and converts it to a signal.

2- A communication channel works as a transmission medium, also called the physical
medium that carries the signal between the transmitter and the receiver. It may be wired
or wireless, include the atmosphere for sound communications, glass optical fibres for
some kinds of optical comm., coaxial cables for communications by way of the
voltages and electric currents in them, and free space for communication using visible
light, infrared waves, ultraviolet light, and radio waves.
3- A receiver that takes the signal from the channel and converts it back into usable
information for the recipient.
2
Figure 1.2 illustrates the Electromagnetic Spectrum for communication and its
applications.
Figure 1.2 Electromagnetic Spectrum for Telecomm.
1.1 Classification of Communication Systems

Communication systems may be classified based on different criteria and application
to different types such as;
1.1-1 According to the direction of transmission,
Communication systems may be classified to three types as shown in figure 1.3:
rheIimtppah raoatcah
a- Simplex Communication System
rheIimtppahrC raoatcahCrC
raoatcah rheIimtppah
b - Half duplex Communication System
arettissnarT rnvnicnaTrT
rnvnicna arettissna
c- Full duplex Communication System
Figure 1.3 Communication Systems
a- Simplex communication system

Simplex communication is possible only in one direction. There is one powerful
transmitter and one (many) low power but sensitive receiver(s). Radio and television
broadcasting are simplex comm. systems. It is mainly long distance comm. system.
3
b- Half-duplex communication system

Half-duplex communication is possible in both directions between two entities
(computers or persons), but not at the same time. This is because one frequency is used for
the two directions. The Intercom, and the Vax systems use this approach. Both stations at
A, and B are transceivers. The hand set at each location can be switched between transmit
and receive modes. The person who wants to talk presses a talk button on his handset to
start talking, and the other person's handset will be in receiving mode. When the sender
finishes, he terminates it with an over message. The other person can press the talk button
and start talking. These types of systems require limited channel bandwidth, so they are
low cost systems.
c. Full-duplex communication system
In a full-duplex communication system (two-way systems), the two parties-the caller
and the called-can communicate simultaneously, as in the telephone systems (PSTN, and
mobile). This is because two frequencies are used one for each direction. Both stations A,
and B are transceivers.
1.1-2 Based on the number of communicating transceivers, the communication systems
can be of different types as shown in figure 1.4.
c- Multi-Point-to-multi-point
Figure 1.4 Different communication systems
4
a- Point-to-point communication
In this type, communication takes place between two end points. For instance, in the
case of voice communication using telephones, there is one calling person and one called
person. Hence the comm. is point-to-point. It is bi-directional comm. system.
b- Point-to-multipoint communication
In this type of communication, there is one sender and multiple recipients. For example,
in voice conferencing, one person will be talking but many others can listen.
The message from the sender has to be multicast to many others. It is simplex comm.
system. In a radio or TV broadcasting system, there is a central location from which
information is sent to many recipients.
c- Multipoint -to-multipoint communication

In this type of communication, there is multiple sender and multiple recipients. The
Internet is a global network of computers and has become the most popular multipoint -to-
multipoint comm. system.
1.1-3 Based on the distance between the transmitter and the receiver,
there are mainly two system types:
a- Short distance communication system, where the distance between the transmitter
and the receiver is short such that the information base band signal can be transmitted
through it without a carrier to carry it.
Only amplification may be required to compensate its attenuation. )no retnu( xelpud fla
.nlnpul en )l u()ltnu
b- Long distance communication system, where the distance between the transmitter
and the receiver is large such that the information base band signal can't be transmitted
through it without a carrier to carry it.
Amplification is not enough to compensate its attenuation. A carrier with higher frequency
and power is required to carry the information base band signal from the transmitter to the
receiver. Simplex (broadcasting), and full duplex (telephone) systems are common
examples.
5
1.1-4 Depending on the type of the transmitted signal:

Analog (continuous) and digital (discrete) signals as shown in figure 1.5, comm.
systems can be broadly divided into analog comm. systems and digital communication
systems.
Figure 1.5-a Analog Sinusoid Signal
Figure 1.5-b Digital Signal

Many different advantages are gained using digital system such as; Ease of multiplexing,
use of modern technology, signal regeneration, good quality, easiness of encryption and to
store in a memory ..etc.
All these advantages are achieved because two voltage states (high and low) are easier to
be processed than a continuous range of states.
Following the above discussion, we can define that telecommunications in which multiple
transmitters and multiple receivers have been designed to cooperate and to share the same
physical channel are called multiplex systems. The sharing of physical channels using
multiplexing often gives very large reductions in costs, increase of channel efficiency,
easiness of hardware implementation and maintenance of the communication networks.
Multiplexed systems are laid out in telecom. networks, and the multiplexed signals are
switched at nodes through to the correct destination terminal receiver.
A telecommunication network is a collection of transmitters, receivers, and comm.
channels that send messages to one another. Some digital comm. networks contain one or
more routers that work together to transmit information to the correct user.
An analog comm. network consists of one or more switches that establish a connection
between two or more users. For both types of network, repeaters may be necessary to
amplify or recreate the signal when it is being transmitted over long distances. This is to
combat attenuation that can reduce the signal extraction from the noise.
6
There are number of basic network requirements such as:

i- The network must provide the required grade of service in terms of fast on demand
connection with high transmission quality.
ii- Every subscriber connection to the system must be uniquely identified.
iii- The system must indicate to the caller the availability state of the required outlet.
iv- The system must protect each connection from interference from other connections.
v- The system should generate information to allow revenue to be collected for the call.
vi- The system must be well dimensioned in such a way as to be economical.
vii- The system must be capable of expansion.
viii- The system should be able to cater for future improvements and new services.
The Internet is a worldwide network of computers and computer networks that
communicate with each other using the Internet Protocol (IP). Any computer on the
Internet has a unique IP address that can be used by other computers to route information
messages to it. These messages carry with them the originating computer's IP address
allowing for two-way communication.
Depending on the situation of the comm. terminals; stationary or mobile, comm. networks
may be classified to stationary (PSTN) or mobile. Both of these technologies have its
advantages and disadvantages. Also, mobile comm. networks may be divided into
terrestrial and space comm. network.
Communication network is built for economic factors in a hierarchy as shown in figure
1.6. It may be divided into a number of network layers as follows:
a- Local networks, comprising the subscribers' terminals, lines to the local exchange and
the local exchange itself.
b- Junction networks, comprising the junctions between adjacent local exchanges and
between local and trunk exchanges.
c- Trunk networks, which consists of the trunk exchanges and interconnecting
transmission links.
d-The international network, consisting of the gateway exchanges and interconnecting
transmission links to other countries.
For operational purposes, the hierarchy structure has advantage in producing easily
7
controlled units in terms of size and complexity. It also enables each level of the hierarchy
to be developed in a manner best suited to that level, taking into account local and national
conditions.
The levels in the hierarchy can be linked to form the coherent world communication
system.
Local networks are joined up to form national networks, and later these were
interconnected by an international network.
Figure 1.6 A Hierarchy Communication Network
8
The links between the tiers in the hierarchy are the plans (transmission plans, numbering
plans, signaling plans etc.) which form the infrastructure of the system. In the broadest
sense, they enable to achieve the world global system and, within the various tiers, set the
parameters within which the network planner must operate.
9
CHAPTER 2
FUNDAMENTAL PLAN
The purpose of a telecommunications network planning is to ensure that the

expected requirements will be met in an economical way both for subscribers and
operators.
The network preplanning process begins with the acquisition of external information. This
includes:
- The economic information aspects,
- Forecasts of how the new network / service will operate.
Economical information concerning with the allocated budget that has to be defined to
properly estimate the capital costs of expenditure, CAPEX;
To properly dimension the network, we need to estimate the traffic offered load. This is
achieved not only by collecting the information about the current status of the traffic
aspects by making traffic measurements and defining subscriber's density and their
distribution, but also making forecasting about future services scenarios, traffic demands
volumes, traffic per subscriber and population growth profiles.
Forecast periods also have to be defined for reliable networks. So, Traffic forecast defines
the estimated traffic growth in the network over the planning period.
The forecasting process involves several steps as follows:
- Definition of problem,
- Data acquisition,
- Choice of forecasting method,
- Analysis/Forecasting,
- Documentation and analysis of results.
10
CHAPTER 1 FUNDAMENTAL PLAN
This plan involves the Financial, Forecasting, and Provision Timing Plans where planners’
attention is drawn at the outset to the crucial role of economics to plan a cost efficient
network by making selection between number of practical alternatives, economic
comparisons which take account of the timing of expenditure.
2.1 Financial Plan

It is very important aspect in telecommunication network planning, as it deals with
the investment evaluation studies in the telecommunication sector, so it is required to
obtain the correct balance of cost between subscribers’ lines, junction lines, switching
equipment, and buildings, where the choice between alternative schemes should be based
on economic studies. Optimum network is the one designed for efficient performance, and
minimum cost. The solutions actually achieved are usually far from optimum, since the
available Exchanges’ locations, that need to be at the central points of the service areas as
possible, and the distribution of subscribers around them produce constraints. Practical
considerations, such as the availability of sites play as important a part in determining the
layout of a telecommunication network as do considerations of equipment cost. There is an
economic conflict between the number of exchanges (and the cost of the circuits
interconnecting them) and the cost of subscribers’ lines. Usually sparsely populated areas
contain few, small exchanges, even though subscribers’ lines are long, and therefore
expensive. Densely built-up areas have large, or many exchanges and short subscribers’
lines, which requires a large number of circuits between exchanges, and the cost of
providing these and the associated switching equipment may be too great. The principle
which is adopted to produce something like an optimum network is to try to design for
minimum cost. The main function of the technical plan is to select from the various,
available, possible schemes the most economic, that able to achieve the required level of
development and expansion. The economic environment adds another layer of complexity
to the network planner's role. The economic environment affects the resources available
for network operation and expansion, the cost of money, and the money available for
customers to spend on telecommunications. The economic environment varies from
11
country to country, and its future evolution is hard to predict. Naturally the economic
environment is linked to the competitive environment.
Finally, there can be public health concerns such as those recently raised concerning
wireless technology and high frequency electromagnetic fields.
Many of these same issues affect-either directly or indirectly-user organizations, where the
economic environment appears in the form of budgetary concerns. Tran-border
information flow regulations cause concern in multinational organizations. Standards are
important for user organizations desiring multiple vendor sources for critical equipment.
Financial plan can be considered in three levels:
2.1-1 Main Financial Plan

Main financial plan deals with the amount of wealth a country spends in
telecommunication sector. Of course, the stage of development of a particular country and
whether economy is controlled, free-running, or somewhere in between will be strongly
the contributing factors. It is very important for evolving nations to achieve a balance
among sectors for efficient development. Telecommunication should be given its proper
place with the other sectors such as transport, industry, agriculture, and social programs.
2.1-2 Medium Financial Plan

Mid-range economy refers to the method used by an operating company (or
administration) to raise capital. For a private telecommunication company in a free
running economy the capital investment (e.g., new plant) will be financed by internally
generated funds and from the sale of securities. The dividends and interest to be paid on
those securities will drive from the tariff revenue. When the telecommunications operating
administration is owned by the government, capital for new plant may be taken from tax
payers, probably a legislated budget for the year. Then tariff revenues are returned to the
national treasury.
2.1-3 Micro Financial Plan

It deals with costs and return on investment for the expansion or upgrading of
12
a telecommunication plant. It is project-oriented or area-oriented. An example may be the
installation and cut over of several new exchanges with restored serving areas to meet
demand in a certain metropolitan area several years hence. The micro financial issue deals
with the problem of getting the best for the least expenditure.
Such projects and improvements or plant extension must follow the economic and
technical plans. Financial plans permit the extending of capital and will show payback to
management.
Network dimensioning is the first phase of the network planning process to initially draft
the network configuration and deployment strategy for the long-term. All the information
collected during the network analysis phase is required in the dimensioning phase.
The process of dimensioning delivers a lower bound on the number of network elements
and its configurations required to provide given services for a number of customers in
a service area with certain quality and within a specific period of time in order to deploy
the network.
Network dimensioning should be performed for the busy hour traffic, which ensures that
the availability for users is as good as possible. It also needs to be noted that dimensioning
is repeated in the case of network expansion. The result of dimensioning has two aspects;
it tells the minimum number of Exchanges due to coverage or capacity reasons. It is also
important to understand the forecasts for the subscriber growth and also the services that
are going to be deployed. The traffic history over the area has to be utilized to forecast
(identify) traffic increases during the next years. The better the traffic forecasts the better
the configuration can be optimized for network evolution using the dimensioning results.
Well dimensioned network has just enough capacity during busy periods. Any more
capacity than necessary adds network cost for operators reducing profit margins. Too little
capacity and customer applications will not be able to get network capacity so will deliver
a poor customer satisfaction.
2.2 Forecasting Plan

The growth of a telecommunication network is governed by the increased demand, and
13
the behavior of subscribers, which depend on economic factors, such as local employment,
income and purchasing power, and development of residential and industrial areas.
Forecasting should be carried in two adjoin areas; the subscriber density, and traffic per
line.
2.2-1 Forecasting of Subscriber Density
Forecasting of future subscriber density growth is an essential step in long-term
planning, where, many cases of severe performance degradation are due to the lack of
good performance estimation techniques. Good forecasting can be achieved by the
sensible use of mathematical models.
The importance of these models arises from the fact that the capacity of a loosely designed
network is soon exhausted and it will be a formidable and highly expensive task to change
the network to meet a denser tele-traffic demand.
For the local network planning, it is essential to have a forecast for the number of
subscribers at future dates. This can be based on a national forecast of the population and
of the communication terminals density. Reliable results could be obtained using sufficient
statistical data available from the past. Local knowledge of the telephone penetration and
growth of subscriber requirements for different types of industrial, commercial and
residential premises, together with knowledge of planned building developments enable
national forecasts to be built up.
The number of years evaluated in the past should be at least equal to the future period for
which a forecast is required.
In trunk network planning, the forecast is required in terms of busy-hour traffic in Erlangs.
Growth on particular routes is primarily determined by local factors, whereas total traffic
is more responsive to business activity and Gross National Product (G. N. P.).
Mathematical models can only be reliable if the network being investigated is sufficiently
large (i.e. a national network or at least a large local network or group of smaller local
networks). For example, in a large town there may be two or more trunk exchanges, each
with its own forecasts of growth, but only one repeater station carrying the combined
14
traffic. For smaller areas, adequate forecast can only be made by experienced staff with
local knowledge. However, it is useful to be able to check these detailed local forecasts by
means of a national forecast of the population and of the communication terminals density,
i.e.; the number of communications terminals per 100 inhabitants. Local variations
between forecasts and the national planning growth rate are to be expected. However,
where these are large and widespread geographically they should be examined specially.
Some of the forecasting models shown in figure 2.1 are:
2.2-1.1 Linear Model

For small number of years it is reasonable to assume that the number of subscribers'
terminals in a network increases linearly with time, so we have to find the best fit line for
the given data points. This line is known as Least Squares Line, or Linear Regression Line,
as shown in figure 2.1-a, such as:
M = Mo + K 1 t (2.1-a)
where, M : number of subscribers at time t years,
Mo : initial number of subscribers,
K1 : annual growth factor.
So, for n pairs of observations, (t1, M1), (t2, M2),......., (tn, Mn),
we can calculate the values of Mo, and k1 as;
Mo = (2.1-b)
K1 = (2.1-c)
After defining the values of Mo and, K1, we can predict the expected number of subscriber
terminals at certain time using equation (2.1-a). This model is rarely used, and the growth
factor may be approximated by a constant percentage per year.
15
2.2-1.2 Exponential Model
If the annual increase in the number of subscribers' terminals is a constant
percentage, the number will increase exponentially, as shown in figure 2.1-b, represented
by the following equation:
M = Mo exp (t / T1) (2.2)
the percentage increase per annum is 100 /T1
Figure 2.1 Forecasting Models.
This relation cannot provide an accurate model over an indefinitely long period of time,
because M ultimately becomes infinite.
In reality, however when the population gets large enough there are environmental factors
that slow down the rate of growth, Examples are food supply, predators, overcrowding,
and so on. These factors cause the rate of growth, dM , eventually decrease. In practice,
dt
it is reasonable to assume a boundary to the population size, and consequently there must
be some limit to the demand for communication terminals, so the terminals' density must
16
tend towards a saturation value MS, where, 0 < M < MS, and as M → MS, dM
0
, and
dt
the demand size tends to be stable.

Nevertheless, in many countries of the world, the number of terminals installed is so far
below the potential number of subscribers' terminals that the exponential curve provides
an accurate model for forecasting purposes.
2.2-2 Forecasting of Traffic per line

For planning a junction or trunk network, it is necessary to have a forecast of the
traffic originated in each local exchange area (i.e. Erlangs at the BH), rather than the
number of subscriber's lines. Growth on particular routes is primarily determined by local
factors, whereas total traffic is more responsive to business activity and G. N. P. In the
local network, a picture must be built up of growth of subscriber requirements year by
year, both in terms of numbers and of location. Subdivision into business and residential
subscribers (with the latter subdivided into classes which indicate the probable calling
rate) is valuable for exchange-design requirements.
It has been found that economic growth makes very little difference to average
subscribers' calling rates. This may be due to the increase that occurs in the number of
telephones. If new subscribers have a lower calling rate than average, they will offset an
increase in the calling rate of existing subscribers. It is therefore reasonable to assume that
the average calling rate for each class of subscriber (e.g. residential or business) in an area
remains constant and that the increase in total traffic will therefore be proportional to the
forecast increase in the number of subscribers' lines. Thus, the linear model can be used to
estimate future traffic originating in an area from the forecast G. N. P.
If G. N. P. was the only factor affecting the growth of traffic originating in an area,
all areas would be affected equally and traffic on all routes between them could be
expected to grow by the same percentage, i.e. the average traffic per subscriber between
any two areas would remain constant. However, areas develop differently because of other
factors. If the numbers of subscribers in two areas increase by unequal proportions, one
17
cannot assume that the traffic per subscriber between them will remain constant. For
example, if the number of subscribers in area Q increases by a greater percentage than the
number in area P and the average traffic per subscriber in P to and from subscribers in Q
remains constant, then the average traffic per subscriber in Q to and from subscribers in P
must decrease. More probably, the traffic per subscriber will change by a smaller amount
in each area.
2.3 Provision Timing Plan

In a telecommunication network, the amount of plant provided must provide an
adequate grade of service for the traffic at the busy hour. Although the traffic offered to
a telecom. system may grow continuously, it is usually expensive to extend the system by
very small amounts at frequent intervals. Instead, to meet the growth requirements of
a network, more substantial extensions are made at intervals of several years with demand.
Each extension is designed to cater for the growth forecast over a number of years known
as the provision period. The date at which the next extension is forecast to be required is
known as the design date, and the period for which a plant increment will cater is known
as the design period. In a developing network, the selection of the correct plant provision
periods is important as they are normally chosen to satisfy number of technical
requirements and to avoid wasteful expenditure.
Delays in the provisioning of equipment and accommodation can lead to distortion of the
plans.
If plant is provided in large increments, the unit cost of the installed plant can be low and
interruptions to service from installation work are infrequent. However, high initial capital
cost is incurred before it is necessary, leading to a large burden of spare plant not earning
revenue. On the other hand, provision in smaller increments gives higher unit costs and
greater risk of interruption to service, but with a lower burden of spare plant. Optimization
of the design period involves consideration of the fixed and variable costs, life of plant,
interest rates on capital and growth of traffic, together with such practical constraints as
the time needed to install the equipment.
18
For example, in the simplest case where the cost of a discrete extension takes the form
y = m x + c (where c is the cost not related to the size of the installation, m is the
incremental unit cost and x is the size of the extension) then economies of scale can be
achieved by providing the plant in bulk. An economic study is necessary, however, to find
the right period. The final result is influenced by such factors as the fixed and variable
costs, and the availability of capital. With the network consisting of many items of
equipment with differing provisioning and design periods, typical provision periods
currently in use are:
Sites 20 - 40 years
Exchange numbering schemes 30 years
Exchange accommodation 10-20 years
Switching equipment 2 - 3 years
Transmission equipment 2 - 3 years
Cable duct 20 years
H F trunk cables about 10 years.
Some plant items do not readily lend themselves to the use of standard provision periods,
particularly where costs are critically dependent on rates of growth.
In carrying out the annual review, forecasts are made of significant exchange traffic
parameters (such as trunk or junction terminations, originating traffic etc.) for each year of
the review, the starting point for the forecast being the current measured traffic in the
appropriate section of the exchange. In preparing the forecasts, account is taken of local
factors. However, the basis is the national planning growth rate. The traffic parameters for
which forecasts are prepared are chosen as key indicators of plant requirements in the
exchange. It is however, necessary to take special measures to ensure that this timely
provision takes place in practice, in the changing environment of the network where both
traffic needs and plant availability change with time.
The character and rate of introduction of new developments has to be governed by the
ability to continue to manage the growth of the total system and the need to inter-work
19
with existing plant. The resources available, new technology and range of the foreseen and
unforeseen services all lead to a desire for a flexible general-purpose approach to
development, with a recognition of the economic penalties if judged in simple terms
against the initial capital cost.
Sheet – 1
1-The number of subscriber lines put in service in certain Exchange area during the interval 2012-
to-2017 was registered in the following table.
Year 2012 2013 2014 2015 2016 2017
Number of lines x 103 100 104 108 111 111 114

Calculate the expected number of lines required to be connected at year 2030.
Solution: the annual growth is not equal so, the linear model is used.
M = M o + K1 t
year ti Mi ti Mi t i2
2012 1 100 100 1
2013 2 104 208 4
2014 3 108 324 9
2015 4 111 444 16
2016 5 111 555 25
2017 6 114 684 36
n 6 n 6 n 6 n 6
 ti  21,  M i  648 ,   91,  ti M i  2315
t i2
i 1 i 1 i 1 i 1
 n 2  n   n  n 
  
Mo = i 1
t i    M i     t i    t i M i 
   i 1   i 1   i 1 
2
 n 2  n 
n   ti     ti 
 i 1   i 1 
20
 Mo = 91648  212315  98.6 (1)
6 91  212
 n   n  n 
n   ti M i     ti    M i 
 K1 =  i 1   i 1   i 1 
2
 n 2  n 
n   ti     ti 
 i 1   i 1 
 6 2315  21648
K1 =  2.69 (2)
6 91  212
The approximated line will be;
M = Mo + K1t = 98.6 + 2.69 t
Year 2030 gives t = 19
 M = 98.6 + 2.69 (19) = 149.71 = 150

2- Calculate the period of time during which the communication demands will be equal to double
its value now if the subscriber annual growth is 0.05.
Solution: the annual growth is constant, then the exponential model is more suitable in this case
with;
1 / t1 = 0.05, M = 2 Mo;
 M = Mo exp ( t / t1)
 2 Mo= Mo exp (0.05 t)
 lin 2 = 0.05 t, t = 100 lin 2 / 5 = 20 lin 2 ≈ 14 years

3- Calculate the expected number of terminals that will be required to be connected in the LAN
after 5 years, if the number of served users now is 1000 subscribers and the annual growth rate
is 0.05.
st
4- Calculate the capacity of the 1 stage of an exchange required to be working over 20 year to
reach its maximum capacity of 25000 subscribers. The expansion would be provided each
quarter of its life time as 35 % growth rate.
5- Fined the expected maximum number of users terminals in a communication network, at which
the maximum rate of growth occurs, with graphical illustration.
6- A local Exchange with capacity 8000 lines, one year ago the initial number of lines put in
21
service in this Exchange was 500, and now there are 2000 line in service, provided that
Exchange expansion follows Logistic function. How many lines will be in service three years
from now.
7- The number of subscriber lines put in services in certain exchange area over the period from
2017- to - 2023 was registered in the following table.
Year 2017 2018 2019 2020 2021 2022 2023

4 583 615 646 697 738 802 844
Number of lines x 10
Compare between using two different forecasting models to predict the lines required to be
connected 10 years ahead.
22
CHAPTER 3
TECHNICAL PLAN
This plan is concerned with defining the major technical standards and detailed
guidelines to be followed to assure the required flexibility of a communication network,
with its compatibility among its parts and to guarantee that service performance meets the
desired standard. The important attribute of technical plans is that they specify the
technical details of the network’s capabilities.
Once the initial decisions have been made, the network planning process involves three
main steps:
● Topological design: This stage involves determining network hierarchical structure and
topology, i.e., where to place the components and how to connect them. This involves
determining the cost of transmission, the cost of switching, traffic routing and location of
switches and concentrators.
● Network-synthesis: This stage involves determining the size of the components used,
subject to performance criteria such as the Grade of Service (GoS), and using this
information to calculate a routing plan.
● Network realization: This stage involves determining how to meet capacity
requirements, and ensure reliability within the network, and involves determining all
information relating to demand, and reliability.
These steps are interrelated and are therefore performed iteratively, and in parallel with
one another. The planning process is highly complex, meaning that at each iteration an
analyst must increase his planning horizons, and in so doing, he must generate plans for
the various layers outlined above.
Communication networks must be planned properly to use the limited resources efficiently
to meet present and future users' demands. This requires that the network must provide:
23
CHAPTER 3 TECHNICAL PLAN
- High grade of service on demand,
- High quality of service.
This plan is normally divided into a number of sub-plans typically: Switching, Numbering,
Signalling, Capacity, Routing, Charging, and Transmission Plan.
3.1 Switching Plan

It is one of the main functions of the communication process, where it deals mainly
with the allocation of communication resources available in the network, that enables the
information to proceed between any pair of end users over the network satisfactory and
economically. Switching nodes are distributed along the network to set-up communication
path on demand and release it when it becomes no longer needed. It performs logical
functions to establish the path and automatically charges the user for his connection
period. The main requirements of the switching systems are:
1- Connection reliability between any pairs of end users on demand,
2- Connection or release time should be relatively small compared to information transfer
time,
3- Secure information transfer and privacy.
Switching nodes are interconnected in different ways to form different network topologies,
described in the following section.
3.1-1 Network Topologies

The structural topology of a network is important to determine the suitable protocol
used to satisfy user’s requirements. Network topology refers to the physical and logical
structure of a network, where the physical topology refers to the way the network nodes
are interconnected, while the logical one refers to the way signals flow over the network.
One method of network interconnection with high reliability provides a direct route
between any two nodes. Each of the N nodes must be connected with all other (N-1)
nodes, calling for N (N -1) links. As each node is a transceiver, then the actual number of
required links is N (N - 1) / 2, as given in figure 3.1.
24
It is clearly uneconomical, inefficient for every subscriber wishing to be served to have an
exclusive line to every other subscriber with whom he may wish to communicate. Since
each terminal has N -1 incident lines, a selector is needed at each terminal to connect it to
one channel at a time.
Resource Sharing; is a more efficient and economic way to connect a group of similar or
dissimilar, spreading around subscriber’s terminals, with each other using some form of
switching centre (Exchange), at which routes can be selected between message source and
destination.
Figure 3.1 Fully Connected Network.

Figure 3.2, shows a generalized centralized switched network, where a large community of
N users are able to interchange information using a single switch (Central Office), capable
of interconnecting the users' terminals to each other by any one of a number of possible
methods.
Sharing of a common communications medium, regardless of its physical or electrical
form requires a set of rules, or a protocol, that governs the behaviour of the group of users
so that all users are able to gain access to the central resource. Some formalized structure
is necessary so that the resources of the shared communications medium can be fairly
allocated among the large group of connecting users.
25
Figure 3.2 Centralized Control Connected Network.
There are different network topologies such as; star, ring, loop, bus, and tree topology,
where each of them has its own advantages and disadvantages and the choice between
them depends on the application.
Usually a practical telecommunications network is formed as a hybrid topology by
combining different network configurations such as the combination of star and bus or
tree topology.
All the above topologies are used for local access networks, each interconnects
a variety of devices, and spans a local (limited) area typically within a facility. The
geographical extension of such network may extend from less than 30 meter to over 10
km. As communications demands grew, and users desired longer distance connections, it
became necessary to interconnect the individual service areas with links between the
central offices. The overall telecommunication network known as Public Switching
Telephone Network, PSTN consists of local networks, interconnected by a long distance
network as illustrated in figure 3.3. End users usually connected to LAN nodes that carry
out switching functions using Subscribers’ Loops, while each two local switches are
connected by an outgoing Direct Trunk to facilitate communications between local areas,
as shown in figure 3.4.
26
Figure 3.3 PSTN Network.
Figure 3.4 Direct Route used to connect local loops
A Tandem Exchange (traffic concentrator) is used to route transit calls between local
exchanges. Routing through a Tandem Exchange is an important economic expedient for
a telephone company or administration.
A Direct Trunk Circuit is used when there must be sufficient traffic (over 20 Erlang for
BH) to justify such a circuit.
For a connectivity with traffic intensity under that value, the traffic should be routed
through a Tandem Exchange as indicated in figure 3.4. Figure 3.5, illustrates direct route
used for the connection between tandem centres.
27
3.1-2 Hierarchical Network

The PSTNs of the world universally used a Hierarchical Network as indicated in figure
1.6, that recommended by CCITT. This network created for the purpose of concentrating
traffic from the lower levels of the hierarchy (Subscribers’ Loops) to the higher levels
(International Network). This concentration reduces the transmission cost for making
a connection between two distant telephones at the expense of increased switching cost in
the hierarchy of switching centres.
Networking therefore involves a trade-off between transmission cost and switching cost at
each level of the network hierarchy. This trade off between transmission costs and
switching costs depends strongly on the technologies used for building transmission and
switching facilities
28
Figure 3.5 Direct Route used for the connection between Tandem Centres in large Networks.
Today there is a trend to reduce the number of levels in the hierarchy by using intelligent
switching centres
3.1-3 Switching Techniques

The choice of switching method used to construct the communications network
serving the various traffic classes is a major design decision. There are three main
switching technologies: Circuit Switching, Message Switching, and Packet Switching.
Each of these switching techniques has advantages with respect to voice or data
transmission. Therefore, it is important to understand their capabilities and limitations in
order to be able to apply the techniques to meet a particular communication requirement.
3.1-3.1 Circuit Switching

29
In circuit switching, a dedicated line (actual connection) is necessary before data can be
transmitted. The information offered at the input of a switch appears almost
instantaneously in the same form at its output. Figure 3.6-a illustrates the process of setting
up a dedicated physical path between communicating subscribers, while figure 3.6-b
shows a circuit-switched network providing services using the established physical path
between communicating subscribers.
Figure 3.6-a Circuit Switching Network.
The PSTN is an example of circuit switching application.

A person or terminal places a call by entering into the switching system the directory
number of the called subscriber.
Figure 3.6-b Circuit Switching Network.
The switch then sets up a connection between subscribers, consisting of a sequence of

point-to-point circuits, joined together by switches at the junctions between them. The
connection is set up by a signalling message which passes all the way through the network.
30
This is known as Circuit Establishment phase. A return signal confirming circuit
establishment, telling the source that voice or data transmission can begin during the Data
Transfer phase.
The connection exists between the two communicating parties until they decide to "hang
up", or terminate the connection after data transfer completion in Circuit Disconnection
phase. They use the circuit exclusively. Although some multiplexing may take place in
a portion of the transmission system, the parties will not notice it. The three phases of
connection are shown in figure 3.7-a.
Advantages of Circuit Switching
Circuit switching has several advantages such as;
1- Fully transparent; i.e., once connection is made delivering is guaranteed and sequential.
2- Delay is constant and small, encountered prior to data transfer (of the order of 10-20 sec.)
3- It is appropriate if the users need to communicate at a fairly constant rate for a long
period of time.
4- Simultaneous availability of the two parties is necessary.
5- Interactive session is possible, i.e., it allows customers to interact until they are satisfied
that the information transmitted is correct, so customers or machines can converse with
one another as rapidly as they please, communication takes place in real time.
Disadvantages of Circuit Switching
The disadvantages of circuit switching are as follows;
1- Inefficient; for burst traffic which has a high peak-to- average information transfer rate,
where channel may be idle for a significant portion of time, yet remain unavailable to
other users. This is clear since λX <<1, which is the fraction of time during which the
link allocated to the message is busy, then the link is idle most of the time
L
(inefficiently utilized), where; X= is the transmission time of a message with length
CL
31
Figure 3.7 Connection phases of different Switching Techniques
L bits over a given link with bit rate CL in the path (expected link busy time between
arrivals), and λ is the message arrival rate. CL should be high enough to minimize X.
2- Slow; delay prior to data transfer of few seconds for only few micro seconds for data
transmission.
3- Relatively expensive; charge is based on time and distance not on data transmitted.
4- One line is exclusively and continuously used per connection, the connection between
customers must be made before they can communicate, and the connection (consisting
of links and switches) must be available during the whole duration of information
32
exchange.
5- Blocking may occur when all lines are busy.
3.1-3.2 Message Switching

It is a switching technique, where customers exchange information by sending
each other discrete messages. Figure 3.8 indicates that, a message which the user wishes to
have transported as a whole is moved from node to node.
Figure 3.8 Message Switching Network
To ensure quick, reliable communication, route the message to the correct destination, and
to control congestion over the subnet, a header information is appended to the message,
and the message works its way through the network, from link to link, passes from one
node to another.
At each node (switching centre), the message is received, inspected for errors, stored
on disk until link (facility) become available, it is then forward on to the next node closer to
the destination. This storing and forwarding process continues until the message reaches its
destination. It is therefore called a store and forward (Fully transactional) switching
method. The node is a general-purpose minicomputer with sufficient storage used for entry
into the network, for buffering, and for routing messages to subsequent nodes towards the
destination. An example of message-switching network is; air line reservation network.
Since message may be competing with other messages for access the facilities, a queuing
delay may be incurred while waiting for the link to become available. The length of the in
33
transit delay depend on the network capacity, and the related probability that the circuits
are busy. Various forms of conversion may be applied to the data stream as; time, (the
message can be forwarded at any desired time according to the chosen charging rate, and
the data is retrieved from a mailbox at a time determined by the receiver), delivery order
according to the priority discipline, bit rate (between local and trunk circuits), flow rate,
code, protocol, message format, message composition. The time diagram of message
switching is displayed in figure 3.7-b.
Advantages of Message Switching
1- No dedicated physical path is necessary in advance between the two stations.
2- Simultaneous availability of the two communicating parties is not necessary.
3- No blocking is encountered but storing delay may occur.
4- It can operate more efficiently with a higher trunk utilization as channel is used only
when transmitting data.
5- Capable to send the message to many destinations by appending addresses.
6- Providing priority to users is possible.
7- If error is discovered, it is possible to request retransmission.
8- Speed and code conversion can be performed, in order to allow terminals with different
operating characteristics to converse with one another.
9- No limit on message size.
10- It is more flexible in adapting to traffic peaks, since it can store messages during the
peaks and send them on their way later when a path becomes free. Therefore, systems
do not have to be designed for peak loads, and thus cost less. Since a higher percentage
of attempted calls are completed, more revenue is collected for a given capital
investment.
Disadvantages of Message Switching
1- Large storage capacity must be available at each node.
2- Long storing delay with great variance may occur at each node on route to destination.
3- Not suitable for interactive connections.
4- It is designed for the one-way delivery of messages, so it is not suitable for real-time
34
conversational interactions between people or computers.
Usage of Message Switching
1-It is not used in LAN due to delay and storage limitations.
2-It is used in governmental, military and business applications for legal and historical
reasons.
3.1-3.3 Packet Switching
Transmitting very long messages as units in a subnet is harmful in several ways
including delay, buffer management, and congestion control. Thus, messages are usually
broken into shorter bit-strings called packets. Depending on the implementation and the
form of the information, there may be more than one level of subdivision. For example,
the user messages are divided into "segments", and the segments are further divided into
packets as shown in figure 3.9-a. Packets move around the network, from node to node, on
a hold-and-forward basis. These packets are hold temporary at each node to check for
a free link and then forwarded along the best available path to the next node as shown in
figure 3.9-b, through the subnet as individual entities with quite high speed. So, Packet
switching is devised to overcome the long transmission delays inherent in message
switching.
Figure 3.9-a Message Packetization
35
Figure 3.9-b Packet Switching Network
For example, for a message of 1000 bits and a transmission speed of 48 k bits / sec, the
transmission time is 21 m sec. Because of this short transmission time, circuit switching is
really inappropriate, because the time to set up and clear circuits is much longer than the
message transfer time. Packet switching technique is more appropriate for data traffic
than circuit switching. There is an upper limit to the block size to be transmitted (e.g., of
the order of 1000 - 5000 bits).
Each packet must have additional bits added for address and administrative purposes. For
example for a message of 1000 bits and a transmission speed of 48 k bits /sec, the
transmission time is 21 m sec. Because of this short transmission time, circuit switching is
really inappropriate, because the time to set-up and clear circuit is much longer than the
message transfer time. Packet switching technique is more appropriate for data traffic than
circuit switching. It is similar to message switching except that the complete message is
not sent at one time. Figure 3.9-c shows the packets of a particular message, sent packet by
packet over diverse routes through the network to the destination.
36
Figure 3.9-c Voice Communication Packet Switching Network
Only at the destination the message packets are collected and reassembled in proper order,
making the voice intelligible. Figure 3.9-d focuses attention on packets flowing between
users A and B, where other packets flowing between other users would be simultaneously
moving throughout the network. Each node holds a copy of the transmitted packet in
temporary storage until the node is sure that it has been received correctly by the next
node, which transmits back (+ ve) ACK confirming successful reception.
Figure 3.9-d Packets Flowing between two Users A & B
The flow of the message is initiated by the transmission of packet 1 between user A and
switch 1. Upon fully receiving the first packet, switch 1, following a set of routing rules,
transmits packet 1 toward its destination by sending it via switch 3. In the meantime,
37
packet 2 is moving from user A into switch 1. During this time, the conditions in the
network change (for instance, a large amount of traffic from switch 5 arrives at switch 2),
so the second packet of the message from A to B is routed via switch 4. The third packet
of the message, arriving at switch 1 soon after the second packet, is similarly routed via
switch 4.
Each packet is error-checked along the way, each time another wide band link is
traversed. After switch 4 received correctly packets 2, and 3, the second packet is
transmitted to the destination switch, switch 3, but assuming that during that transmission,
an error occurs. Such errors can result from a burst of noise on the transmission path, static
from lightning or switching equipment. When switch 3 receives packet 2, the error-
detection mechanism is able to detect the error and requests a retransmission of packet 3.
However, while this, is occurring, packet 3 has been transmitted immediately behind the
first (in error) copy of packet 3. As a result, the second (correct) copy of packet 2 is
received at switch 3 after packet 3. If the network is viewed from the perspective of switch
3, first packet 1 is received, then packet 3, and finally packet 3. If switch 3 delivered the
packets to the destination (user B) in the same order they arrived at switch 3, user B would
receive the packets in a different order than they entered the network, so switch 3
reassembles packets into complete messages, which are then presented to the subscriber.
Packet Switching is suitable for interactive traffic.
Advantages over Message Switching

1- Pipelining: packet 1 is transmitted from node B to node C at the same time that packet 2
is transmitted from node A to B. This simultaneous use of communication circuits
yields considerable gains in efficiency and minimizing delay.
2- Similar to message switching, the transmission channel is occupied for the duration of
transmission of the packet only. The channel is then available for use by packets being
transferred between different data terminals. Also, many packets of the same message
may be in transmission simultaneously. Thus, packet switching fully utilizes the
relatively expensive transmission links.
3- Lower errors: as packets are short, they are less likely subject to errors than complete
38
messages. This leads to lower probability of retransmission.
4- Lower congestion: As packets may be routed through the network independently,
congestion may be minimized by taking different routes to their destinations.
5- Each communication link is fully utilized whenever it has any traffic to send. Using
communication links on a demand basis markedly decreases the delay in the network
relative to the circuit switching approach.
6-The speed of the network is higher than with message- switching. With conventional
message-switching systems, the emphasis is on low errors, rather than speed, in the
information transfer between customers. It is not intended that customers should interact
rapidly with each other through a message-switching network. In packet- switched
systems, customers are expected to interact with each other by exchanging packets, in
much the same way that they would interact by exchanging information through
a circuit-switched connection. Information is exchanged in the form of short packets,
and the time of transit of these messages is kept low.
Thus, a packet-switching network can be expected to deliver its packet in a fraction of
a second, while a message-switching system delivers its message in a fraction of an
hour.
Packet switching therefore combines the features and major advantages of both circuit and
message switching, i.e., speed and transmission efficiency. However, it has the dis-
advantage that the queueing delays in the nodes are hard to control. The packets queued at
a node come from inputs at many different sites, and thus there is a need for control
mechanisms to slow down those inputs when the queueing delay is excessive, or even
worse, when the buffering capacity at the node is about to be exceeded. There is
a feedback delay associated with any such control mechanism. First, the overloaded node
must somehow send the offending inputs some control information (through the links of
the network) telling them to slow down. Second, a considerable number of packets might
already be in the subnet heading for the given node. This problem is caused by the store-
and-forward approach and is largely nonexistent in the circuit switching approach.
39
Network-Introduced Problems
Figure 3.9-d, shows a number of acknowledgement packets flowing on the various links in
the network, in the opposite direction from the information packets. These packets are the
key to the error-detection mechanism that is needed to insure the integrity and accuracy of
the transmitted data. Any information packet that is properly received is immediately
acknowledged back to the sender with one of these short acknowledgement packets. In this
way, the sending switch knows that the information packet has been received properly by
the next switch along the path toward the destination.
If an acknowledgement is not received within a certain period of time (known as the
timeout period), the sending switch presumes that the packet was received in error and
retransmits the packet. This presumption is necessary because it is quite possible that any
transmitted packet could be so badly garbled that the receiver could not even make enough
sense of the packet to intelligently ask for a retransmission. If a packet is received with
only a minor error, a negative acknowledgement asking for a retransmission, avoids
having to wait for the full timeout period to elapse.
Packet sequencing is only one of a number of possible network protocol-introduced
problems that can occur in the packetizing process. The other two most serious problems
are the undetected loss of a packet and the duplication of a packet that is successfully
transmitted.
Packet Sequencing: The problem of packet sequencing is a direct result of the hold-and-
forward mode of operation, arising from the need to protect each transmission from
network- introduced errors. Differential delay; along the many paths through the network
also introduce the possibility that packets will be received out of proper sequence. In order
to protect users, the packets have to be "reassembled" into the same basic message
structure they had upon initial transmission into the network. The process of packet
reassembly is done at the destination witch, (switch 3) using packet sequence information
(such as a serial number) which has to be carried through the network along with the user-
introduced information.
40
Losing a Packet: In figure 3.9-e, user A enters the first packet into the network via switch
1. Switch 1 routes packet 1 via switch 2, which receives packet 1 correctly and
immediately acknowledges it. However, before packet 1 is transmitted from switch 2
toward switch 3, something goes wrong, and switch 2 fails. Having received an
acknowledgement for packet 1, switch 1 is no longer concerned about it. However, upon
the failure of switch 2, the network routing plan changes, so that future packets avoid the
failed switch.
Figure 3.9-e Packet Switching Network with a switch Failure: Loss of a Packet
Thus, packet 2 and packet 3 are routed via switch 4 and are soon thereafter received
properly at the destination switch (switch 3). But what about packet 1?. It was lost when
switch 2 failed. Acknowledging packet 1 meant that switch 1 was no longer responsible
for this packet, yet switch 2 failed before it had a chance to relay packet 1 on through the
network.
For the destination user B, packet 1 is irretrievably lost, and all he receives are packets 2
and 3. The basic packet switching protocol has thus introduced the possibility of lost
packets.
There are a number of ways, the network can protect against this problem. First,
a switch could be restricted from sending an acknowledgement until it has actually relayed
the packet on. Second, ultimate responsibility for the packet could rest with the originating
switch. Third, the sending user could be required to fill in missing (lost) packets at the
41
request of the destination switch. All these possible protections, as well as others, have
their advantages and disadvantages, which have to be explored during design and
implementation of operational protocols. The key point is the need to layer multiple
protective checks and protocols to deal with the possibility of lost packets.
Duplicate Packets: In figure 3.9-f, the flow of the message from user A to user B is
initiated at switch 1 with packet 1 being routed via switch 3. Packet 1 is received correctly
by switch 2 and is immediately acknowledged by switch 3. However, just as the
acknowledgement for packet 1 leaves switch 2, the line from switch 2 back to switch 1
fails, in the process destroying the acknowledgement for packet 1.
Figure 3.9-f Packet Switching Network with a Line Failure: Duplicate a Packet
Not receiving an acknowledgement for packet 1 within the timeout period causes switch 1
to retransmit packet 1. Detecting the failure of the line between switch 1, and switch 2
causes the network routing plan to send the retransmission of packet 1, together with the
transmission of packets 2 and 3, via switch 4.
In the meantime, the first transmission of packet 1 was really received properly by
switch 2, even though the acknowledgement of that first copy was never received by
switch 1. Having no way to know about the error, switch 2 relays packet 1 to the
destination switch, switch 3. Soon thereafter comes the second copy of packet 1, together
with packets 2 and 3. What user B is likely to see, therefore, is a duplicate packet 1
together with the rest of the message in its entirety.
42
Packet Switching Formats

There are two basic modes for packet transmission; datagram and virtual circuit packet
switching. Their time diagram is shown in figure 3.7-c, d.
i- Datagram
In the datagram mode (dynamic routing), each packet is treated independently with no
prior route determination, where routing decision is required for each packet at each node
through the network. Packets travel along any available path to the destination and are
delivered in the sequence they arrive at the receiving terminal, and delivery is not
guaranteed. No attempt is made to deliver packets in the sequence they entered the
network, so packets may arrive out of sequence or be lost or duplicated.
It is more flexible, and preferred when short data stream is transmitted. Electronic mail,
that requires the exchange of short unrelated segments of information, is ideal for datagram
usage.
ii- Virtual Circuit

In the virtual circuit mode, a particular route is set up when a session is initiated for
all packets exchanged during the session life between users using a path finder packet,
where each packet travels the same path through the network. Packets arrive in the order in
which they are sent, such as delivery of packets in proper sequence with no errors is
guaranteed.
Routing decision is not required at each node, it is made only once for each
connection, but a packet still buffered and queued. Logical connection is established (call
request, call accept, clear request). It is like circuit switching as it appears to be as
a dedicated physical circuit between source and destination although the circuit is shared
among multiple users. It is virtual in the sense that transmission capability is only used as
needed (on demand).
Virtual circuit routing is generally used in practice, although there are many interesting
intermediate positions between virtual circuit routing and dynamic routing. Multi-packet
43
messages are best handled by virtual circuit operation which is good for long messages
and for relieving stations of processing burdens.
Comparison of Circuit, Message, and Packet Switching

The choice of which switching technique to use is dependent on the particular
application, performance requirements, cost, ….,etc. Taking the user's Delay Time as
a comparison parameter between Circuit, Message, and Packet Switching, then;
For Circuit Switching Technique;

There is a delay in setting up the path through the network which increases as the
load on the network increases. However, once the path is established, information
exchange takes place at transmission rate C, dependent only on the physical properties of
the circuits, and customer’s equipments, which make up the path connection. Call setup
consists of a connection delay at the switch followed by transmission of a setup signal.
This cycle is repeated until the terminating switch is reached, at which point a return signal
is generated and sent back to the originating switch. The data are then sent from
originating to terminating switch with relatively little delay.
Thus, with circuit switching, the start of an interaction will be delayed as the load on the
network increases, but once begun it will proceed at a speed unaffected by loading on the
network, so circuit switching delay time is calculated as;
TDCS = Tset-up+Tproc.+ Ttrans.+Tprop.+Tdiscon. (3.1)
where;
Tset-up : Time taken by some nodes to establish the circuit before message transmission.
Generally, it is a constant value depends on the number of nodes connected in the
transmission path. The connection acknowledgement time can be neglected since the
circuit is already exist.
Tproc. : The processing time is the time taken by the transmitting node to process any function on
the message as A / D conversion, modulation, multiplexing, etc. It has a constant value.
Ttrans : Time taken by the transmitting terminal to send-out the information message. It depends
on the speed of the originating terminal, and the message length, and the number of
44
nodes in the path. Hence;
Ttrans = L (message length ) in bits

C ( node, link data rate) in bits / sec .
Tprop. : The propagation time is the time taken by the entire information message to move from
one switching node to the next towards the destination node. It depends on the data
speed on the junction, and the junction length, and the number of the links on the path.
Hence;
Tprop=
where, N is the number of links on the path.

Tdiscon : It is the time starting by the clear-down signal sent by the call terminating terminal to
his local switching node that send a disconnect message releasing the individual
trunks along the complete circuit. It depends on the number of the switching nodes on
the path.
For Message Switching Technique;

The entire message is transmitted from the originating switch to the next switch along
the network path. Following the receipt of the message, the sequence is repeated, until the
terminating switch is reached. The time necessary to transfer a message through the
network consists of the time to travel between switches at the transmission rates of the
respective links, and a message handling time at each switch, including the time taken to
read the message into and out of storage. An increase in load causes queues to develop at
switches while messages wait to be processed, or for links to become free.
This increases the time required for messages to pass through the network. An increase in
load can result in congestion. To the user, this may not be apparent for a while, because
the network continues to accept messages, storing them for later delivery. However,
messages may not be able to be forwarded for some time. Eventually, the system breaks
down and no messages are forwarded unless the congestion is relieved. The message
switching delay time is calculated as;
TDMS = Tproc..+ Ttrans.+ Tprop + Tstor. + Tack. (3.2)
45
where;
Ttrans= ( N +1 )
H : is the message header, contains all information required to help for correct message
receiving at the destination such as source and destination addresses, parity bits for
error detection and correction, etc.
Tstor. : The storage time is the time taken to store the message in the nodes buffers to check
the address, check the message for error if any and correct it if possible, otherwise
request retransmission, check for available path to forward the message toward the
destination. It depends on the message length, the buffer size, and the traffic load on
the network.
The values of Tproc. and Tstor. are calculated by multiplying their values per switching
node by the number of switching nodes on the message path.
Tack. : The acknowledgement time is that time taken by the acknowledgement signal to
inform the previous switching node for the correct reception of the message or to
ask for retransmission. Its value is calculated as the summing of both transmission
and propagation time multiplied by the number of links on the message path.
For Packet Switching Technique;

Following the receipt of the first packet at a node, it can be relayed to the next node
while the second packet is being received. Similar to message switching, the time of transit
of a message through a packet-switched network is a function of network load. However,
the transit time for individual packets is purposely kept low, as a result of using high-speed
lines and not having to read the message into and out of off-line storage at each switch.
Thus, in general, messages experience lower delay in a packet-switching network than in
a message-switching network. Packet switching delay time is calculated as;
For Datagram Packet Switching Technique;

Due to pipelining, the delay time can be calculated as;
TDDG = Ttrans. 1st pkt + Ttrans. of the rest+ Tprop.
46
=
( P  h) (N + 1) + ( L  H )  ( P  h) + T . (3.3)
prop
C C
where, P : Packet size in bits,
h : Packet header size in bits,
N : Number of switching nodes in the path
The processing, and queueing time are neglected in this case, but any retransmission time
should be added, with the waiting time before retransmission to minimize the probability
of a second collision, which assumed to be ½ (P+1) T, with T is the packet length in
seconds.
For a delivery guaranteed packet switching (no pipelining), the second packet will not
transmitted until receiving the acknowledgement that the first packet has been received
correctly, and so on. So, the delay time in this case will be;
TDDG = Tproc+ Tque..+ Ttrans + Tprop.+. Tack.

where, Tproc. , Tque..are calculated by multiplying the time per packet by both number of
packets in the message, and number of switching nodes on the path.
Tack. is calculated as transmission, and propagation time per acknowledgement packet
multiplied by both number of packets in the message, and number of switching nodes
on the path.
For Virtual circuit Packet Switching Technique;

TDVCS = Tset-up+ Tproc.+ Ttrans+ Tprop+ Tack. (3.4)
Figure 3.10, shows the variation of the message delay of circuit, message, and packet
switching, for a fixed line utilization of  = 0.5, as a function of the average message
length. For message and packet switching, a network queuing model incorporating the
queuing delay at individual nodes, retransmission of packets or messages in error, and re-
assembly of packets at the destination node, is used to yield an average message
transmission delay. The delay analysis of circuit switching is used to develop
47
a comparable circuit-switched delay which is a combination of message transmission
delay and line setup time. This setup time is measured from the time a request to transmit a
message occurs to the time an idle channel is reserved in order to transmit that message.
Figure 3.10 Variation of the Message Delay as a function of the average Message Length.
Under a fixed line utilization, as the message size decreases, the number of messages
increases. Thus, there is more delay experienced under circuit switching in trying to
reserve a channel in order to transmit the message. In fact, line setup time tends to saturate
at short message lengths, which accounts for the sharp increase in the delay curve
representing circuit switching in figure 3.10 as the message length decreases. Although the
time to transmit a message decreases at short message lengths, this is more than offset by
the saturation effect of line setup time.
At long message lengths, the message transmission time increases. This effect is
predominant over line setup time and results in an overall increase in delay for circuit
switching as the message length increases. For message and packet switching, long
messages mean a longer time spent waiting in queue, causing an increase in overall
message transmission delay. This effect is greater for message switching than it is for
packet switching where the long messages can be broken up into small packets so that
long queuing delays can be avoided.
In figure 3.10, we note that there are regions in which each type of switching technology is
effective. Packet switching has lower average delay than message switching over the
48
whole range of average message lengths. Circuit switching delay is less than that of packet
switching above message lengths of approximately 5000 bits.
Figure 3.11 illustrates the variation of the throughput for circuit, message and packet
switching with the message length. Throughput is defined as the volume of traffic which
the source node of an n-node, sequential network model can process during a unit time for
a message transmission delay of 200 m. sec.
The throughput for circuit switching, however, must take into account the saturation effect
of line setup time at short message lengths.
At a certain value of line utilization, line setup time increases sharply. This value is
different, depending on the average message length specified. Traffic throughput is
defined as the line utilization at which the rate of change of this increase is approximately
the greatest. This quantity, which is dimensionless, is arbitrarily taken to be 0.9 times the
saturation point of line utilization for the different message lengths.
Figure 3.11 Variation of the Throughput of Different Switching

Techniques with the Message Length.
Note that the top curve for packet switching corresponds to a maximum packet size of
1000 bits while the bottom curve corresponds to a variable packet size, chosen to give the
maximum packetizing efficiency for buffer utilization. Packet switching is always better
than message switching in the former case while, in the latter case, message switching has
a higher throughput until a message length of about 1000 bits.
49
These results illustrate that for time-sharing systems, where average message length is on
the order of 1000 bits, packet switching can be very effective. However, for remote job
entry or file transfer traffic, where messages are several thousand bits long or more, and
should be sent at the same time circuit switching is more useful.
It is difficult to compare exactly between these techniques we hazard a few observation:
‫ ٭‬For interactive traffic, message switching is not suitable
‫ ٭‬For light or intermittent loads, circuit switching is the most cost effective, as the PSTN
can be used via dial-up lines.
‫ ٭‬For very heavy and sustained loads between two stations, a leased circuit switched line
is the most cost efficient.
‫ ٭‬Packet switching is to be preferred when there is a collection of devices that must
exchange
a moderate to heavy amount of data; line utilization is most efficient with this technique.
‫ ٭‬Datagram packet switching is good for short messages and for flexibility.
‫ ٭‬Virtual circuit packet switching is good for long exchanges and for relieving stations for
processing burden.
The following table gives a quick comparison between the three different switching techniques.
50
Comparison between different switching techniques
51
Sheet-2
1- A message with length 3200 bit is transferred over a communication network with 100 km hop
distance, and four intermediate nodes. The data rate on all links is 9.6 kbps, the propagation
delay per hop is 0.0015 sec., and total packet size 1024 bits. The set-up time is 0.2 sec., while
the processing time, and the average queuing delay at each node are 0.25, 0.35 sec.
respectively. Compare between the end-to-end delay time for; circuit, and datagram switching
techniques.
Solution: L = 3200 bit, D = 100 km, N = 4, C = 9.6 kbps, Prop. Delay/hop = 0.0015 sec.,
Total pkt size = 1024 bit, Tset-up = 0.2 sec.,
Tproc./node = 0.25 sec, Tque./node = 0.35 sec.,
For circuit switching;
Tend-to-end, CS = Tset-up + Tproc./ node (1) + Ttr.+ Tprop./hop (N+1)
= 0.2 + 0.25 + 3200/9600 + 0.0015 x 5
= 0.45 + 0.333 + 0.0075 = 0.791 sec.
For datagram, consider Pipelining;
Tend-to-end, DG =Tproc./ node (N+1) +Ttr.+Tprop./hop (N+1) +Tque/node(N+1)
st
Ttr. = Ttr. of 1 packet + Ttr. of the rest of the message
Tend-to-end, DG = 0.25x5 +{[(P+h)(N+1)+((L+H) – (P+h))(1)]/C}+ 0.0015 x 5 + 0.35x5
H = Summation of header sizes for all packets = number of packets x h = (L / P) x h
H = (3200 / 960) x 64 = 256 bit
Assuming P + h = 1024 = 960 + 64
Tend-to-end, DG = 1.25 +{[(1024)(5) + ((3200+256) - (1024))(1)]/9600}+ 0.0075 + 1.75
= 1.25 + [0.533 + 0.253] + 1.7575 = 3.8 second.
2- A switching network with 5 nodes between the source and the destination transferring
messages with 3200 bits in length. The data rate on all links is 9600 bps, with packet size 960
bits with 64 bits overhead address. The call set-up time is 0.2 sec., with propagation delay
0.0015 sec. per hop, calculate the end-to-end delay for the following (assuming no
acknowledgment);
i- circuit switching network,
ii- message switching network.
iii- virtual circuit packet switching network.
Solution: L = 3200 bit, P = 960 bit, h = 64 bit, N = 5, C = 9.6 kbps, Prop. delay/hop = 0.0015sec.,
a- For circuit switching;
Tend-to-end, CS = Tset-up + Ttr.+ Tprop./hop (N+1)
= 0.2 + 3200/9600 + 0.0015 x 6
= 0.2000 + 0.3330 + 0.0090 = 0.5420 sec.
b- For Message switching;
Tend-to-end, MS = Ttr.(N+1) + Tprop./hop (N+1) = ((3200 + Hm)/9600)(6) + 0.0090
where Hm is the message header assumed to be 200 bit as example;
Tend-to-end, MS = ((3200 + 200)/9600)(6) + 0.0090 = 2.125 + 0.0090 = 2.134 second
The storage, and processing times are neglected.
c- For Virtual Circuit; number of pkts = 3200 / 960 = 4
52
Tend-to-end, VCS = Tset-up + Ttr./pkt (number of pkts) (6) + Tprop./hop (6)
= 0.2 + (1024 / 9600)(4)(6) + 0.0090 = 0.2000 + 0.3330 +0.0090 = 2.769 sec.
The queue, and processing times are neglected although the processing and queue (storage)
times are neglected,
Tend-to-end, CS < Tend-to-end, MS < Tend-to-end, VCS
3- A message of 6000 byte is transferred through a network with 3 nodes between the source and
the destination. The data rate on all links is 96 kbps, with packet size 250 bit and 50 bits as
a header. The setup time is 0.2 sec, with processing time at each node is 0.025 sec, while the
average queueing delay at each node is 0.15 sec. The propagation speed over any link is
250 m / μ sec., with 60 km distance between each two nodes. Compare between circuit and
datagram packet switching end-to-end delay time.
Solution 3; For circuit switching;
TDCS = Tset-up + Tproc. + Ttr. + Tprop.
= 0.2 + 0.025 + L / C + (D / S)(N+1)
= 0.2 + 0.025 + 6000 x 8/96000 + (60000 / (250x106)) (N+1)
= 0.2 + 0.025 + 0.5 + 9.6 x 10 -4 = 0.72596 sec
For datagram switching;
TDDG = Tproc. + Tque. + Ttr.1st-pkt + Ttr. rest+ Tprop.
-4
TDDG = 0.025x4 + 0.15x4 + (p+h)x 4 / C + [(L+H) - (p+h)]/C + 9.6 x 10
-4
= 0.1+ 0.6 + (250+50)x4/96000 +{[(6000x8)+(6000x8x50/250)]-(250+50)}/96000 + 9.6 x 10
-4
= 0.7 + 0.0125 + 0.596875 + 9.6 x 10 = 1.31 sec
TDDG / TDCS = 1.31 / 0.72596 = 1.8
4- A message with 4500 bit is transferred over a communication network with 100 km hop
distance, and three intermediate nodes. The data rate on all links is 20 kbps, with speed
350 m/µ sec, and packet size 128 bits, with 52 bit overhead. The set-up time is 0.25 sec., while
the processing time, and the average queueing delay at each node are 0.25, 0.35 sec.
respectively. Compare between the end-to-end delay time and throughput for circuit, and
datagram switching techniques.
Solution 4, L = 4500 bit, P = 128 bit, h = 52 bit, C = 20 kbps,
D =100 km, N = 3, S = 350 m /µ sec., Tset-up = 0.25 sec.
Tproc.= 0.25 sec., Tqueueing at each node = 0.35 sec.
For circuit switching;
TDCS = Tset-up + Tproc. + Ttr. + Tprop.
= 0.25 + 0.25 + L / C + (D / S)(N+1)
= 0.25 + 0.25 + 4500/20000 + (100000/(350x106)) (4)
= 0.5 + 0.225 + 1.143 x 10-3= 0.726 sec,
For datagram;
TDDG = Tproc. + Tque. + Ttr.1st-pkt + Ttr. rest+ Tprop.
-3
= 0.25 x 4 + 0.35 x 4 + (p+h) x 4 / C + [(L+H) - (p+h)]/C + 1.143 x 10
Assume H = 300 bits
TDDG = 2.4 + (128+52) x 4/20000 +{[(4500) + (4500 x 52/128)] - (128+52)}/20000
+ 1.143 x 10 -3 = 2.436 + 0.31 + 1.143 x 10 -3 = 2.747 sec,
53
TDDG/ TDCS = 2.747 / 0.726 = 3.784
When packet switching technique (data gram) is delivery guaranteed using a mini packet of
70 bit, there is no pipelining but packets are transmitted in sequence. Each packet is
acknowledged before transmitting the following packet. The total delay time is calculated as;
Ttr., dg = Ttr.pkt * no. of pkts * (N+1)
= (180 / 20000) (4500 / 128) (4) = 1.266 sec.
Tprop., dg = Tprop. * no. of pkts * (N+1)
= (100000/(350x106)) (4500/128) (4) = 0.04 sec.
Tack = (Ttr. + Tprop.)mini pkt.* no. of pkts * (N+1)
= ((4.5 x 10 -3) + (0.286 x 10 -3))* 36 * 4 = 0.545 sec.
TDDG, dg = Tproc. + Tque. + Ttr.+ Tprop.+ Tack.
= 0.25 + 0.35x4 + 1.296 + 0.04 + 0.545 = 4.531 sec.
When the acknowledgement is negative packet re-transmission is required for some packets only.
Assuming the number of retransmitted packets are Nret., So the retransmission time is added with
a value of;
Tretr. = (Nret / 36) * (Ttr., dg+ Tprop., dg + Tack.)
For circuit switching datagram, set-up time is added to the total delay time.
Throughput for each case = L / (Ttr. + Tprop.)
5- It is required to transfer a message with size 8000 bytes from a source to destination with two
nodes between them. The data rate for all links is 56 kbps. The packet size is 200 bits and the
overhead address is 60 bits. The call set-up time is 0.1 sec. The processing time at each node is
0.15 sec., while the average queueing delay at each node is 0.2 sec, and the data speed over
a link is 200 m/µ sec. and the hop distance is 50 km. Calculate the end-to-end delay time, and
throughput for the following cases;
a- Circuit Switching network
b- Message Switching network
c- Virtual circuit packet switching network with delivery guaranteed using a mini packet of
70 bit.
d- datagram packet switching network without delivery guaranteed.
6- Consider the transfer of a file of 2 M bits from a source to a destination with three nodes
between them. The data rate of all inks is 9600 bps. The packet size is 48 bytes and the
overhead address is 5 bytes. The call set up time is 0.02 sec. The processing time at each node
is 0.006 sec while the average queuing delay is 0.03 sec. The propagation speed on the link is
200m /µsec, while the hop distance is 30 km. Calculate the end -to-end delay and throughput
for the cases:
a- Circuit switching network,
b- Message switching network.
c- Virtual circuit packet switching network with delivery guaranteed with a mini packet of
60 bits.
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Sheet -3
1- A message of 5000 bits is transferred through a switching network with 5 nodes between the
source and the destination. The data rate on all links is 10 kbps with packet size 256 bit and
80 bits as a header. The setup time is 0.2 sec. with processing time 0.1 sec at each node while
the average queueing delay at each node is 0.22 sec. The propagation speed on the link is
200 m/µsec with 800 km distance between source and destination. Compare between circuit,
message and virtual circuit switching end-to- end delay time without acknowledge.
2- It is required to transfer a message with a size of 80 k bytes on a communication network by
3 nodes, from the user to the server. The data rate on all links is 2 Mbps, the packet size 1 k bit
and 100 bits as a header. The setup time is 0.12 sec. with processing time 0.08 sec at each node
while the average queueing delay at each node is 0.2 sec. The propagation speed on the link is
500 m/µsec where the hop distance is 60 km. Calculate the end-to- end delay time and
throughput for;
a- circuit switching network
b- virtual-circuit packet switching network without delivery guarantee. Each packet is
acknowledged with a mini-packet of 50 bits.
c- Datagram packet switching networks without delivery guarantee.
3- A 64 kbps circuit switched path between two users utilizing 9 switching stages take 5 seconds
to be set-up. How long will it take, in total, to directly transfer the entire contents of
a 1.4 M byte floppy disk between the two users using this channel?
4- The above circuit switched path is replaced by a 9 node-packet switched system utilizing
1 M bps links. How long does it take, to transfer the 1.4 M byte file, assuming that there is no
delay at any of the switching nodes and that packet payload is 256 bytes, and packet overheads
are equivalent to 18 byte.
5- Consider a work station attached to a 1 Mbps local network that generates on average three
messages per minutes, with message averaging 500 bits. Calculate the connected number of
stations on this network. Comment.
6- Consider a set of stations generate PCM voice packets on a 10 M bps local network. Data are
generated at the rate of 64 k bps. For 0.1 sec. packets, calculate the break point number of
stations on the network.
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3.2 Numbering Plan

It is a hierarchical plan, and has an international standardized country code at the
highest level. The International Telecommunication Union (ITU) is a specialized agency
of the United Nations responsible for telecommunications. It has over 200 member
countries, and standardization work is divided between two major standardization bodies:
ITU-T and ITU-R. The Consultative Committee International for Telegraph and
Telephone (CCITT/ITU-T) presently called ITU-T, has therefore prepared a world
numbering plan. The object of this numbering plan is to allocate a distinct number to each
subscriber connected to the network. ITU-T and ITU-R publish recommendations that are
in fact strong standards for telecommunications networks. ITU-T works for the standards
of public telecommunications networks (e.g., ISDN), and ITU-R works with radio aspects
such as the usage of radio frequencies worldwide and specifications for radio systems.
Many parties participate in their work, but only national authorities may vote. ITU-T,
formerly CCITT, has created most of the current worldwide standards for public networks.
To set up a connection with another subscriber in the network, you begin by transmitting
the number of that called subscriber to your local exchange.
The number contains the information the exchange requires to perform two important
functions:
i- Selects the circuits that have to be connected to route the call through the network after
establishing it,
ii- Activates the necessary apparatus for proper call charging.
The number that must be dialed is determined by the numbering plan of the operating
administration. The route selected for setting-up the connection depends on the call
destination, and this must be determined by switching equipment from the number dialed.
The switching equipment also determines the charging rate for the call from its
destination, as given by the digits dialed. Numbering, routing and call charging are
therefore very closely related. It is essential that the plans developed by an administration
for these three aspects of its network operation should be consistent with each other.
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The plans must also be compatible with a national transmission plan and signaling plan.
For example, either transmission or signaling standards may limit the size of a local area,
or the number of links that may be used in tandem for a trunk connection.
So, the national plans for numbering, routine, charging, transmission and signaling thus
form a set of inter-related standards that govern planning both for a national network and
its constituent local networks.
International Subscriber Dialing (I.S.D.) makes it necessary for national numbering plans
to conform with the international plan. At first, each numbering scheme applied only to
a single exchange, and exchanges were identified by the names of their towns. Later,
linked numbering schemes were applied to multi-exchange areas. A linked numbering
scheme allows a uniform dialing procedure to be used between all subscribers in the
numbering area. The subsequent introduction of Subscriber Trunk Dialing (S.T.D.) for
long-distance calls required the development of national numbering plans.
Consideration should also be given to the pattern of traffic flow to make the dialing
procedure as simple and uniform as possible for the majority of calls. Since each
subscriber has a directory number in his own exchange, a unique national number can be
obtained by adding to this number a code identifying the exchange, as the country is
divided into geographical areas, each having a separate code and containing number of
local exchanges. The originating trunk exchange then only requires to identify the area
code to determine the routing and charging rate for each call.
ISD makes it necessary for every subscriber’s terminal in the world to be identified by
a unique number. The CCITT has therefore prepared a world numbering plan. The object
of this numbering plan is to allocate a unique number to each subscriber connected to the
network. The telephone system is organized as follows; the country is divided into number
of districts, each assigned its own code number. The sectors in a district are addressed by
its digit and the local networks by the third digit as given in figure 3.12. For incoming
long-distance traffic these networks are arranged within the national network as if they
were district exchanges. Outgoing long-distance traffic goes in the normal way through the
junction exchange to which the local network is connected. The STD code must always be
56
preceded by a zero to inform the local exchange that a long-distance connection is to be set
up.
Figure 3.12 National Telephone Network

3.2.1 Constraints on Numbering Plan
Each world telephone number consists of a country code that is preceded by the
international prefix (international gateway number), and followed by the subscriber’s
national number.
The existence of a world numbering plan places restrictions on the national numbering
plan of each country to enable subscribers in other countries to make international calls
into its national network. The constraints on the development of the numbering strategy
are:
1- It must provide each subscriber with a distinct number within the national network.
2- The dialing procedure should be simple and uniform.
3- It must be able to meet forecast growth of the subscriber density for several decades
(50 years).
4- The subscriber numbers must comprise all numerous and no letters.
5- The number of digits in a subscriber world number should be minimum and must not
exceed that recommended by CCITT.
It is therefore limited to an absolute maximum of 13. In practice, international numbers
57
are limited to 11 digits, excluding any code prefix digits. So, the maximum number of
digits available for a national numbering plan is thus (11 - nd) where nd is the number
of digits in the country code in the world numbering plan.
6- It is desirable to have numbering areas boundaries coincide with geographical or
political boundaries.
7- The boundaries of numbering areas should coincide with call charging boundaries.
3.2.2 International Number

For international availability of connections to be possible, each subscriber in the
world must have a unique number and each country must be identified by a number differ
from that of all other countries. That is achieved by agreement through CCITT on the way
that each country has its own code. The first digit of the country code is the ZONE code.
The world is divided up into nine (9) zones and each country belongs to a zone as
displayed in the following table. In all, but two, of the zones, one or two digits are added
to the zone number to produce the country code. Egypt has 20, UK is 44 and France is 33.
The two exceptions are zones 1 and 7. In Europe, there are many countries with large
national networks that have 9 digits in their national number.
The worldwide zone codes
Area code Area code
North America 1 Australia 6
Africa 2 USSR 7
Europe 3 Eastern Asia 8
Europe 4 Far &Middle East 9
South America 5 Spare 0
A two digits country code is required at the maximum for these countries. This can only be
achieved by allocating two zone numbers to Europe.
Additional prefix is used to differentiate between trunk calls and international calls. For
example, if the international prefix is "010", the first digit "0" causes the call to be routed
to the trunk exchange and the next two digits "10" causes the call to be switched to the
58
international gateway exchange. In dialing, the 3-digit prefix is followed by the

international code of the required country and the national number of the called subscriber
in that country. Therefore, the international number of a subscriber consists of the
following:
a-International Prefix (International Access Code)
The combination of digits to be dialed by a calling subscriber making a call to
a subscriber in another country to obtain access to the automatic outgoing international
equipment.
b- Country Code
The combination of one, two or three digits characterizing the called Country.
c- Trunk Prefix
It is a digit (usually '0') or combination of digits to be dialed by a calling subscriber
making
a call to a called subscriber in his own country but outside his own numbering area. It
provides access to the automatic outgoing trunk equipment.
d- The National Significant Number
The national significant number of a subscriber, shown in figure 3.13, comprises three
parts:
1-An Area Code (Trunk Code)
This identifies the particular numbering area of the called subscriber and thus
determines the route for a trunk call and its charge. It is a digit or more (not including
the trunk prefix) characterizing the called numbering area within a country.
The originating trunk exchange then requires to identify the area code. These areas may be
small or large. Each has its merits such as:
For Small Numbering Areas:
i- Junctions from local exchanges to trunk exchanges will be short and inexpensive.
ii- The number of trunk switching centers, however, will be large.
iii- Nearby exchanges having considerable traffic between them may be in different areas,
thus complicating the dialing procedure for these calls.
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For large Numbering Areas;

i- Large area necessitates long and expensive junctions from local exchanges to central
trunk switching centers.
ii- As large number of subscribers is contained, more digits are dialed for calls within the
area, but fewer digits for trunk calls to other areas.
2- An Exchange Code
This identifies the particular local exchange within the area and so determines the route
from the incoming trunk exchange of the area to the terminating local exchange. It is
a digit or a combination of digits of the subscriber number which identifies the local
exchange to which the called subscriber belongs.
3- The Subscriber 's Local Exchange Number
This selects the called subscriber's line at the terminating exchange. The number to
be dialed to reach a subscriber in the same numbering area is (Exchange Code +
Subscriber's Local Exchange Number) and is called the subscribe number.
For a within-area call, the calling subscriber dials only the exchange code followed by the
number of the called subscriber. For a trunk call, the calling subscriber dials a trunk prefix
digit (e.g., the digit '0') before the full national number of the called subscriber.
Recognition of the prefix causes the originating exchange to route the call to a trunk
exchange. By this means, only codes beginning with the prefix digit are excluded from the
area numbering scheme.
In calculating the length of the national number, the prefix is not included. The detailed
structure of a national significant number is illustrated in figure 3.13.
national significant number
0 XX XXX XXXX
Trunk Prefix area code exchange code sub's number
Figure 3.13. Elements of national significant number
Example:
Consider the following telephone number "01020483661334".
60
Let, us divide this number to its elements:
a- The first three digits "010" is the international prefix dialed to get access to the
international gateway.
The first digit "0" is the trunk prefix used to inform the local exchange that the
requested call is not a local call and therefore get access to the trunk exchange.
The following two digits "10" is to request the international access.

b-The following two digits "20" is the country code of the called subscriber, this is of
Egypt, and consists of digit "2” is the zone code of Africa, and digit ''0" is that code
assigned to Egypt within Africa zone. The remaining digits ''483661334" is the national
significant number of the called subscriber,
c- The two digits "48" is the area code (or trunk code) of the called subscriber which is the
code of Menoufia numbering area,
d- The following three digits 366 is the number of the local exchange of the called
subscriber with capacity 104 subscribers),
e- The remaining digits "1334" is the number of the called subscriber in the local
exchange.
Also registers to recognize area codes are only required in trunk exchanges. So, out of the
total of nine digits required to identify a subscriber within the national network, two digits
define the area, (Area Code, or Trunk Code), three define the Exchange within the area,
(Exchange Code), and four digits identify the particular subscriber. Short- code dialing
procedures are usually provided for certain classes of call to avoid always having to dial
long numbers.
The method can thus be used with non register local exchange. An additional prefix can
also be used to differentiate between trunk calls and international calls. For example, if the
international prefix is '010', the first digit (0) causes the call to be routed to a trunk
exchange and the next two digits (10) cause the call to be switched to the international
gateway exchange. In dialing, the 3-digit prefix is followed by the international code for
the required country and the national number of the called subscriber in that country.
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If numbering areas are large, wide variations will arise in the distances available for the
same charge, which may lead to complaints from subscribers. Large areas necessitate long
and expensive junctions from peripheral local exchanges to central trunk-switching
centers. Also, an area must not be so large that the number of subscribers it will ultimately
serve is beyond its numbering capacity. Numbering areas containing a large number of
subscribers necessitate more digits being dialed for calls within the area, but fewer digits
for trunk calls to other areas.
However, it is necessary to apportion the numbering range to the individual Exchanges so
as to allow for the different growth rates likely to be encountered. The numbering plan
must also allow additional exchanges to be added when the increase in telephone density
in particular locations makes this desirable. The numbering range available for individual
exchanges is reduced by the use of code digits for trunk and international prefixes and for
special services (e.g. operator assistance, inquires, telegrams etc.). For the convenience of
subscribers, these codes should be standardized throughout the country (e.g. the use of
'100 ' for operator assistance and '999' for emergency services).
In areas using register controlled Exchanges, either 2-digit or 3-digits exchange codes may
be used depending on the number of exchanges in the area.
3.3 Signaling Plan

Signaling is defined as the exchange of information specially concerned with the
establishment and control, maintain, and termination of connections and the transfer of
user-to-user management information on a communication network. Both call
management and overall network management require that information be exchanged
between subscriber and switch, among switches and between switch and network
management center. Basically, it informs a local switch that a subscriber desires service
and then gives it the data necessary to identify the required distant subscriber, with
supervising the call along its path. Signaling also gives the subscriber the required
information about specific status such as Dial and Busy or Ringing Tones. Signaling plans
interact closely with routing plans. If the signaling is slow and limited in repertoire, then
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call setting-up times can be excessive and the facilities provided by the network are
limited. Badly planned signaling can also result in excessive variety and complexity of
equipment. For local networks, the maximum length of subscriber's line (and the size of
the exchange area) is determined by signaling limits in addition to transmission limits.
Similar considerations apply to audio-frequency junctions which use d.c. signaling.
3.3-1 Functions of the Signaling System

Conventional signaling has evolved with the telephone network, so many of the
techniques to be dealt with are applicable for telephone calls transfer over
a communication network. The signaling system should carry out the following functions;
Supervisory, Address, Call Forwarding audible-visual.
3.3-1.1 Supervisory Signaling
Supervisory signals provide information about the state of the call, and line or circuit
condition. It informs a switch whether a calling party is off-or on-hook, whether a circuit
(internal to the switch) or a trunk (external to the switch) is busy or idle, and when a called
party is on-or off-hook.
Supervisory information (status) must be maintained end-to-end on a telephone call,
whether voice, data, or facsimile is being transported over a particular path that marked
busy. It is necessary to know when calling subscriber lifts her/his telephone handset,
thereby requesting service. It is also important to know when the called subscriber answers
(i.e., lefts the telephone handset) to start charging the call. Also, it is required to know
when the calling and called subscribers return their telephone handset to the on-hook state,
to stop charging the calling subscriber, and inform the talk path as well as the switching
points to be ready for use by another pair of subscribers. Supervisory control signals are
used to check whether a needed resource is available and, if so, it seizes it. They are also
used to communicate the status of requested resources. Therefore, supervision signal deals
with detecting the condition or changing the state of network elements either lines or
trunks such as;
i- Detecting the Condition
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When a calling subscriber goes off-hook indicating a request for service, this completes
the loop with his local exchange that alerted by detecting the line current. When the
subscriber finishes a call, he replaces the handset (goes on-hook state), the loop is
disconnected and the exchange detects the disappearance of the loop current. In addition to
these signals from the subscriber to the exchange, signals are also required in the reverse
direction to inform the subscriber of the state (free or busy) of the exchange or the called
subscriber. These signals are normally audio-frequency tones (e.g., busy tone, ringing
tone, …).
The exchange performs the following functions;
1- Recognize the calling subscriber condition.
2- Connect register equipment to the subscriber line for receiving address (dialed)
information.
The register reverts dial tone in the handset stays for short period, signaling the calling
subscriber, that the exchange is ready to receive dialed information. The subscriber
then dials digits (called subscriber address).
Equipment at the exchange senses the frequencies of the tones and determines the dialed
digit.
3- The dialed digits stored in the register to select the required connection. If all registers
are busy, the exchange connects the calling subscriber line to busy tone instead. On
hear in this tone, the subscriber replaces the handset, and subsequently makes another
attempt,
4- Register receives dialed digits, and obtains a connection to the translator that provides
the required information about the require outgoing route,
5- If the called subscriber is not busy, the switch alerts that subscriber to an incoming call
by generating a ringing signal (ring current),
6- Feedback is provided to the calling subscriber by the switch:
a- If the called subscriber is not busy, the switch returns an audible ringing tone to the
caller, while the ringing signal is being sent to the called subscriber,
b- If the called subscriber is busy, the switch sends an audible busy signal to the caller,
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which consists of bursts, with frequency 425 Hz. The ON-time equals the OFF-time
which lasts 0.5 second, to indicate that the called terminal is busy, or lasts 0.25
second to indicate that the distant exchange is not ready to proceed the call to the
called subscriber.
c- If the call can not be completed through the switch, the switch sends an audible
recorded message to the caller.
7- The called party accepts the call by lifting the handset (off-hook), which is
automatically signaled to the switch.
8- The switch terminates the ringing signal and the audible ringing tone, and establishes
a connection between the two subscribers.
9- The connection is released when the calling subscriber hangs up.
When the called subscriber is attached to a different switch not that of the calling
subscriber, the following switch to switch trunk signaling functions are required:
1- The originating switch seizes an idle inter-switch trunk, sends an off-hook indication on
the trunk, and requests a digit register at the far end, so that the address may be
communicated.
2- The terminating switch sends an off-hook followed by an on-hook signal, known as
a "wink". This indicates a register-ready status.
3-The originating switch sends the address digits to the terminating switch.
ii- Change the State

Assuming satisfactory completion of the subscriber to local-exchange connection, the
calling subscriber marked as busy. The next stage in the establishment of the connection
involves the completion of the inter-exchange connections, by a seizure of a direct route if
possible otherwise a second attempt is made via the trunk exchanges, which goes from idle
to busy condition. The next stage is to send forward the necessary information to
succeeding exchanges to enable the call establishment to proceed further.
3.3-1.2 Address signaling
65
This includes functions (the dialing of the digits represent the called subscriber
number) used on call set up to determine exact identification and location of the called
subscriber. Address functions actuate control circuits in the local switch allowing
a connectivity to be set up. When the called subscriber is outside the local exchange of the
calling subscriber, the call will be connected to an outgoing trunk, that is a transmission
pathway that interconnect switches.
Also addressing signaling is associated with the information interchange between
exchanges, and ended by releasing the call set up, and sending ringing current.
3.3-1.3 Call Forwarding

The return of the busy-back signal by the serving exchange is called call progress
(forwarding) signaling. These functions serve to guarantee an efficient use of available
facilities and provide information on calls established in order to:
a- Detect and transmit information regarding congestion of the network,
b- Inform of unavailability of equipment units or of circuits as a consequence of
breakdown or outage of maintenance.
c- Collect data for service charging,
d- Send calculated tariff information to register equipment.
e- Provide a method of sending alarm information for unattended exchanges.
3.3-2 Signaling Techniques:

3.3-2.1 In-Channel Signaling (Per-Trunk Signaling) Traditional control signaling in
circuit-switched networks has been on a per-trunk or in-channel basis. With in-channel
signaling, the same channel is used to carry control signals as is used to carry the call to
which the control signals relate. Such signaling begins at the originating subscriber and
follows the same path as the call itself. This process has the merit that no additional
transmission facilities are needed for signaling; the facilities for voice transmission are
shared with control signaling. Two forms of in-channel signaling are in use;
a- In-Band Signaling:
66
It uses not only the same physical path as the call it serves; it also uses the same
frequency bands as the voice signals that are carried as shown in figure 3.14-a. Therefore,
signaling information is transmitted in the same band of frequency used by the voice
signal. This form of signaling has several advantages. Because the control signals have the
same electromagnetic properties as the voice signals, they can be used on any transmission
medium and go anywhere that the voice signals go. Thus, there are no limits on the use of
in-band signaling anywhere in the network, including places where A/D or D/A
conversion takes place.
In addition, it is impossible to set up a call on a faulty speech path, as the control signals
that are used to set up that path would have to follow the same path.
The main disadvantage of this technique is the need to eliminate mutual interference
between signaling waveform and speech.
Figure 3.14-a In-band Signaling
b- Out-of-Band Signaling
It takes the advantage of the fact that voice signals do not use the full 4-K Hz
bandwidth allotted to them. A separate narrow signaling band within the 4-K Hz is used to
send control signals as shown in figure 3.14-b. Therefore, this technique uses the same
facility as the voice channel but at a different portion of the frequency band. It represents
a form of FDM within a single voice channel. The most common one is the dc signaling
used on most customer loop. The central office recognizes off-hook condition by the flow
of dc current in the line signals uses lower frequencies than those generated by speech.
67
However, there is also out-band signaling that uses a frequency (3825 Hz) above the
speech band (e.g., above 3400 Hz) but below the upper limit of the nominal voice channel
(4000 Hz). The major advantage of this scheme is that there is no possibility of one being
mistaken for the other (i.e., there is no interference between voice and control signals).
However, an out-of-band signaling depends on the transmission system used.
For example, SSB carrier systems filter out the very low frequency associated with
each voice channel. So, the control signal must be converted to something else and
therefore extra electronics is needed to handle the signaling band. Also, the signaling rates
are slower because the signals have been confined to a narrow bandwidth.
Drawbacks of In-Channel Signaling
As public telecommunication networks become more complex, and provide a richer
set of services, the drawbacks of in-channel signaling become more apparent such as;
Figure 3.14-b Out of band Signaling
i- The information transfer rate is quite limited with in-channel signaling. With in-band
signaling, the voice channel being used is only available for control signals when there
are no voice signals on the circuit. With out-of-band signaling, a very narrow
bandwidth is available.
With such limits, it is difficult to accommodate, in a timely fashion, any but the simplest
form of control messages. A more powerful control scheme is needed.
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ii- A second drawback of in-channel signaling is the amount of delay from the time
a subscriber enters an address (dials a number) to when the connection is established.
The requirement to reduce this delay is becoming more important as the network is used
in new ways. For example, computer controlled calls, such as with transaction
processing, use relatively short messages, therefore, the call set up time represents an
appreciable part of the total transaction time. Both of these problems can be addressed
with common channel signaling.
3.3-2.2 Common Channel Signaling

Control signals are carried over paths completely independent of the voice channels
and exclusively dedicated for signaling. One independent control signal path can carry the
signals for a number of subscriber channels and, hence, is a common control associated
with this group of channels. Bell system installed the first in 1976. It is called Common
Channel Interoffice Signaling (CCIS). There are two types of common channel signaling
in existence. European CCITT No. 6 and the North American CCIS. Telephone companies
were moving toward use of common channel signaling equipment to replace older in-
channel signaling equipment. The complete separation of signaling information from user
data greatly increases the flexibility of signaling, makes it possible to offer a wide variety
of additional functions, and has helped to significantly reduce average circuit connection
times below those achieved without common channel signaling. The initial CCS
equipment, based on CCITT recommendations for CCSS number 6 (CCSS #6), used
analog trunks with 2400 bps modems. CCSS #6 is widely used, but it is being supplanted
by CCSS #7 using purely digital trunks at 56 or 64 kbps. CCSS #7 is the ISDN version.
The "principle of CCS is illustrated and contrasted with in-channel signaling in figure
3.15. As can be seen, the signal path for CCS is physically separate from the path for voice
or other bus signal. The common channel can be configured with the bandwidth required
to carry control signals for a rich variety of functions. Thus, both the signaling protocol
and the network architecture to support that protocol are more complex than in-channel
69
signaling. However, the continuing drop in computer hardware costs makes CCS
increasingly attractive.
The control signals are digital messages that are passed between switches as well as
between a switch and the network management center. Thus, the control signaling portion
of the network is, in effect, a distributed computer network carrying short messages.
(b) Separate Channel Signaling

Figure 3.15-a- In-channel Signaling
b- Common-channel Signaling
a- Modes of Operation
Two modes of operation are used in CCS. These are the associated and non-associated
modes shown in figure 3.16.
70
Figure 3.16-a Modes of Common-channel Signaling
In the associated mode, the common channel closely tracks along its entire length the
inter-switch trunk groups that are served between endpoints.
Thus, a separate channel carries signaling information and this channel is routed with the
speech channels it serves. In other words, the control signals are on different channels
from the subscriber signals, and inside the switch the control signals are routed directly to
a control signal processor.
A more complex, but more powerful, mode is the non-associated mode. With this mode,
the network is augmented by additional nodes known as signal transfer points.
Figure 3.16-b Modes of Common-channel Signaling

There is now no close or simple assignment of control channels to trunk groups. In effect,
there are now two separate networks, with links between them so that the control portion
of the network can exercise control over the switching nodes that are servicing the
subscriber calls. With the fully non-associated signaling, the signaling information
71
traverses a routing entirely separate from the voice paths it controls. The signal-transfer
points, used in North America, consists of a processor with signaling terminals and data
modems (in case of CCSS #6) on either side. Signal transfer points allow concentration of
signaling for a large number of trunks and will also provide for circuit viability by
allowing alternative routing of the CCIS signaling path. To improve viability (reliability of
communications), all CCIS signaling paths in North America are duplicated and fully
redundant. In addition, all STPs are fully interconnected to provide several signaling –
path possibilities. The STPs are over-dimensioned, so that either STP can carry the full
signaling load if one STP fails. Such a network has a hierarchical structure, with primary
and secondary centers. Of course, it is fully non- associated.
For cost effectiveness, associated signaling is generally used with large trunk groups.
The concentration aspects of non-associated signaling would make it more attractive for
small trunk groups in the long-distance (toll) network.
b- Advantages of CCS
1- With in-channel signaling, control signals from one switch are originated by a control
processor and switched onto the outgoing channel. On the receiving end, the control
signals must be switched from the voice channel into the control processor. With
common channel signaling, the control signals are transferred directly from one control
processor to another, without being tied to a voice signal.
This is a simpler procedure, and one of the main motivations for CCS as it is less
susceptible to accidental or intentional interference between subscriber and control
signals.
2- Another key motivation for common-channel signaling is that call set up time is
reduced. Consider the sequence of events for call setup with in-channel signaling when
more than one switch is involved. A control signal will be sent from one switch to the
next in the intended path. At each switch, the control signal cannot be transferred
through the switch to the next leg of the route until the associated circuit is established
through that switch. With CCS, forwarding of control information can overlap the
circuit set-up process.
72
3- With non-associated signaling, a further advantage emerges; one or more central control
points can be established. All control information can be routed to a network control
center where requests are processed and from which control signals are sent to switches
that handle subscriber traffic. In this way, requests can be processed with a more global
view of network conditions.
c- Disadvantages of CCS
Of course, there are disadvantages to CCS; these primarily have to do with the
complexity of the technique. However, the dropping cost of digital hardware and the
increasingly digital nature of telecommunication networks makes CCS the appropriate
technology. Even in a network that is completely controlled by CCS, in-channel signaling is
needed for at least some of the communication with the subscriber. For example, dial tone,
ring back, and busy signals must be in channel to reach the user.
3.4 Capacity Plan

Capacity planning is an extension of the network dimensioning phase and achieved in
parallel with the quality planning. The aim of capacity planning is to define the maximum
and required number of exchanges at each physical location over a certain area. The exact
needed capacity can be defined based on the monitoring of the network which gives the
real information about traffic load on the network. Thus, the result of capacity planning is
actually a specification of the required input information about the traffic load
characteristics, and the required exchanges configurations to achieve the required capacity
in the most cost-efficient way.
The target is therefore to calculate whether the number of the channels in each exchange is
enough for the required capacity. The available number of channels is a critical network
planning parameter where some basic decisions are dependent on it.
The key parameter for the capacity planning is the total traffic over the planning area. The
aim is to divide the traffic load homogeneously to all exchanges inside the planning area
and thus to use an equal number of channels at each exchange. Capacity planning starts by
73
specifying the target traffic load (for example from monitoring results) and the minimum
average exchanges number to cover the service area.
The better the traffic forecasts the better the capacity configuration can be optimized for
network evolution.
In order to study the capacity requirements for a specific service area several more detailed
parameters need factoring into the dimensioning, specifically knowledge of the:
• Size of the service area
• Total traffic over the service area
• Targeted maximum blocking
• Number of available (required) channels
The required number of exchanges can be tuned to correspond to the number accordingly
based on traffic requirements for the service area. Because traffic is increasing year after
year, this analysis has to be done based on differing traffic demands, with the final
network configuration and deployment strategy dependent on this long-term analysis. If
this long-term analysis is not done the exchanges will not be located correctly and the
network configuration will not be cost-efficient due to overcapacity or load and will
require continuous reconfigurations. Note that reconfigurations can not be avoided, but
they can be minimized. Steadily increasing demands upon service area (for example
increased data traffic and data service) cause changes, as in blocking thresholds, and these
changes have to be taken into account at the dimensioning stage because they strongly
influence exchanges locations.
The preliminary capacity planning has already been done in the dimensioning phase of the
network planning process. In that phase only rough figures in the area type level were
estimated without going into the capacities of individual exchanges. The planning
parameters came from customer requirements and the estimated number of users. The user
estimates are for every network roll-out phase.
The capacity requirement per user is different depending on the user profile. The
dimensioning can be simplified by having one user profile per area type; i.e. all the users
inside the same area type have a similar user profile. The user profiles define the average
74
usage as well as the busy hours. The capacity has to be planned based on the maximum
simultaneous usage.
In the capacity planning phase a detailed capacity per exchange level is estimated. The
prior task was to select the exchange locations and calculate the service area using actual
exchange parameters. The capacity allocation is based on these density maps and traffic
estimates, which can be a separate layer on the map of the planning tool.
As stated, the maximum simultaneous usage is the main planning target for the network
capacity.
The capacity peaks define the fate of unsuccessful call attempts due to lack of resources
known technically as the Grade of Service, GoS. It is defined in the ITU-T
Recommendation E.600 as: A number of traffic engineering variables to provide
a measure of adequacy of a group of resources under specified conditions. These grade of
service variables may be probability of loss. This parameter which should be in accepted
level has already been defined by the customer at the beginning of the planning process.
The amount of traffic is expressed in Erlangs, which is the magnitude of telecom. traffic.
The setting of grades of service targets (i.e., the proportion of calls or messages
allowed to fail due to congestion) is a matter for subjective judge, based on determining
the right balance between the service to be given to the subscriber and the cost of
providing that service. Internationally, it has been accepted that the only practicable way
of ensuring an acceptable service is to specify the grade of service to be given by each
individual switching node and connecting link. Grades of service for international links
themselves are laid down by the CCITT. However, those applicable to a national network
are left to the individual administration and are very dependent upon the arrangement of
the network and the distribution of calls over it.
In deciding the grade of service appropriate to each link in the national network, the
function of the link as well as its cost should be taken into account. Some links may be
used for single-link traffic only, but others may carry traffic passing over a number of
links in tandem.
75
The volume of traffic to be carried by the link is also important in relation to the overall
grade of service given to the subscribers. A poor grade of service on a small unimportant
route will matter far less than on a large route. Thus, while total expenditure is minimized
by prescribing a worse grade of service for the most costly circuits, consideration of the
relative importance in the network of the link concerned may lead to a contrary decision.
Grades of service may be prescribed as a given proportion of lost calls, independent of the
number of circuits. However, although a large group of circuits carries traffic more
efficiently than a small group, its grade of service deteriorates more quickly with traffic
overload. Thus, at the normal loading the larger the number of circuits the better the grade
of service. This method thus recognizes the relative importance of large routes, in addition
to providing an adequate overload margin.
From the user’s point of view, the grade of service of the network is of prime importance.
Teletraffic design formula is used to predict accurately the behavior of the network during
operation. Telecommunication network is dimensioned to meet the maximum usage
demand period known as Busy Hour, BH. Normally there are two periods where, PSTN
traffic demand is maximum, one in the morning and one in the afternoon, as shown in
figure 3.17.
Figure 3.17 Busy Hour traffic
76
The figure indicates that PSTN is very inefficient as it approaches total capacity utilization
for only several hours during the working day. After 10 P.M. and before 7 A.M. capacity
utilization may be 2 – 3 % .
Busy hour traffic intensities are used to dimension the number of trunks required on
a connectivity as well as the size of (a) switch(es) involved. Now a PSTN company
(administration) can improve its revenue versus expenditures by cutting back on the
number of trunks required and making switches "smaller". Of course, network users will
do a lot of complaining about poor service. Let's just suppose the PSTN does just that cuts
back on the number of circuits. Now, during the BH period, a user may dial a number and
receive either a voice announcement or a rapid-cadence tone telling the user that All
Trunks are Busy (ATB) and to try again later.
From a technical standpoint, the user has encountered blockage. This would be due to one
of two reasons, or may be due to both causes. These are: insufficient switch capacity and
not enough trunks to assign during the BH. Networks are sized /dimensioned for a traffic
load expected during the BH. The sizing is based on probability, usually expressed as
a decimal or percentage. That probability percentage or decimal is called the grade of
service, which defined as; "the proportion of total calls, usually during the BH, that cannot
be completed immediately or served within a prescribed time."
Grade of service and blocking probability are synonymous.
Here we are going to introduce the principles of classical teletraffic theory used in the
Fixed Terrestrial Network.
The stochastic processes involved in telephone messages, flowing through a network is
called teletraffic theory. The theory is intimately connected to probability theory, as we are
concerned with the probability of calls being blocked, forced to terminate, and so forth, and
how these relate to efficient utilization of system capacity. Like any other fields, teletraffic
theory has its own technical terms and teletraffic design formulas.
The fate of call attempt depends on the used system. There are two types of systems deal
with traffic load (offered traffic).
77
First system is known as loss system, where call attempts arrive the system at the
instant where there are no free channels, leave the system without any subsequent effect
on the system performance. Traffic system serving call trials on this basis is known as
blocked calls cleared (BCC) or lost calls cleared (LCC) system.
Second system is known as Delay system, where call attempts arriving the system at
the instant where all the channels are busy do not leave the system but they are allowed to
stay in a queue waiting for free channels. This system is known as Blocked Call Delayed
(BCD) system.
Starting with loss system analysis where the attempted call rate in a BCC system with
N channels depends on the number of users generating the offered traffic. If the number of
users is assumed to be infinite, the arrival call rate is independent on the number of busy
users, the traffic is said to be random, where the arrival rate is randomly distributed, that
follows Poisson distribution, such as;

P(r) =  t  e   t
r
(3.5)
r!
It defines the probability of arriving r call trials during period of time, t with average
arrival rate λ.
For finite number of sources, the arrival call rate will depend on the number of busy
users and the traffic is said to be quasi-random. The analysis of a BCC system under these
two assumptions is different and results in two different teletraffic models known as
Erlang loss formula and Engset formula, respectively.
3.4-1 Loss System with Infinite Number of Users

The block diagram of such system is shown figure 3.18. Suppose we have a network with
N channels and the call arrivals are independent. The arrival and termination of a call can
be represented by a birth and death processes respectively. The number of carried calls in
the network is represented by states j = 0,1,2,... N, where j = 0 means there are no calls
being handled by the network, and J = N means that all the N channels are busy, and the
system is blocked (congested).
78
To develop a mathematical model to analyze the properties of the network, we consider

the call arrival rate when the network is in state j is λj, and the departure rate is μj.
The state-transition diagram of the loss system with infinite number of users is shown in
figure 3.19.
Figure 3.18 LCC system with infinite number of users.
Figure 3.19 State-transition diagram for the Birth-Death process.
We are interested in the steady state condition, which is reached under statistical
equilibrium. The statistical equilibrium means that chance of finding the network in any
specified state is the same.
λ : average call arrival rate expressed in calls per sec.

 : average service rate or departure rate = 1 , where Tc is the call duration, which is
Tc
randomly distributed, that follows the negative exponential distribution, with mean value
Tc.
Equivalently, the statistical equilibrium state probabilities are defined when the probability
to move to state j is equal to the probability of leaving state j;
λj-1P(j-1) + μj+1P(j+1) = ( λj + μj) P(j) (3.6)
Changing ;
79
λj = λ, j = 0, 1, ….. N-1.
μj = jμ, j = 0, 1, ……. N. (3.7)
P(-1) = P(N+1) = 0
For j = 0 μ P(1) = λ P(0)
 P(1) =  P(0) = A P(0)


where A =  = λ TC is the Offered Traffic, (traffic intensity, or volume), defined as the

average number of incoming call trials during the average call duration TC.
The preferred unit of traffic intensity is the Erlang, named after the Danish mathe-
matician A. K. Erlang (Copenhagen Telephone Company, 1928). The Erlang is a dimen-
sionless unit. One Erlang represents a circuit occupied for 1 hour.
A A2
Similarly, with j = 1, P(2) = P(1) = P(0)
2 2
Aj
By iteration P(j) = P(0)
j!
which describes the probability of finding the system at state j, i.e.; the probability of
finding J calls in progress, or J channels busy.
P(0) is obtained by setting the summation of all probabilities to unity, such that;
1
 N j
P(0) =   A 
 j  o j ! 
The formula of such loss system is known as Erlang loss or Erlang-B formula. In this
formula the blocking probability is expressed as:
PB = P(N) = AN / N ! (3.8)
N
 A j / j!
j 0
80
From the above discussion it is clear that the congestion problem can be solved by
computation, from graphs, or tables, because it is difficult to handle it. The following table
3.1 may be used to analyze the probability of blocking PB given by equation 3.8.
Table 3.1 LCC system with infinite number of users
81
The Lost Traffic, defined as the average number of lost calls during the average call
duration will be;
AL = A PB (3.9)
By the same way the Carried Traffic defined as the average number of carried (successful)
calls during the average call duration is calculated as;
AC = A (1 – PB) (3.10)
Formula 3.8 describes two types of system congestion, the first is time congestion which is
defined as the (fraction) percentage of time during which all channels are occupied, (system
is blocked or congested), and the second is the call congestion which refers to the
probability that a call arrives during the congestion period and hence is lost.
The average duration of the congestion (blocking) state, and the rate of its occurrence are
calculated as;
82
By considering that, the offered traffic A, corresponding to the average number of call
trials arriving the system during average call duration. Tc . So for very long period of time
T` , the average number of calls will be;
A` = A ( T ` / Tc )
So, the number of occasions the system state changes from (N-1) to N busy channels due to
a call arrival during the period T` is, A P (N-1). The total time during which N channels are
occupied (system is blocked) during the long period of time T` is, T` P(N). Then the
average duration of the congestion state is calculated as the ratio of the total duration of
congestion during T`, to the number occurrence of congestion during the same period T`,
giving;
TB = T` P(N) / A` P(N-1) = Tc / N (3.11)

The rate of occurrence of congestion is;
λB = λ P (N - 1) (3.12)
3.4-2 Loss System with Finite Number of Users

In a network with finite number of users (but still greater than the number of
channels), the probability of a call arrival during an interval will not be independent of the
network state because it will depend directly on the number of idle users. Idle users only
generate new requests during this time. Therefore, the probability of call arrival may
fluctuate considerably with the state of the traffic.
The quasi-random arrivals used here is in contrast to the Poisson random arrivals assumed
for the case of the Erlang-B formula while negative exponential holding time distribution
is assumed. Figure 3.20 shows the block diagram of such system, with a call rate λ j,
generated by a finite number of users, offered to a BCC system having N channels. The
arrival call rate λ j depends on the number of engaged users j.
A lost call means that the number of users who can contribute to the offered traffic is
increased by unity.
83
Figure 3.20 BCC system with finite number of users.
Consider that the number of users is M and that each user independently generates
requests at any time at rate u when idle and rate zero otherwise. Using the state transition
diagram of such system given by figure 3.21:
Figure 3.21 State transition diagram of finite number of users LCC system.
(3.13)
where u is the arrival rate per free user. When j calls are in progress (busy users), then
(M-j) users are free at that instant, where any one of them is ready to make a call trial.
Assume state j is in statistical equilibrium, then;
[M- (j-1)]u P(j-1) + (j+1) P(j+1) = [(M-j)u + j] P(j)
dividing both sides by , with u /  = a, describes the traffic offered by a free user, or the
probability of a call trial generated by a free user, then;
[M - (j-1)] a P(j-1) + (j+1) P(j+1) = [(M-j) a + j] P(j) (3.14)
changing j from 0 to N, with P(-1) = P(N+1) = 0, then
for j = 0, P(1) = M a P(0)
84
for j = 1, P(2) = M  1 a P(1)

2
=
M ( M  1) a 2 P(0)
2
and generally,
P(j) =  M  aj P(0)
 j 
M  N
 a
So, P (N) =  N  (3.15)
N M 
   ak
k 0  k 
which is the probability that all the channels are busy; and hence the system is blocked
(the time congestion),
N M 
where, P(0) = [ ak ] –1 (3.16)
  
k 0  k 
Time congestion does not necessitate call congestion because although all the channels
may be busy, calls may arrive during this interval. The call congestion or the probability of
call blocking (loss) is the conditional probability of call arrival during the congestion
(blocking) time and is given by;
(3.17)
which known as “ Engset equation “.

Practically, the parameter a is an inconvenient quantity to determine. It can be shown that,
a = A / M – A (1 - PB) (3.18)
where A is the total traffic offered.
Equation 3.17 may be solved using computer or tables such as the following table 3.2
85
Table 3.2 LCC system with finite number of users, N = 10

Users A, PB = 0.005 A, PB = 0.01 A, PB = 0.05
∞ 3.96 4.46 6.22
50 4.16 4.64 6.4
25 4.6 5.1 6.8
20 4.81 5.32 7.06
15 5.27 5.78 7.41
12 5.97 6.47 8.02
10 10 10 10
Lost Calls Retry

In the lost calls cleared analyses, it is assumed that unserviceable requests leave the system
and never return. This assumption is most appropriate for trunk groups whose blocked
requests overflow to another route and are usually serviced elsewhere. However, lost calls
cleared analyses are also used in instances where blocked calls do not get serviced
elsewhere. In many of these cases, blocked calls tend to return to the system in the form of
retries. Some examples are subscriber concentrator systems, corporate tie lines and PBX
trunks, calls to busy telephone numbers, and access to WATS lines (if alternatives are not
used). To derive blocking probability relationships for lost calls cleared systems with
random retries, the following analysis involves three fundamental assumptions regarding
the nature of the returning calls:
l. All blocked calls return to the system and eventually get serviced, even if multiple
retries are required.
2.The elapsed times between call blocking occurrences and the generation of retries are
random and statistically independent of each other. (This assumption allows the analysis
to avoid complications arising when retries are correlated to each other and tend to
cause recurring traffic peaks at a particular waiting time interval.)
3.The typical waiting time before retries occur is somewhat longer than the average
holding time of a connection. This assumption essentially states that the system is
86
allowed to reach statistical equilibrium before a retry occurs.

Obviously, if retries occur too soon, they are very likely to encounter congestion since the
system has not had a chance to "relax." In the limit, if all retries are immediate and
continuous, the network operation becomes similar to a delay system. In this case,
however, the system does not queue requests-the sources do so by continually "redialing."
When considered in their entirety, these assumptions characterize retries as being
statistically indistinguishable from first-attempt traffic, (First-attempt traffic is also
referred to as demand traffic: the service demands assuming all arrivals are serviced
immediately. The offered traffic is the demand traffic plus the refries). Hence blocked
calls merely add to the first-attempt call arrival rate.
Consider a system with a first-attempt call arrival rate of λ, If a percentage PB of the calls
is blocked, PB times λ retries will occur in the future. Of these retries, however,
a percentage PB will be blocked again. Continuing in this manner, the total arrival rate λ',
after the system has reached statistical equilibrium can be determined as the infinite series;
λ' = λ + PB λ + PB2 λ + PB3 λ + PB4 λ +…= λ / (1 - PB)
The following figure shows the relationship between lost calls cleared and lost calls retry.
Blocking probability of lost calls retry.
87
Sheet -4
1- Assuming each of 10,000 subscriber lines originates one call per hour, how often do
two calls arrive with less than 0.01 sec. between them?
Solution: The average arrival rate is;
λ = 10,000 / 3600 = 2.78 arrivals / sec.
The probability of no arrival in a 0.01 sec. interval is;
P(0) = e - λ t = e – 2.78 (0.01) = e – 0.0278 = 0.973
P(two calls arrive with less than 0.01 sec. between them) = 1- P(0) = 1 - 0.973 = 0.027
Thus, 2.7 % of the arrivals occur within 0.01 sec. of the previous arrival. Since the arrival
rate is 2.78 arrivals per second, the rate of occurrence of inter-arrival times less than 0.01
sec. equals;
2.78 x 0.027 = 0.075 times / sec.
2- A message-switching node that normally experiences four arrivals per minute, what is
the probability that eight or more arrivals occur in 30 sec. interval?
Solution: The average number of arrivals in a 30 sec. interval is;
λ = 4 x 30 / 60 = 2 arrivals / sec.
The probability of eight or more arrivals (when the average is 2) is;
 7
P>8 (2) =  Pi (2) = 1 -  Pi (2)
i 8 i 0
 21 2 2 23 27 

= 1 – e-2 1    .....  = 0.0011

 1! 2! 3! 7! 
3- What is the probability that a 1000 bit data block experiences exactly four errors during
transmission over a transmission link with a bit error rate (BER) of l0-5?
Solution: Assuming independent errors (a questionable assumption on many transmission
links), we can obtain the probability of exactly four errors directly from the Poisson
distribution. The average number of errors (arrivals);
λt = 103 x 10-5 = 0.01. Thus,
P(4 errors) = P4 (0.01) = 0.01 e 0.01  4.125 x 10-10
4
4!
4- An Exchange with 12 channel handling 200 call trials per hour with 2 minutes expected
duration. What is the probability that two calls are offered during 1 minute. If the lost
traffic is offered to another switch with 6 channels, what is the probability that both
switches are blocked simultaneously.
5- During a busy hour ‘A’ Erlang is offered to a single channel Exchange that was
occupied over a period of 49 minute. When the lost traffic is overflowed to a multi-
88
channel Exchange, it was blocked during 1.2 minute. Determine;

a- the average number of busy channels,
b- the probability of finding 2 channels are free,
c- the average blocking period if one call is offered every 40 seconds.
6- A group of users using an Exchange each with average period of 200 second, the
probability that non user using the Exchange during a period of 100 second is 8 %, and
the probability of founding non channel free is 10 %. Show that to achieve 0.2 blocking
probability a number of channels more than double the average number of calls must be
used.
7- The state probabilities of an Exchange with four channels are as follows; P(0) = 0.12,
P(1) = 0.15, P(2) = 0.2, P(3) = 0.5. Calculate the channel efficiency.
8- a- Consider A Erlang traffic offered to a single channel, show that the lost traffic value
equals the carried traffic multiplied by the offered traffic.
b- Two calls were terminated every 4 minutes in an exchange with 6 channels to give
0.02 grade of service. Calculate:
1-The average number of call trials entered the switch during a period of 4 minutes.
2-The probability that all channels are free in the switch.
3- The channel efficiency.
4-The number of channels required when the traffic is doubled with the same grade
of service. Comment.
9- There are 20 people in an office. During the course of an 8 hours day, they each
attempt to make an average of 16 telephone calls which last an average of 3 minutes
each. There are three telephone lines in the office. What is the probability of loss.
Solution 9; A = 20 x 16 x 3 min / 8 x 60 = 2 Erlang
PB(2,3) = (23/3!) / (20/0! + 21/1! + 22/2! + 23/3!)
= (8/6)/(1/1+2/1+4/2+8/6) = 4/19 = 0.2105
we can also obtain this result iteratively,
PB(2,0) = 1
PB(2,1) = (2 PB(2,0)) / (2 PB(2,0) + 1) = 2 / 3
PB(2,2) = (2 x 2/3) / ((2x2/3) + 2) = 2/5
PB(2,3) = (2 x 2/5) / ((2 x 2/5) + 3) = 4/19.
89
Sheet -5
1- Assume that a trunk group has enough channels to immediately carry all of the traffic
offered to it with an arrival rate of one call per minute. Assume that the average holding
time is 2 min. What percentage of the total traffic is carried by the first five circuits,
and how much traffic is carried by all remaining circuits?
Solution 1: The traffic intensity (load)of the system is;
A = 1 x 2 = 2 Erlangs.
The traffic intensity carried by i active circuits is exactly i Erlangs. Hence the traffic
carried by the first five circuits can be determined as follows;
A5 = 1 P1 (2) + 2 P2 (2) + 3 P3 (2) + 4 P4 (2) + 5 P5 (2)
 2 3 4 5

= e -2  2  2 x 2  3 x 2  4 x 2  5 x 2  = l.89 Erl
 
 2! 3! 4! 5! 
All of the remaining circuits carry: 2 - 1.89 = 0.ll Erlang
2- A Tl line is to be used as a tie-line trunk group between two PBXs. How much traffic
can the trunk group carry if the blocking probability is to be 0.1?. What is the offered
traffic intensity?
Solution: From the utilization curves it can be seen that the output circuit utilization for
PB = 0.1 and N = 24 is 0.8. Thus the carried traffic intensity is 0.8 x 24 = 19.2 erlangs.
Since the blocking probability is 0. l, the maximum level of offered traffic is;
A= 19.2  21.3 erlangs
1 0.1
3- Four clusters of data terminals are to be connected to a computer by way of leased

90
circuits, as shown in figure. In figure a, the traffic from the clusters uses separate
groups of shared circuits. In figure b the traffic from all clusters is concentrated onto
one common group of circuits. Determine the total number of circuits required in both
cases when the maximum desired blocking probability is 5 %.
Assume that 22 terminals are in each cluster and each terminal is active l0 % of the
time.
Solution: The offered traffic from each cluster is:

22 x 0.l =2.2 Erlangs. Since the average number of active circuits is much smaller than the
number of sources, an infinite source analysis can be used. Using tables, the number of
circuits required for PB = 5 % at a loading of 2.2 Erlangs is N = 5. Thus, the configuration
of figure a requires a total of 20 circuits.
The total offered traffic to the concentrator shown in figure b is 4 x 2.2 = 8.8 Erlangs.
From tables, N = l3 circuits are required to support the given traffic load.
4- What happens to the blocking probabilities in figure a and b discussed in problem 3
when the traffic intensity increases by 50 %?
Solution: If the traffic intensity of each group increases from 2.2 to 3.3 Erlangs, the
blocking probability PB of the configuration of figure a increases from 5 % to almost l4 %.
In the configuration of figure b, an 50 % increase in the traffic intensity causes a 400 %
increase in the PB (from 5 to 20 %).
5-The states of a local Exchange with 5 channels are as follows; P(1) = 0.2, P(2) = 0.27,
P(3) = 0.18, P(4) = 0.1, and P(5) = 0.05. The lost traffic can overflowed to an overflow
Exchange that is able to carry double this lost traffic using 2 channels, calculate:
a- The probability that all local Exchange' channels are free.
b- The overflow Exchange carried traffic.
c- The probability that both Exchanges are blocked simultaneously, and their blocking
rate.
6- Show the effect of increasing the number of channels from N = 4 to N = 6 on a link of
an Exchange on its performance when the offered traffic A is 2.0 Erlang.
7- Calculate the capacity, and channel efficiency degradation factors resulted when
replacing one Exchange with 36 channel by two exchanges to achieve GoS of 0.02.
91
8- What is the smallest number of channels which allow an offered load of 0.5 Erlangs to
suffer no worse than 2 percent loss.
Solution: PB(0.5,0) = 1
PB(0.5,1) = (2 PB(0.5,0)) / (2 PB(0.5,0) + 1) = 1 / 3
PB(0.5,2) = (2 x 1/3) / ((2x1/3) + 2) = 1/13
PB(0.5,3) = (2 x 1/13) / ((2 x 1/13) + 3) = 1/79,
So, the answer is 3.
Sheet -6
1- Compare between the traffic load on a set-up channel used to receive call set-up
requests from 90 traffic channels in an exchange and their traffic load, when the set-up
time is 100 m sec, with 25 m sec between calls.
2- a group of 7 channels are used to carry traffic of 4 Erlang, find:
a- the grade of service,
b- the probability that only one channel is busy,
c- the probability that only one trunk is free,
d- the probability that at least one trunk is free.
3- Consider 10 channels group serving a first attempt offered traffic load of 7 Erlangs.
What is the blocking probability. Find the blocking probability of the random retries for
all blocked calls.
4- Consider a link between two packet routers. Assume that,
– on the average, 10 new packets arrive in a second,
– the mean packet length is 400 bytes, and
– the link speed is 64 kbps.
Show the effect of increasing the link speed to 150 Mbps
92
Sheet -7
1- What is the blocking probability of a PBX to a central office trunk group with 10
circuits servicing a first- attempt offered traffic load of 7 Erlangs? What is the blocking
probability if the number of circuits is increased to 13? Assume random retries for all
blocked calls.
Solution: It can be assumed that the 7 Erlangs of traffic arise from a large number of PBX
stations. Thus, an infinite source analysis is justified. The blocking probability for
A = 7 Erlangs and N = l0 servers is about 8 %. Thus, the total offered load, including
retries, is approximately 7.6 Erlangs. With N = l0 and A'= 7.6, the blocking probability is
10 %. Two more iterations effectively produce convergence at A' = 8 Erlangs and
PB = l2 %. If the number of circuits in the trunk group is increased to 13, the blocking
probability of a lost calls cleared system is 2 %. Thus, a first approximation to the
returning traffic intensity is 7/0.98 = 7.14 Erlangs. Hence the blocking probability
including all returning traffic increases only slightly above the 2 %. This demonstrates that
the effect of returning traffic is insignificant when operating at low blocking probabilities.
At high blocking probabilities, however, it is necessary to incorporate the effects of the
returning traffic into the analysis.
Relationship between blocking probability of lost calls cleared and lost calls returning.
2- A group of callers generate requests at a rate of five calls per hour per telephone
(including incoming and outgoing calls). Assuming the average holding time is 4 min,
what is the average calling rate of each idle source? How many callers can be supported
by a l2-channel concentrator / multiplexer if the maximum acceptable blocking
probability is l %?
Solution: Since each caller is typically active for 20 min of every hour and places an
average of five calls during the 40 min of inactivity, the calling rate for idle sources;
λT = 5 / 40 = 0.l25 calls per minute. The offered load for M sources, assuming all traffic is
carried, is 0.33 M. From the finite number of users LCC table, we find the largest M such
93
that 0.33 M is less than or equal to the maximum offered load for PB = l % and N = 12.
Using interpolation for M = 2l reveals that 12 servers can support 7.11 Erlangs at
PB = 1 %.
Since 21 x 0.33 = 6.93 is the offered load, 2l sources is an acceptable solution. If 22
sources are used, the offered load of 7.26 Erlangs is higher than the 7.04 Erlangs
obtainable from interpolation in the table as the maximum offered load for
PB = l %.
It is worthwhile comparing this result to a result obtained from an infinite source analysis
(Erlang-B). For a blocking probability of l %. Erlang table reveals that the maximum
offered load for 12 servers is 5.88 Erlangs. Thus, the maximum number of sources can be
determined as 5.88 / 0.333 = 17.64. Hence in this case an infinite source analysis produces
a result that is conservative by l5 %.
3- Two trunk groups are to be used as direct routes between two switching systems. The
first group has 12 channels and the second group has 6 channels. Assume 10.8 Erlangs
of traffic is offered to the l2-channel group and overflows are offered to the 6-channel
group when the first group is busy. What is the blocking probability of the first group,
and how much traffic overflows to the second group? Using the overflow traffic
volume as an offered load, determine the blocking probability of the second trunk
group. What is the probability that both trunk groups are busy? Compare this answer to
the blocking probability of one 18-channel trunk group.
Solution: Using a lost calls cleared analysis with an infinite source, we determine that
blocking of the first group is l5 % (A= l0.8, N = 12). Therefore, the overflow traffic is
10.8 x 0.l5 = 1.62 Erlangs. The blocking probability of the second group is 0.5 %
(A = 1.62, N = 6). The probability that both trunk groups are busy simultaneously can be
determined (assuming independence) as:
PBT = 0.15 x 0.005 = 0.00075
In contrast, the correct blocking probability of an l8-channel trunk group with an offered
load of 10.8 Erlangs is;
PB = 0.013
For A = 10.8, N1 = 12, and N = 18 reveals that the appropriate conditional probability B2/l
is 0.033. Thus, the composite blocking probability PBC = 0.15 x 0.033 = 0.005. The
remaining inaccuracy (0.005 versus 0.013 actually) is due to nonrandom characteristics of
overflow traffic.
94
3.4-3 Delay System

In a delay system, calls finding free channels at the instant of their arrival, will be
serviced immediately, while calls arriving when all the channels busy are allowed to join
a queue waiting for service. They will be served when channels become free in the order
of their arrival, on first input first output, FIFO basis. The waiting time spent in the queue
may be limited or unlimited depending on the system discipline. Figure 3.22, illustrates
the delay process, with both service, and delay time specification.
Figure 3.22 Delay process
Queuing theory specialists have developed a notation for indicating the assumptions on
Input process, Service process, and Number of output channels. This notations list in
order, represented by letters and slashes, such as:
M / M / N, where the First letter M indicates that the input process is Markovian,
Memory-less random process that follow Poisson distribution, where the number of users
is assumed to be infinite. Second letter M stands for Markovian service process as well.
The third letter N represents that there are N server (channel) in the system.
M / M / 1 model gives the same representation except that the number of servers is only
one M / G / 1 representation has the exception that the service process is general. Figure
3.22' illustrates the general notation of the queueing system.
95
3.4-3.1 Multi-channels delay system

Considering M / M / N delay system, with infinite number of users, where calls arriving
the system randomly with Poisson distribution with mean arrival rate λ. Figure 3.23
illustrates the block diagram of the BCD system. Using the state transition diagram of such
system indicated in figure 3.24, every arrival affects a change of the system state, where j
represents the number of users in the system either in service (number of busy channels) or
in the queue;
Figure 3.22' Queueing system notation
Figure 3.23 Blocked calls delayed system.
Figure 3.23 Blocked calls delayed system

Figure 3.24 State transition diagram for BCD system. 96
λj = λ j = 1,2,3,.… ∞ (3.19)
and,
It is clear that call trials start to join the queue after j = N, so the probability that the queue
is empty is given by;
N
P ( Md = 0 ) = P (j ≤ N) =  P ( j ) (3.20)
j 0
where Md represents the number of call trials in the queue.
Using the state transition diagram given in figure 3.24, with the assumption that state j is
in statistical equilibrium;
λ P(j-1) + (j+1) μ P(j+1) = (λ+jμ) P(j) N≥j>0 (a)
λ P(j-1) + N μ P(j+1) = (λ+Nμ) P(j) j>N (b)
putting A = λ / μ, and changing j from 0 to N, then equation (a) will give the general
equation;
N
P(N) = A P(0) (c)
N!
with j = N+1, N+2, ……., equation (b), will give;
jN
P (j) =  A  (d)
  P( N )
N
The Probability of Delay will be calculated as;
  jN
PD = P (j ≥ N) =  A
 P( j )     P( N ) (e)
jN jN  N 
 PD = P(N)   A  A
2
 A
3 
1        .........

  N   N  N 

97
 PD = P(N)  N 

 N  A

P(0) can be calculated using the equation;

 P ( j ) 1
j 0
then,
1
P(0) =  N 1 A k A N  N  (3.21-a)
    

 k 0 k ! N !  N  A 

The probability of delay will be;

 N 
PD = A N / N!  (3.21-b)
 N  A

N 1
   N 
 A / k!  A / N ! 
k N
 N  A

k 0
which is known as Erlang delay or Erlang-C formula

The average number of delayed calls, which is the Delayed Traffic, defined as the average
number of incoming call trials during the average delay time, will be calculated as;
 
Md = AW =  ( j  N ) P( j )   j P( N  j )
j  N 1 j 1
 Md = AW = PD A / ( N – A ) (3.22)
The mean Delay (Waiting) Time of all calls is calculated by taking, AW = λ W , then;
1  Aw
W=  ( j  N ) P( j ) =
 j  N 1 
W = PD (Tc / N-A) (3.23)
while the mean Delay Time of the delayed calls is;
98
TD = Tc / N-A (3.24)
Equation 3.21 describes the probability of delay occurrence for call trials, but the
unconditional probability of delaying all call trials longer than specific period of time t is
calculated as;
P (W > t) = PD exp [- (N-A)μt ] (3.25)
The conditional probability of delay, given that the delayed call trial will stay in the queue
longer that specific time t is;
Pc (W > t) = exp [ - (N-A) μt ] (3.26)
The System Traffic calculated as the average number of calls in the system either being
served or waiting in the queue is calculated as;
 N 1 
AST=  jP( j )   jP( j )  NPD   ( j  N ) P( j ) (3.27)
j 0 j 0 j  N 1
which may be rewritten as ;

AST = AS + AW = λ (TC +W) = λ S (3.28-a)
AST = MS + Md = MST
where, S is the average system time.
The carried traffic by a group of N channels defined as the average number of busy
channels, given by;
AC = AS =  N1jP ( j )  NP = A (3.28-b)
D
j 0
The delayed traffic, defined as the average number of delayed calls, given by;
AW =  = A PD (3.28-c)
 ( j  N ) P( j ) N A
j  N 1
The figure of merit of such system is given by;

pD
FMDN = W = (3.29)
Tc NA
99
3.4-3.2 Single-channels delay System

For M / M / 1 delay system, a simple queue is formed in front of a single channel,
as shown in figure 3.25. It is assumed that call trials arrive at random and served in the
order of their arrival.
Figure 3.25 Single channel queuing system

The channel holding time is assumed to be negative exponential distributed. The state
transition diagram is illustrated in figure 3.26.
It is said that the system is in state j when there are j calls in the system including the one
being served. Assuming the
system in statistical equilibrium then,
λj = λ , j = 1, 2, 3, … (3.30)
μj = μ , j = 1, 2, 3, ……
Figure 3.26 State transition diagram for M / M / 1 delay system

Assuming infinite queue space is provided, and that customers are served in a FIFO basis,
then for state j,
λ P(j-1) + μ P(j+1) = (λ + μ )P(j)
using, j = 0, 1, 2, …….then;
P (1) = A P(0), P(2) = A2 P(0)

………. .
P (j) = Aj P(0)
100

Using the equality  P ( j ) = 1, we can get;
j 0
P(0) = 1 – A (3.31)
which is the probability that an incoming call trial will find a queue of zero length and will
accordingly served, so
P(j) = Aj (1 - A) j = 0, 1, 2, …… (3.32)
which is the probability of finding j customers in the system.
The probability of finding the buffer empty is given by;
N
P (Md = 0) =  P ( j ) = 1 – A2 (3.33)
j 0
The average number of calls in the system will be;
 
AST =  j P( j )   j A j (1  A)
j 0 j 0
 AST = A (3.34)
1 A
The average system time will be;
S=
1 (3.35)
 
The carried traffic will be given using equation 3.28-b;
AC = MS = A (3.36)
The probability of finding at least j customer in the system may be calculated as;

P ( MST ≥ j ) =  P (i ) = Aj (3.37)
i j
with j = N = 1, in M / M / 1 queuing model,
P ( MST ≥ N ) = A (3.38)
It is left to the student to prove that this probability will be corresponding to:
101

PD =  P ( j ) = P (W > 0)
j 1
PD = P (MST ≥ N) = P (MS = N)
but, the probability of finding j call trials in the queue is given by ;
P (Md = j) = P(j+N) (3.39)
Using equation 3.28-c, the delayed traffic will be;
PD A A2
AW =  (3.40)
1 A 1 A
The average delay time for all calls will be calculated using equation 3.23, such as;
A
W=
w  PD TC  A TC (3.41)
 1 A 1 A
The average delay time for delayed calls will be calculated using equation 3.24, such as;
TD = Tc / 1 – A (3.42)
The figure of merit of such system will be calculated using equation 3.29, such as;
F.MD1 =
W = A (3.43)
T 1 A
c
Equation 3.38 describes the probability of delay occurrence for call trials, but the
unconditional probability of delaying all call trials longer than specific period of time t is
calculated as;
P (W > t) = PD exp [ - (1-A) μt ] (3.44)
The conditional probability of delay, given that the delayed call trial will stay in the queue
longer that specific time t is;
Pc (W > t) = exp [ - (1-A)μt ] (3.45)
102
Sheet -8
1- A message switching network is to be designed for 90 % utilization of its transmission

link. Assuming exponentially distributed message length with average arrival rate of 10
messages per minute. What is the average waiting time and what is the probability that
the waiting time exceeds 3 minutes?
2- Assume that packets arrive at rate λ = 50 packet/s and the link speed is C = 64 kbps.
Calculate the probability that an arriving packet of 1000 bits has to wait ≥ 0.1 sec.
3- For a single server queuing system with average arrival rate of 1 message/sec. and
service rate of 3 messages/sec. Calculate the probability of receiving no messages in
a 5 sec. period. Also find the probabilities of queue lengths of 0, 1, 2, 3. if the queue
length is limited to 4 what percentage of messages will be lost?.
4- Consider a switch at which packets arrived with 3 packets per sec. mean arrival rate,
and 100 m sec. mean service time. Assume the packet comprises 70 bytes, and the
output transmission rate is 5.6 k bit/sec. How long does a packet have to wait in the
queue?
5- An X.25 packet switch has a single outgoing transmission link at 2 M bits/sec. the
average length of each packet is 960 bytes. If the average packet system time through
the switch is to be less than 15 msec., determine;
i- The maximum gross input packet rate to the switch;
ii- The average length of the queue;
iii- The utilization factor through the switch if each packet in the input is converted into
ATM cells having 48 bytes of data as a payload, plus 5 bytes as overhead.
103
Sheet -9
1- A message-switching network is to be designed for 95 % utilization of its transmission

links. For exponentially distributed message lengths and an arrival rate of 10 messages
per minute, what is the average waiting time, and what is the probability that the waiting
time exceeds 5 min.?
Solution: Assume that the message-switching network uses a single channel between each
pair of nodes. Thus, there is a single server and a single queue for each transmission link.
Since ρ is given to be 0.95 and λ = 10 arrivals per minute, the average service time can be
determined as tm = 0.95 / 10 = 0.095 min. The average waiting time (not including the
service time) is easily determined as;
t = 0.95 x 0.095 1.805 min
1 0.95
The probability of the waiting time exceeding 5 min as;
- ( 1- 0.95) 5 / 0.095
P (>5) = (0.95) e = 0.068
Thus 6.87 % of the messages experience queuing delays of more than 5 min.
2- Determine the number of digit receivers required to support 1000 telephones with an
average calling rate of two calls per hour. Assume the dialing time is exponentially
distributed with an average service time of 6 sec. The grade of service objective is to
return dial tone within 1 sec of the off-hook signal for 99 % of the call attempts.
Compare the answer obtained from a delay system analysis to an answer obtained from
a loss system analysis at PB = l %. If the PB is less than l %, fewer than l %of the calls
are delayed.
Solution: The calling rate λ and the offered traffic intensity A are easily determined as
0.555 calls per second and 3.33 Erlangs, respectively. since the number of servers N
cannot be solved for directly from the equations, Using the following figure b to obtain
a value of eight servers for:
t / tm = 1 .
6
Erlang table reveals that 99.5 % of the call attempts can be serviced immediately if there
are nine digit receivers. Thus, in this case the ability to delay service provides a savings of
only one server. It is clear that a blocking probability analysis produces approximately the
same results as a delay system analysis when the maximum acceptable delay is a small
percentage of the average service time.
104
The two results are almost identical because, if a digit receiver is not immediately
available, there is only a small probability that one will become available within a short
time period. (With an average service time of 6 sec, the expected time for one of eight
digit receivers to be released is 6/8 = 0.75 sec. Hence the delay operation in this case
allows a savings of one digit receiver.)
Because a digit receiver must be available within a relatively short time period after
a request is generated, digit receiver group sizing is often determined strictly from
a blocking probability analysis. The fact that digit receiver access is actually operated as
a delay system implies the grade of service is always better than that calculated.
Delay system capacity, a- P(>t) = 10 %, b- P(>t) = 1 %.

3- A packet-switching node operates with fixed-length packets of 300 bits on 9600-bps
lines. If the link utilization is to be 90 %, what is the average delay through a node?
What percentage of packets encounter more than 0.35 sec of delay? What is the average
delay if the offered load increases by 10 %?
Solution: Message lengths of 300 bits and a data rate of 9600 bps imply that the fixed
length service time is 300 / 9600 = 0.031 sec. The average waiting time is;
t
t = 2 (1 m )
105
0.9 x 0.031
t = 2 (1 0.9)  0.14 sec .
The total average delay through the node, excluding processing, is obtained by adding the
average waiting time to the service time:
Average delay = 0.140 + 0.031 = 0.171 sec.
Since the service time is 0.031 sec, 0.35 sec of delay occurs when the waiting time is
0.35 - 0.031 = 0.319. This corresponds to 0.319/0.031 = 10 service times. From the
following figure, the probability of delay for t / tm = 10 is approximately 0.12. Thus l2 %
of the packets experience delays of greater than 0.35 sec. An increase of 10 % in the traffic
intensity implies that the new offered load is 0.99 erlang. The average waiting time
becomes;
0.99 x 0.031
t = 2 (1 0.99) 1.53 sec .
Thus, when the offered load increases by only l0 %, the average delay through the node
increases nine fold to a value of
1.53 + 0.031 = 1.56 sec !
This demonstrates the same characteristic for heavily utilized delay systems that was
demonstrated for loss systems: The performance is very sensitive to increases in traffic
intensity. Thus, flow control is a critical aspect of a packet-switching operation,
particularly when there are real-time delivery objectives.
The average delay of only those arrivals that get delayed is commonly denoted as;
tm
= 0.031
N  A 1 0.99  3.1sec .
tW =
106
Delay probabilities of single-server systems
3.5 Routing Plan

Traffic routing patterns describe the route set choices and route selection rules for each
origin-destination pair. They may be hierarchical or non-hierarchical. Routing refers to the
procedures that determine which paths in a network are assigned to a particular
connection. Routing of traffic must be carried out in an economical manner within the
network constraints. Mainly it is categorized into two types; Direct, and indirect routing.
The term, 'direct route' implies that it consists of one or more circuits interconnect two
adjacent network nodes (switching centers) such that calls are routed from one node to
another without passing through an intermediate node. If circuits from node A to node B
pass through node C, this is still a direct route if the circuits are not switched at C.
National networks are hierarchical, there is a backbone route called indirect route consists
of a series of circuits joining each switching center at the lowest level to the highest level
center via intermediate (tandem, transit centers) nodes, together with traverse branches
107
that are used whenever there are free channels. There may also be some other direct
(shortest) routes that are usually chosen to interconnect nodes at different levels. If
possible, connections should use the most direct routes at the lowest levels of the network.
This is because they use fewer network facilities and generally provide better transmission
quality. However, economic considerations often limit the capacities of the direct routes so
that alternate routes are needed.
Indirect route is a final route as it is the last choice route in the sense that no traffic streams
using this route may overflow further. The routing plan must have regard to the following,
to choose between direct and indirect routing:
a- The transmission, and signaling plans, which influences the maximum number of
tandem links in a route.
b- The setting up time, which is influenced by the capabilities of the signaling types and
switching equipment in use.
c- The economics of routing direct or via a tandem (indirect) switching point. These are
affected by the relative costs of transmission and switching plant and by the amount of
traffic to be carried by the route.
d- The grade of service required end to end.
e- The number of required routing digits.
These limit the maximum number of tandem links that may be used to set-up connections.
Routing a stream of traffic via a tandem switching center is always more economic than
using the direct route. In fact even greater economic is often possible if just a proportion of
the traffic is routed directly, which known as “alternative routing”. In this approach a few
direct trunks between two centers are provided to form a high usage route. Majority of
calls use these high usage routes and only a small proportion of calls (residual traffic) are
rerouted via a tandem center using the network back bone, which is most effective in busy
hours, when all the direct trunks are busy.
Also increasing the capacity of an existing trunk route always requires fewer additional
trunks than the provision of a new direct trunk route. This is due to the nonlinear
108
relationship between number of trunks and carried traffic calculated using Erlang formula.
With 0.01 grade of service, a single trunk carries 0.01 Erlang, two trunks carries 0.15
Erlang, and three trunks carry 0.45 Erlang, while adding the 11th trunk increases the
carried traffic by 0.7 Erlang than that carried by 10 trunks. Suppose we have three areas
served by 3 switches A, B and C.
8 Erlang on the routes AB and AC requires 15 trunks on each route. The 2 Erlang on the
route BC requires 6 trunks using the direct route. However, on the main route an
additional 2 Erlangs could be carried by the addition of only 3 trunks as shown in figure
3.27.
Figure 3.27 Comparison between Tandem and Direct Routing

(a) traffic flows for example, (b) Direct Route, (c) Tandem Route
Consider a higher traffic load on route BC of 10 Erlangs, so the choice is between 18
direct trunks or 12 extra trunks via the tandem center, which is cheaper.
The above discussion was based on the choice of all direct or all tandem routing. It is clear
that routing a stream of traffic via tandem exchange is more economic rather than routing
it directly between local centers.
3.5-1 Routing Structure
A routing structure is hierarchical such as originated calls are offered to the direct
route toward the terminating exchange. If this route is busy, these calls will be overflow to
an alternative route in the same level via an overflow exchange irrespective of the routes
already tested. If this route too was busy, calls are overflow across to the next higher
exchange. If all the routes across were busy, then the calls could ascend to the next higher
originating exchange and attempt to cross over again. The routes in the set will always be
109
tested in the same sequence, although some routes may not available for certain types of
calls.
An example of hierarchical routing in a non-hierarchy network of nodes is shown in figure
3.28.
3.5-2 Routing Schemes
Routing scheme defines how a set of routes made available for calls between a pair of
nodes. Routing scheme may be fixed or dynamic.
Figure 3.28 Hierarchical routing in a non-hierarchy network.
3.5-2.1 Fixed Routing Scheme

Routing patterns in the network may be fixed where the set of routes in the routing
pattern is always the same. Changes to the route choices for a given type of call attempts
require manual intervention and represent a "permanent change" to the routing scheme.
The introduction of new routes requires a change to a fixed routing scheme.
3.5-2.2 Dynamic Routing Scheme

It incorporates frequent automatic variations. Such changes may be time-dependent,
state-dependent, and/or event-dependent. The updating of routing patterns may take place
periodically or a periodically, predetermined, depending on the state of the network or
depending on whether calls succeed or fail in the setup of a route. This capitalizes on two
factors:
1- Selection of minimum-cost paths between originating and terminating switches;
110
2-Design of optimal, time-varying routing patterns to achieve minimum cost trunking by

taking advantage of non coincident network busy periods.
Time-Dependent Routing. Here the routing patterns are altered at fixed times on a pre-
planned basis during the day or week to allow for changing traffic demands and achieve
maximum trunk utilization, and minimum network cost. For each originating and ter-
minating exchange pair, these changes for a particular route pattern is preplanned and
implemented consistently over a long time period depending on the time of day and day of
the week as shown in figure 3.29. A weekday, for example, can be divided into different
time periods, with each time period resulting in different route patterns being defined to
route traffic streams between the same pair of nodes.
Figure 3.29 Time-Dependent Routing

This type of routing takes advantage of idle circuit capacity in other possible routes
between the originating and terminating exchanges that may exist due to non coincident
busy hours. Crank back may be utilized to identify downstream blocking on the second
link of each two-link alternative path.
.Slate-Dependent Routing. In this case, routing patterns are varied automatically
according to the state of the network, which called adaptive routing schemes.
A centralized routing processor is employed to select optimum routing patterns on the
basis of actual occupancy level of circuit groups and nodes in the network which are
111
monitored on a periodic basis (e.g., 10 s) as illustrated in figure 3.30. Qualitative traffic

parameters may also be taken into account in determining the optimal routing pattern.
Figure 3.30 State-dependent Routing
This routing technique inherently incorporates fundamental principles of network

management in determining routing patterns. These principles include:
• Avoiding occupied circuit groups.
• Not using overloaded nodes for transit.

• In overload circumstances, restriction of routing direct connections.
Event-Dependent Routing. Here routing patterns are updated locally on the basis of
whether calls succeed or fail on a given route choice. Each exchange has a list of choices,
and the updating favors those choices which succeed and discourages those which suffer
congestion.
In a fully connected (mesh) network, calls between each originating and terminating
exchange pair try the direct route with a two-link alternative path selected dynamically.
While calls are successfully routed on a two-link path, that alternative is retained, a new
two-link alternative path is selected. This updating, for example, could be random or
weighted by the success of previous calls.
This type of routing scheme routes traffic away from congested links by retaining routing
choices where calls are successful. It is simple, adapts quickly to changing traffic patterns,
and requires only local information. Such a scheme is illustrated in figure 3.31.
3.5-3 Route Selection
112
Route selection is the action to actually select a definite route for a specific call. The
selection may be sequential or non sequential.
Figure 3.31 Event-Dependent Routing in a Mesh Network.

In sequential selection, routes are always tested in sequence and the first available route is
selected. For the non sequential case routes are tested in no specific order.
The decision to select a route can be based on the state of the outgoing circuit group or the
states of series of circuit groups in the route. In either case, it can also be based on the
incoming path of entry, class of service, or type of call to be routed. One example of the
above is selective trunk reservation.
3.5-4 Routing Applications

The dynamic, or time-varying, nature of the routing scheme is achieved by introducing
several route choices. Alternative routing may be applied in any of the following ways;
3.5-4.1 Automatic Alternative Routing; AAR
It is a type of progressive (irreversible) routing. When a node has the option of using
more than one route to the next node, an alternative routing scheme can be employed. Two
main types of AAR are available:
1- When there is a choice of direct-circuit groups between two nodes.

2- When there is a choice of direct and indirect routes between the two nodes.
113
Alternative routing takes place when all appropriate circuits in a group are busy. Several
circuit groups may be tested sequentially. The test order is fixed or time-dependent.
3.5-4.2 Automatic Rerouting; ARR (Crank back)
It is a routing facility enabling connection of call attempts that congested during the
initial call setup phase. Its performance can be improved through the use of different
signals S1 and S2 for example to indicate congestion as illustrated in figure 3.32 such as;
1- Signal S1 received at the originating node A indicating that congestion has occurred on
outgoing trunk from node B subsequent to the seizure of an outgoing trunk from node
A, the call can be rerouted at A.
2- Signal S2 received at the originating node A indicating that congestion has occurred
further down stream on outgoing trunks from node D.
The action to be taken at node A upon receiving S1 or S2 may be either to block the call or
to reroute it.
A call from A to D is rerouted via C because the circuit group B-D is congested
(S1-indicator) and a call from A to F is rerouted via E because circuit group D-F is
congested (S2-indicator).
One positive consequence of this alternative is to increase the signaling load and number
of call setup operations resulting from the use of these signals.
Note- Blocking from B to D activates signal S1 to A. Blocking from D to F activates signal

S2
Figure 3.32 Automatic Rerouting (ARR) or Crank Back.
If such an increase is unacceptable, it may be advisable to restrict the number of re-

routings or limit the signaling capability to fewer nodes.
114
To route a call between Cairo and Tanta through Benha for example, the originating node
retains control over a dynamically routed call until it is either completed to its destination
at Tanta or blocked. A call overflowing the second leg of a two link connection, for
example, the Menoufia - Tanta link of the Cairo - Menoufia - Tanta path is returned to the
originating node Cairo for possible further alternate routing. Control is returned by
sending a common channel signaling crank back from the Benha node to the originating
node.
Care must be taken to avoid uncontrolled routing that can results in exceeding the
permitted number of links being connected in tandem, or a connection failing to be
completed because the routing turns back upon itself, (circular routing) that returns the call
to the point at which blocking previously occurred during call setup.
3.5-4.3 Load Sharing

All routing schemes should result in the sharing of traffic load between network
elements. Routing schemes can, however, be, developed to ensure that call attempts are
offered to route choices according to a preplanned distribution. Figure 3.33 illustrates this
application of load sharing, which can be made available as a software function of stored
program controlled, SPC, exchanges. The system works by distributing the call attempts to
a particular destination in a fixed, ratio between the specified routing patterns.
Figure 3.33 Preplanned distribution of Load Sharing

Note that on one hand routing design is based on traffic engineering considerations:
115
for example, alternative routing schemes are based on cost efficiency considerations,
dynamic routing methods are based on considerations of robustness under focused
overload or failure conditions or regarding traffic forecast errors. On the other hand,
network dimensioning is done by taking into account routing methods and routing
patterns.
Large groups of circuits are more efficient than small groups because of their higher
occupancy (i.e. traffic per circuit). If there is a large amount of traffic between two
exchanges, it is economical to provide a direct route between them. If there is less traffic,
it becomes economical to use automatic alternative routing; the direct circuits are used
efficiently and the overflow traffic shares tandem circuits with traffic to other destinations.
If there is very little traffic between two exchanges, it is uneconomical to provide any
direct circuits and all calls should be routed via a tandem exchange. The traffic levels for
which these three different routing methods should be used depend on the relative costs of
direct and tandem circuits (including associated switching and signaling equipment). The
numbers of high-usage and tandem circuits required should be carefully determined for
automatic alternative routing to be used. Alternative routing is usually used and is most
effective when busy hours on different routes do not coincide, as in large countries
because of the time difference between east and west.
Alternate routing of traffic is primarily useful for accommodating local overloads by
transferring traffic to underutilized routes. During network with overloads, however,
alternate routing is undesirable for two reasons. First, alternate routes imply that a greater
number of transmission and switching facilities are needed for a connection. If these same
facilities could be assigned to two or more direct connections, the total number of links per
call could be reduced and the network could carry more traffic.
Second, the probability that an alternately routed call can acquire all the necessary
resources is relatively low. Trying to set up a connection with a large number of facilities
is undesirable if the probability of getting all of the facilities is low.
116
Assuming that a country has been divided into local exchange areas and the locations of
their exchanges have been decided, a routing plan should be developed to include as a
minimum:
1- Description of the hierarchy; i.e.,
i- Which local exchanges should be interconnected by direct junctions.
ii-Which connections made indirectly via tandem switching centers.
iii- The number and location of tandem switching centers
iv- The number of levels of tandem switching to be used in the network.
v- Whether automatic alternative routing is to be used and, if so, under what conditions.
2- Definition of full direct, high usage, and overflow routes and the criteria for choosing
between them.
3- Specifications of grade of service on trunk and local routes.
4- Principal route layouts geographically.
5- Guidelines for the selection of transmission media for major and minor routes.
In a national plan, the routing of traffic can usually be considered separately for the trunk
and local networks. However, the routing plan for a local area must be compatible with the
national transmission and numbering plans, because long-distance connections are set up
through both the local network and the trunk network, possibly involving the maximum
number of links in tandem.
The interface between a local network and the trunk network is the local tandem exchange
3.6 Charging Plan

The allocation of charge areas must take account of technical and numbering aspects.
In addition, it is necessary to consider the administrative and political viewpoint to ensure
that adjacent subscribers are not in different charge groups. The charging areas should be
kept closely coincident with exchange and numbering areas to simplify code analysis and
the associated charging equipment.
The cost incurred in making a call is dependent on the amount of plant involved and,
hence on the distance involved and the duration of the call. Since these parameters are
117
readily determined, the charge made for a call is normally proportional to distance x time.
However, in some countries local calls are included in the subscriber's rental (the flat-rate
tariff). In determining the charge to be made, it is normally sufficient to have a knowledge
of the location and class of the calling subscriber, the location of the called subscriber
(determined by the actual number dialed) and the call duration. The processing of this
information and the presentation to the customer can take a variety of forms. Extremes
extend from merely billing the total number of 'units' of charge incurred, to a detailed
account listing each call and the resulting charge. The charging plan must satisfy the
following:
1- The charging plan should be easily understood by subscribers.
2- Charging for calls to and from subscribers who are geographically close should be
similar.
3-The charging areas should be kept closely coincident with numbering and exchange
areas in order to simplify code analysis and the associated charging equipment.
4- The expenses providing a telephone services comprises:
i- Capital costs: include that of line plant, switching equipment, building and land.
ii- Current operating expenses: include staff salaries, the cost of maintenance and
power.
All these costs must be met by the income obtained from subscribers.
5- The charges paid by subscribers are:
i- An initial charge for installing the subscriber's station apparatus and connecting it to
the network.
ii- An annual rental or leasing charge.
iii- Charges for individual calls made.
3.6-1 Local Call Charging
There are three methods for charging the local calls. These are the message rate
charging, the flat rate tariff, and the hybrid method.
3.6-1.1 Message Rate Charging Scheme
118
In this approach, a separate fee is charged for each call made by the subscriber.
Each subscriber's line is equipped with a meter (message register) and operate this once to
make a unit charge on each call when the called subscriber answers. All completed calls
are charged. Thus, the metering circuitry must also sense call supervision to respond to
call completion in order to stop meter pulses. Message call charging schemes make the
charge for each call proportional to its duration. A local call timing circuit is fitted in the
supervisory unit to operate the calling subscriber's meter at the appropriate intervals (e.g.,
once every three minutes). This method is called periodic pulse metering. It is common to
make a metering rate vary with the time of the day. This is done by changing the pulse
repetition frequency of pulse generator upon a control of the clock.
3.6-1.2 Flat-rate Tariff Scheme
Some administrations do not make a separate charge for local calls but include them
in the subscriber's rental. This is because the major of the cost of local networks is
independent on the traffic. It avoids the capital cost of providing subscriber's meters and
operating costs of tariff. This method may be unfair due to different calling rates.
To avoid this, two rental charges are made for business and residential subscribers.
3.6-1.3 Hybrid Scheme

In Europe, message call rate charging is adopted. In North America, flat-tariff is
chosen. Some administrations combine the features of flat-rate and message-rate charging.
The subscriber's rental payment covers a particular number of calls per annum
(representing the average number of calls per subscriber) and only calls above this number
are charged for. This method retains the disadvantages of message-rate charging and loses
the advantages of flat-rate charging since charging meters still need to be provided and
read.
3.6-2 Trunk Call Charging
For trunk calls, a separate charge for each trunk call is made. The cost of a trunk call
depends both on distance and duration, so charges are often, roughly based on the product
(distance x time). A typical tariff is based on a minimum charge for up to three minutes,
119
together with a charge (of less than a third of this amount) for each subsequent minute. For
charges to be based on distances between trunk exchanges, boundaries between charging
areas must coincide with boundaries between numbering areas.
3.6-3 International Charging
International call charging is carried out by the same methods as for trunk calls.
International calls require checking the digits of the country code that will then set the
meter to pulse at a higher rate. Some form of number translation is required to convert
dialed number information to key the metering circuit for the proper number of pulses per
unit period.
120
References
[1] Tarmo Anttalainen, "Introduction to telecommunications network engineering", Artech House
nd
INC., 2 ed., 2003.
nd
[2] Roger L. Freeman, "Fundamentals of Telecommunications", John Wiley & Sons, Inc., 2 ed.
2005.
th
[3] Annabel Z. Dodd, "The Essential Guide to Telecommunications", 6 ed., 2019.

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Menoufia University

Faculty of Electronic Engineering

Telecommunication Network Planning

Prof. Sami A. El-Dolil

Department offering the program: Electronics and Electrical Communications Engineering

Course Code: ECE 424 Course Title: Academic Year: 2022-2023

1. Describe, explain, analyze, and evaluate the performance of common

engineering fundamentals, basic science and mathematics.

B.3 Design and implement elements, modules, sub-systems, or systems in

identifying software tools and mathematical methods required to optimize their

10- List of text books and references:

11-Course contents – Course related program competencies

Level A Level B Level C

Level A Level B Level C

13- Assessment methods - Course related program competencies

Course related program comptencies

Authorized from department board at:

2.2 Forecasting Plan

2.3 Provision Timing Plan

3.2 Numbering Plan

3.3 Signaling Plan

3.4 Capacity Plan

3.5 Routing Plan

3.6 Charging Plan

Telecommunications (long distance communications) technology is paving the way to

The concept of telecommunications (long distance communications) is generally

Figure 1.1 Model of a communication system

1- A transmitter that takes information and converts it to a signal.

Figure 1.2 Electromagnetic Spectrum for Telecomm.

1.1 Classification of Communication Systems

a- Simplex Communication System

a- Simplex communication system

b- Half-duplex communication system

c- Multipoint -to-multipoint communication

1.1-4 Depending on the type of the transmitted signal:

Figure 1.5-a Analog Sinusoid Signal

Figure 1.5-b Digital Signal

There are number of basic network requirements such as:

Figure 1.6 A Hierarchy Communication Network

The purpose of a telecommunications network planning is to ensure that the

2.1 Financial Plan

2.1-1 Main Financial Plan

2.1-2 Medium Financial Plan

2.1-3 Micro Financial Plan

2.2 Forecasting Plan

2.2-1.1 Linear Model

M = Mo exp (t / T1) (2.2)

the percentage increase per annum is 100 /T1

Figure 2.1 Forecasting Models.

the demand size tends to be stable.

2.2-2 Forecasting of Traffic per line

2.3 Provision Timing Plan

Year 2012 2013 2014 2015 2016 2017

Number of lines x 103 100 104 108 111 111 114

The approximated line will be;

M = Mo + K1t = 98.6 + 2.69 t

Year 2030 gives t = 19

 M = 98.6 + 2.69 (19) = 149.71 = 150

 2 Mo= Mo exp (0.05 t)