Monash University Electrical and Computer Systems Engineering

ECE2041 Telecommunications
Information for Exam and details of work

The Examination will have six questions: Each worth 30 marks: Total for paper: 180 marks
(1 mark per minute)
Calculators will not be allowed in the examination. Leave numerical answers at the stage
where you would normally use a calculator for the final answer.
Any formulae required in the exam will be provided on a formulae sheet, but knowledge of
basic relationships such as frequency and wavelength, or speed, distance and time, decibel
power calculations, etc. are assumed.
Material such as the use of MATLAB that was covered in the laboratories but not in the
lectures is not examinable. (Material that was covered in lectures and to which the
laboratories were related may be on the exam)
The topics covered and the relevant parts of Leon-Garcia, or the printed notes are listed
below. (Note that some of the Sections from Leon-Garcia were mentioned under more than
one topic, but they are listed and summarised only once below.)
Note that the parts of the course covered by notes but not covered in Leon Garcia (Noise
in Digital Communication Systems and the lecture slides on OFDM) are examinable except
where noted below.

Topic 1 - Intro to telecoms

Chapter 1 - Communication Networks and Services; Section 1.1 Evolution of Network
Architectures and Services, Section 1.2 Future Network Architectures and Their Services

Components of a telecommunication network and relevant disciplines.

Connection-oriented and connectionless communications. Concept of a message. Packet
networks.
IP networks, Internet Protocol and DNS. Best effort networks. TCP and UDP.
Internetworking. Components of the Internet. How ISPs are interconnected.
Alternative network topologies and their features. Scale of applicability of different networks.
Standards development. Different access networks and their features.
Chapter 7 - Packet Switching Networks, Section 7.1 Network Services and Internal
Network Operation, Section 7.2 Packet Network Topology, Section 7.3 (skip 7.3.3)
Datagrams and Virtual Circuits

Difference between circuit and packet switching. TDM and FDM implementations of circuit
switched connections.
Calculations of transmission times and network utilisation.
Packet switching, statistical multiplexing and store and forward operation at routers.
Routing and delivery of datagrams. Virtual circuits.
Carriage of packet-switched communications over circuit switched networks.
Fibre-to-the-Home and HFC (Hybrid Fibre-Coax) access networks.
Monash University Electrical and Computer Systems Engineering

Topic 2 - OSI tcpip nettools

Chapter 2 - Applications and Layered Architectures, Section 2.1 Examples of Protocols,
Services and Layering (skip the SMTP protocol related paragraphs), Section 2.3 Overview
of TCP/IP Architecture, Section 2.5.2 File Transfer Protocol (FTP), Section 2.5.3 Hypertext
Transfer Protocol (HTTP) and World Wide Web, Section 2.5.4 IP Utilities

Purpose of layering and protocols. The layers in the TCP/IP protocol stack. The role and
services of DNS, HTTP, TCP and UDP protocols
5-layer TCP/IP protocol model. Functions, services and interaction of layers.
Connectionless and Connection-oriented services. Segmentation and re-assembly.
Multiplexing and demultiplexing.
The need for TCP/IP protocol stack. IP address and MAC address. How different headers and
layers are used to do routing. How encapsulation is achieved. The value of network analyzing
tools such as Wireshark.
Internet Standards (RFCs Request for Comments). The functions of Telnet, FTP and HTTP.
The role and use of PING, Traceroute, ipconfig, netstat.

Topic 3 - TCP/IP
Chapter 8 - TCP/IP, Section 8.1 The TCP/IP Architecture, Section 8.2 The Internet
Protocol (skip 8.2.8), Section 8.3 IPv6 (only 8.3.1, 8.3.2 and 8.3.4), Section 8.4 User
Datagram Protocol, Section 8.5 Transmission Control Protocol (only 8.5.1 and 8.5.2)

Reasons for internetworking. Encapsulation of protocol packets in TCP/IP suite. Difference

between IP and MAC addresses. Fields in IPv4 header, Checksum calculations.
Classes and reservations in IP addresses. ARP and ICMP operations. Use of ICMP with ping
and traceroute. Subnet related calculations. Packet routing in networks with subnets
Comparison between IPv6 with IPv4. Fields in IPv6. Interworking between IPv4 and IPv6.
TCP and UDP:
checksum calculations in UDP
multiplexing in UDP and TCP
fields in TCP header
checksum calculations in TCP
well-known port no, socket and connection
reliability, flow control and congestion control
steps involved in TCP connection management
Routing and forwarding in hosts and routers. Autonomous systems. Organization of routing
in the global internet, intra and inter-AS routing
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Chapter 7 - Packet Switching Networks, Section 7.4 Routing in Packet Switching

Networks, 7.4.1 Routing Algorithm Classification, 7.5.1 Bellman-Ford Algorithm
Comparison between IPv6 with IPv4. Fields in IPv6. Interworking between IPv4 and IPv6.

Topic 4 - Digital Transmission Fundamentals

Chapter 3 Digital Transmission Fundamentals

Section 3.1. Digital representation of information

Different requirements for different types of information block oriented and stream
oriented.
Use of binary to represent information. An n bit word can represent 2n different values.
Use of terms kilobit, megabit and gigabit in communications and computing.
Section 3.1.1 Block Oriented information
For many types of information, the critical aspect of a telecommunication network is how
long it takes to transfer an entire block of information, that is the time from starting to
transmit the first bit, until the final bit is received. This in turn depends on the data rate
supported by the network, the transmission delay of the network and how many bits are
required to represent the block.
Source Coding: The number of bits can often be reduced by source coding. Many
sources such as picture have considerable redundancy. Coding which reduces this
redundancy can compress the information. compression ratio = size of uncompressed
information/ size of compressed information. Examples of compression of text files,
faxs, still and moving images were discussed. Coding can be either lossy or lossless
(Leon-Garcia uses the terms noisy and noiseless). No information is lost in lossless
source coding, the original data can be recovered exactly. The compression results from
elimination of redundancy. In lossy compression, some quality is sacrificed in order to
reduce the number of bits, for example fewer colours may be used in a colour picture.
The original information cannot be recovered after lossy coding.
Section 3.1.2 Stream Information
Some types of information are generated as a continuous stream and must be received as
a continuous stream with very little delay. Typical examples are interactive voice (e.g.
telephony) and interactive video (e.g video conferencing). A delay of more than half a
second in each case makes satisfactory communication impossible. In this case the key
characteristics of the communication system are data rate and delay (also called latency).
Much stream information is derived from an analog source which must first be digitized
e.g. a speech signal. The bit rate depends on the number of samples per second and the
number of bits per sample.
The number of samples per second depends on the bandwidth of the signal. Sampling
must be at or above the Nyquist Rate which is two times the bandwidth of the signal. By
filtering the signal to remove high frequencies, the sampling rate can be reduced, so
telephone systems, limit the bandwidth to less than 4kHz and use 8000 samples per
second. CDs limit the bandwidth to 22kHz and use 44000 samples per second.
The number of bits per sample depends on the number of quantization levels.
The data rates generated by voice signals are more or less constant, whereas the data rates
generated by video systems may vary greatly depending on the complexity of the picture
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and the rate at which the picture changes.

The overall quality of the received stream information depends on latency (the delay
between transmission and reception, jitter ( how much the delay varies) and loss (how
many bits are received in error).
Section 3.2 Why Digital Communication
Simple model of communication system: Transmitter, channel, receiver. Channel
distorts and attenuates signal and noise and interference is added. Received signal is not
an exact replica of transmitted signal.
Section 3.2.1 Comparison of Analog and Digital Transmission
Analog systems: for long distance transmission, signal must be regenerated at intervals,
noise cannot be removed and distortion accumulates limits the distance and quality of
long distance transmission
Digital systems: Data can be covered exactly as long as 0s and 1s can be distinguished.
Always some errors but error correcting or error detecting coding can be used to recover
lost information.
Section 3.2.2 Basic Properties of Digital Transmission Systems
How fast can data be transmitted across a channel? Depends on received signal energy,
noise and bandwidth of channel.
Bandwidth of channel Channels characterized in frequency domain. Bandwidth of the
channel is range of frequencies which can be passed.
Maximum rate at which pulses can be transferred without interference through a channel
of bandwidth is 2W pulses per second where W is the bandwidth of the channel.
Multilevel pulses Two level pulses can carry one bit of information per pulse, four level
pulses can carry two bits. Data rate can be increased by increasing the number of levels.
2 levels can carry m bits. However for a given energy per pulse, as number of levels
m

increases the distance between levels reduces and the system becomes more susceptible
to noise.
Shannon channel capacity maximum data rate at which data can be transmitted reliably
through a channel is given by C =W log2 (1+ SNR) where W is the bandwidth of the
channel and SNR is the signal to noise ratio. Shannon capacity tells fundamental limit
that can be achieved (with sophisticated coding) but not how to achieve it.
Section 3.9 Error detection and Error correction
Bit error rates differ widely between systems but all digital communication systems will
introduce some bit errors.
Two approaches: error detection and retransmission (Automatic Repeat Request: ARQ)
or error correction (forward error correction: FEC).
ARQ: Only useful if delay not important, and there is a return path to advise
transmitter when an error has occurred.
FEC: Used in delay sensitive applications (e.g. mobile telephony) or where there is no
return channel or possibility of retransmission ( e.g. broadcast digital television).
Section 3.9.1 Error detection
Simplest error detection: parity check
Calculations on probabilities of detected and undetected errors. Use formulas for
combinations and simple probability concepts (see powerpoints for examples). Binary
symmetric channels (BSCs) were considered. In these 0 to 1 and 1 to 0 errors are equally
likely and errors occur randomly (not in bursts)
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Error detecting coding requires

Pattern that all code words have (e.g. for even parity, all code words have even parity)
o If a received codeword does not fit the pattern it is known to contain an error
Redundancy: not all possible words can be transmitted (e.g for even parity, words
with odd parity are not used, so only half of the possible words of a given length are
used)
All error codes have
Blindspots (coverage)
o Errors which cannot be detected. These occur when there are so many errors
that one valid codeword is changed into another valid codeword.
Modulo-2 arithmetic
Many error coding schemes use modulo-2 arithmetic, as this is easy to implement in
digital circuits and results in simple code structures.
Some codes also use modulo arithmetic with higher bases.
Section 3.9.2 Two dimensional parity checks
Two dimensional parity checks allow more errors to be detected. However these are not
good codes and are not used in practice so the details will not be covered in the exam.
Section 3.9.3 Internet Checksum
The Internet checksum is designed to be very efficient to implement as it is performed
one every packet at every router in the Internet. It uses modulo 216-1 arithmetic. A
simple example using modulo 24-1 arithmetic was given in the notes. Students are
expected to understand and be able to apply the basic concepts, but not to remember the
details of the particular code used in the Internet.
Section 3.9.4 Polynomial codes
Polynomial codes are also based on modulo-2 arithmetic, but a polynomial (rather than a
vector) is used to represent the data. They are the basis of the cyclic redundancy checks
(CRCs) used in many systems. Students are expected to be able to apply the basic
concepts but not to remember the details of any particular code.
Polynomial codes are important in practice as they can be implemented simply using shift
registers.
Section 3.9.5 and 3.9.6 were not covered in detail and are not on the exam
Section 3.9.7 Linear codes and section 3.9.8 (Bits relevant to Hamming codes)
A (7,4) Hamming code was considered in detail as an example of a simple error
correcting code. It was used to demonstrate the concepts of Hamming distance and
minimum distance. The check matrix for the Hamming code was derived and its use for
detecting whether a received code word was valid. (The use of syndromes for error
correction was not covered). In the practical problems examples were given of the
probabilities of corrected and uncorrected errors for particular cases.

Topic 5 - Transmission Limits

Section 3.3 Digital Representation of Analog Signals
To represent analog signals in digital form they must be sampled and quantized. Analog
signals are continuous and so an exact digital representation would require an infinite
number of bits. The rate at which an analog signal must be sampled depends on the
bandwidth of the analog signal
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Section 3.3.1 Bandwidth of Analog Signals

The concept of bandwidth of a signal depends on the frequency domain representation of
a signal. All signals can be represented by a sum of sinusoidal signals. For repetitive
signals the sinusoidal components can be found by using a Fourier series. For non
repetitive signals a Fourier Transform must be used.
Bandwidth of signal is range of frequencies at which signal contains nonnegligible
power.
Section 3.3.2 Sampling of an Analog Signal
An analog signal can be completely recovered from samples taken at or above the
Nyquist sampling rate. That is if it is sampled at a rate 1/T where 1/T>2W and W is the
bandwidth of the signal. (Sampling rate 1/T means that signals are taken at T spaced
intervals. (The details of the interpolation filter required at the receiver, were not covered
and will not be examined)
Section 3.3.3 Digital Transmission of Analog Signals
Digital transmission of analog signals requires (see Figure 3.20):
At the transmitter:
(bandlimiting essential but not explicitly stated in the textbook)
Sampling at above the Nyquist sampling rate
Quantization representing continuous signal by a finite number of levels this
introduces signal distortion called quantization noise
Representation of quantized signal in digital form
At the receiver
Conversion from digital values to signal pulses
Generation of continuous signal from signal samples using an interpolation filter
Data rate of digital signal is 2Wm (assuming sampling at Nyquist sampling rate) where
there are 2m quantization levels represented by m bits
The level of quantization noise depends on the number of quantization levels and the
statistics of the input signal. Calculation of the quantization noise was not covered in
detail and will not be examined. The key result is that increasing the number of levels
reduces the noise and that in general, each bit added to the digital representation results in
a doubling of the number of quantization levels and a reduction in 6dB in the
quantization noise. (The PCM example was not covered in the lectures and will not be
examined)
Section 3.3.4 SNR Performance of Quantizers
This section was not covered in detail and will not be examined.
Section 3.4 Characterization of Communication Channels
There are many types of communication channels: wires optical fibers, radio etc. From
communications perspective the important characteristics are bandwidth and the signal to
noise ratio at the receiver which in turn depend on the attenuation of the channel and the
noise added.
Section 3.4.1 Frequency Domain characterization
(A lot of this section repeats material covered in 3.2.2). An example is given of the effect
of low pass filtering a square wave.
Section 3.4.2 Time Domain characterization
Channels can be characterized in the time domain as well as the frequency domain. In
Monash University Electrical and Computer Systems Engineering

the time domain the channel is characterized by its impulse response. The impulse
response of an ideal low pass filter is a sin(x)/x function (sinc function). Wide bandwidth
channels have short impulse responses. Narrow bandwidth channels have long impulse
responses. The zero intersymbol interference property of sinc functions is noted here,
though it would have been more appropriately covered in the next section as it is
property of a signal rather than a channel. Sinc functions go through zero at T spaced
intervals where T=1/2W and W is the bandwidth of the pulse.
Section 3.5 Fundamental Limits in Digital Transmission
This section considers baseband transmission. That is transmission at low frequencies
stretching down to zero frequency or approximately zero frequency. (Data carried on
higher frequencies using modulation are considered in a later section).
Section 3.5.1 Nyquist Signaling Rate
Nyquist Signalling rate Sinc functions are one of a family of functions (Nyquist pulses)
which have T spaced zeros. If Nyquist pulses are transmitted at intervals T, ( in other
words at the Nyquist signal rate) there is no intersymbol interference (ISI).
Section 3.5.2 Shannon Channel Capacity
This section justifies the Shannon Channel capacity formula by showing that as the
number of signaling levels is increased the probability of error due to noise increases.
(The probability of error in Gaussian noise in introduced here but considered in more
detail in the separate notes on Noise in Digital Communication Systems. Note that
equation (3.30) has the bit error rate proportional to 2*Q function, not 1*Q as in the
Noise notes. This is because here they are considering interior points where either
positive or negative noise can cause an error).

Topic 6 - Line Coding - Modems - Modulation

Section 3.6 Line Coding
Line Coding Binary ones and zeros can be represented by electrical signals in a variety of
ways. For example positive and negative voltages (bipolar) or positive only (polar or
unipolar). The information can be carried by the voltage level or a difference in voltage
levels (differential coding). Different line codes have different average power levels, DC
components, spectra or are easy or difficult to synchronize to. Some carry higher date
rates for the same bandwidth. The linecodes considered were unipolar NRZ, polar NRZ,
NRZ Inverted (differential coding), Bipolar encoding, Manchester encoding and
Differential Manchester coding. In Manchester coding where 1 data bit is mapped onto 2
transmitted bits. This is an example of a more general class of mBnB codes where m data
bits are mapped onto n transmitted bits.
Section 3.7 Modems and Digital Modulation
Many practical channels are bandpass (rather than baseband). They can transmit a range
of frequencies centred on a center frequency fc. For data to be transmitted through these
channels it must be modulated onto a carrier so that the bandwidth of the signal is within
the bandwidth of the channel. For example broadcast radio and television is modulated
onto radio frequency carrier signals. Modulation is also used so that different signals can
be transmitted through the same medium for example different television channels are
all transmitted at the same time by using different carrier frequencies.
The amplitude, phase or frequency of the carrier signal can be varied to represent the
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information to be transmitted
The textbook considers only digital modulation but the lectures considered analog
modulation as well and notes called Sums and Products of Sines and Cosines and their
use in modulation theory covered modulation in greater detail. The equations for
amplitude modulation (AM) of a carrier frequency by a sinusoidal message signal were
derived showing that an AM signal has an upper and lower sideband and a component at
the carrier frequency.
The book considers amplitude shift keying, frequency shift keying and phase shift keying
where the carrier wave is modulated by a square binary signal representing the data.
Section 3.7.1 Binary Phase modulation
The book shows how a phase modulated signal can be generated and demodulated. The
Sums and products notes cover phase modulation in greater detail showing how
multiplication by the data signal results in sum and difference frequencies and how the
signal can be demodulated by multiplying by a locally generated carrier signal and
filtering out the high (sum) frequencies.
Section 3.7.2 QAM and Signal Constellations
Most new communication systems use quadrature amplitude modulation (QAM) rather
than binary phase modulation as it allows twice as much data to be carried on the same
bandwidth. One stream of data is modulated onto a cosine wave carrier and a second on
to a sine wave carrier. Both streams of data can be recovered at the receiver by
multiplying by locally generated versions of the cosine and sine wave carrier. The Sums
and products notes show in more detail how the two data streams can be recovered.
Because two streams of digital data are used in QAM, the possible values of data being
transmitted in any symbol period can be represented as a point on a two dimensional
plane. This is a constellation diagram. Constellation diagrams for 4QAM. 16QAM and
various PSK constellations were shown.
Section 3.7.3 Telephone Modem Standards
This section was not covered in great detail and will not be on the exam. In the slides an
example from the digital video broadcasting standard was given. This will not be on the
exam.

Topic 7 - MAC and LANs

Chapter 6 - Medium Access Control protocols and Local Area Networks: Section 6.1
Multiple access communications, Section 6.2.3 Carrier Sense Multiple Access (CSMA)
algorithm, Section 6.2.4 Carrier Sense Multiple Access with Collision Detection
(CSMA/CD) algorithm, Section 6.3.3 Token passing rings, Section 6.4.1 TDMA, Section
6.4.2 FDMA, Section 6.6 LAN protocols, Section 6.7 Ethernet and IEEE 802.3 LAN standard,
Section 6.11.1 Transparent bridges, Section 6.11.4 Virtual LAN
Means of sharing a transmission medium. Importance of delay-bandwidth product.
MAC protocols:
ALOHA and slotted ALOHA
Operation of CSMA and CSMA/CD
Concepts and purpose behind binary backoff
What is a LAN? Functions of MAC and LLC sublayers.
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Ethernet:
Header structures of IEEE 802.3 and Ethernet II types
Hubs, bridges, repeaters, routers and switches
How delay-bandwidth product affects scalability
Basic features of Fast, Gigabit and 10 Gigabit Ethernet types
How bridges learn adaptively and build their look-up tables, and avoid loops through
spanning tree algorithm
VLANs

Topic 8 - Properties of media

Section 3.8 Properties of Media and Digital Transmission systems
This section considered the properties of different transmission media like copper cables
and optical fiber. The attenuation of channels is often measured in dB which is a
logarithmic scale which measure relative power. Power relative to 1mW is measured in
dBm. The book puts some emphasis on delay and on the differing attenuation properties
of wired and wireless media. Delay and the difference in attenuation properties of wired
and wireless media were not given great emphasis in the course and will not be on the
exam.
A number of applications are given throughout section 3.8, while these are useful
background knowledge they were not covered in lectures and will not be on the exam.
Wireless media allow users to be mobile (e.g. mobile phones) but have limited available
bandwidth and different users can interfere with each other. Wired media (optical and
electrical) have much greater available bandwidth and interference is reduced or
eliminated but it costs much more to set up a fixed wired communication system and
users cannot be mobile.
Section 3.8.1 Twisted Pair
Pairs of wires twisted together to reduce interference. Attenuation increases with
frequency used for telephones and Ethernet.
Section 3.8.2 Coaxial cables
Inner and outer cylindrical conductors, lower interference and higher bandwidth than
twisted pair but much more expensive.
Section 3.8.3 Optical Fiber
Optical fibers are now used extensively as they can transport very high bit rates with very
low errors and very little signal attenuation. In an optical fiber a core is surrounded by
cladding of lower refractive index so that total internal reflection of the light occurs and
no light escapes from the fiber. The attenuation of fiber depends on the light frequency
being used so frequencies with low attenuation are normally used. There are two general
categories of optical fiber multimode optical fiber and single mode optical fibers.
Wavelength Division Multiplexing (WDM) can be used to greatly increase the capacity
of optical fibers. Many different colours of light are transmitted along the same fiber
each colour carrying different data.
Optical fiber is more difficult to connect than copper wires and it cannot be bent round
sharp corners
Section 3.8.4 Radio Transmission
There is a very large radio spectrum. Different frequencies have different characteristics,
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in particular low frequencies bend more so are better for transmission over long
distances.
Section 3.8.5 Infrared Light
This section was not covered in lectures and will not be examined.

Topic 9 - Noise in Digital Communication Systems

This is not in the textbook, but a set of notes closely aligned to the lecture slides is available on
Moodle covering the topic.

Noise is present in all communication systems. Noise is a random process. To describe

noise need to describe its probability density function (pdf) and its power spectral
density (psd) function. The form of noise most commonly considered in digital
communication systems is Gaussian noise. This has a Gaussian or normal pdf (often
called a bell curve because of its shape). To calculate the probability that noise has a
value within a particular range need to calculate the area under the appropriate section of
the Gaussian pdf (the bell curve). A Gaussian curve is completely described by its
mean and standard deviation. In the lectures a table of Q(x) was used to calculate bit
error rates. If there is a question requiring calculation of bit error rates in the exam, a
table of Q(x) will be provided.
The frequency properties of a random signal are given by its power spectral density. The
details of the calculation of power spectral density and its relationship to autocorrelation
will not be on the examination.
Additive white Gaussian noise (AWGN) is a useful concept in communications theory
although it cannot be achieved in practice (infinite bandwidth and infinite power).
AWGN has equal power at all frequencies from 0 to infinity.
Errors in a communication system due to Gaussian noise depend on the signal to noise
ratio and the form of the receiver.
Examples considered were
A receiver which simply sampled the received signal and based the bit decisions on
these samples
A receiver which applied matched filtering before sampling.
Matched filtering the optimum filter in a receiver if the noise is AWGN is a matched
filter. The impulse response of a matched filter is the time reversed of the signal used to
represent data. The output signal to noise ratio of a matched filter does not depend on the
pulse shape so the pulse shape can be chosen to optimize other factors (bandwidth, easy
of generation)
Bit error rate also depends on form of signalling (unipolar, bipolar, orthogonal). Bit error
rates fall off rapidly with increasing signal to noise ratio.

Topic 10 - OFDM
This is not in the textbook, and nor are there any notes provided beyond the lecture slides. If you'd
like an introduction to the subject, though, a good one is provided in Jean Armstrong's (award
winning!) paper "OFDM for Optical Communications", from the IEEE JOURNAL OF LIGHTWAVE
TECHNOLOGY, VOL. 27, NO. 3, FEBRUARY 1, 2009, pp 189-204. (Should be available from IEEE
Explore via the library).
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Orthogonal Frequency Division Multiplexing (OFDM) is used in many new digital

communication systems because it can be used in multipath channels that is channels
where the received signal is the sum of multiple copies of the transmitted signal with
different delays. OFDM transmits data in parallel on many narrow frequency bands.
OFDM signals can be readily recovered despite multipath distortion because
Parallel transmission means each symbol period is much longer
A cyclic prefix is added to each symbol so that multipath distortion causes only
frequency selective fading not intersymbol interference.
The details of OFDM transmitters and receivers will not be on the exam.

Topic 11 - Circuit Switching

Section 4.1 Multiplexing
Basic concept of multiplexing sharing a resource among different users.
Section 4.1.1 Frequency Division Multiplexing
Different users use different frequencies e.g. different television channels on different
frequencies.
Section 4.1.2 Time Division Multiplexing
Different users allocated different time slots. Some framing information is required so that
receiver knows which bits belong to which user.
Multiplex hierarchies of different telephone standards (US and CCITT)
Difficult in combining different tributary streams of slightly different data rates. Allocate
slots in output for pulse stuffing. Information about whether a slot contains information
or not must also be transmitted. Makes it difficult to extract individual data streams from
a higher order multiplex.
Section 4.1.3 Wavelength division Multiplexing
Different users use different wavelength of light to transmit information along one optical
fiber (same concept as frequency division multiplexing, but optical frequencies are usually
measured in terms of wavelength rather than frequency.)
Section 4.2 SONET
New way of combining different data streams so that individual streams can be easily
extracted. Basis of modern networks which can be reconfigured in the case of faults. Sonet
was only mentioned briefly. Sections 4.2.1 and 4.2.2 were not covered
Section 4.4 Circuit switches
In circuit switching an end-to-end circuit is dedicated to the connection. (Circuit may be time
division multiplexed but in that case given time slot is used only for the particular
connection). Circuit switching was well suited to voice calls but has been replaced by packet
switching in most data applications. Sections 4.4.1 and 4.4.2 were not covered (note this is
different from 2006).
Sections 4.5, 4.6 and 4.7
These were not covered in detail in the lectures and will not be examined. Signalling system
number 7 was mentioned briefly but will not be examined.
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Section 4.8 Cellular Telephone Networks

This was only covered briefly. The only aspect which is examinable is the concept of
frequency reuse in different cells.