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Communication Systems/Print Version

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Introduction
This book will eventually cover a large number of topics in the field of electrical communications. The reader will also require a knowledge of Time and
Frequency Domain representations, which is covered in-depth in the Signals and Systems book. This book will, by necessity, touch on a number of different
areas of study, and as such is more than just a text for aspiring Electrical Engineers. This book will discuss topics of analog communication schemes,
computer programming, network architectures, information infrastructures, communications circuit analysis, and many other topics. It is a large book, and
varied, but it should be useful to any person interested in learning about an existing communication scheme, or in building their own. Where previous
Electrical Engineering books were grounded in theory (notably the Signals and Systems book), this book will contain a lot of information on current
standards, and actual implementations. It will discuss how current networks and current transmission schemes work, and may even include information for
the intrepid engineer to create their own versions of each.

This book is still in an early stage of development. Many topics do not yet have pages, and many of the current pages are stubs. Any help would be greatly
appreciated.

Table of Contents
Introduction
People are prone to take for granted the fact that modern technology allows us to transmit data at nearly the speed of light to locations that are very far away.
200 years ago, it would be deemed preposterous to think that we could transmit webpages from China to Mexico in less then a second. It would seem equally
preposterous to think that people with cellphones could be talking to each other, clear as day, from miles away. Today, these things are so common, that we
accept them without even asking how these miracles are possible.

What is Communications?

Communications is the field of study concerned with the transmission of information through various means. It can also be defined as technology employed
in transmitting messages. It can also be defined as the inter-transmitting the content of data (speech, signals, pulses etc.) from one node to another.

Who is This Book For?

This book is for people who have read the Signals and Systems wikibook, or an equivalent source of the information. Topics considered in this book will rely
heavily on knowledge of Fourier Domain representation and the Fourier Transform. This book can be used to accompany a number of different classes
spanning the 3rd and fourth years in a study of electrical engineering. Knowledge of integral and differential calculus is assumed. The reader may benefit
from knowledge of such topics as semiconductors, electromagnetic wave propagation, etc, although these topics are not necessary to read and understand the
information in this book.

What this Book will Cover

This book is going to take a look at nearly all facets of electrical communications, from the shape of the electrical signals, to the issues behind massive
networks. It makes little sense to be discussing these subjects outside the realm of current examples. We have the internet, so in discussing issues concerning
digital networks, it makes good sense to reference these issues to the internet. Likewise, this book will attempt to touch on, at least briefly, every major
electrical communications network that people deal with on a daily basis. From AM radio to the internet, from DSL to cable TV, this book will attempt to
show how the concepts discussed apply to the real world.

This book also acknowledges a simple point: It is easier to discuss the signals and the networks simultaneously. For this kind of task to be undertaken in a
paper book would require hundreds, if not thousands of printed pages, but through the miracle of wikimedia, all this information can be brought together in a
single, convenient location.

This book would like to actively solicit help from anybody who has experience with any of these concepts: Computer Engineers, Communications Engineers,
Computer Programmers, Network Administrators, IT Professionals. Also, this book may cover all these topics, but the reader doesn't need to have prior
knowledge of all these disciplines to advance. Information will be developed as completely as possible in the text, and links to other information sources will
be provided as needed.

Where to Go From Here

Since this book is designed for a junior and senior year of study, there aren't necessarily many topics that will logically follow this book. After reading and
understanding this material, the next logical step for the interested engineer is either industry or graduate school. Once in graduate school, there are a number
of different areas to concentrate study in. In industry, the number is even higher.

Division of Material
Admittedly, this is a very large topic, one that can span not only multiple printed books, but also multiple bookshelves. It could then be asked "Why don't we
split this book into 2 or more smaller books?" This seems like a good idea on the surface, but you have to consider exactly where the division would take
place. Some would say that we could easily divide the information between "Analog and Digital" lines, or we could divide up into "Signals and Systems"
books, or we could even split up into "Transmissions and Networks" Books. But in all these possible divisions, we are settling for having related information
in more then 1 place.

Analog and Digital

It seems most logical that we divide this material along the lines of analog information and digital information. After all, this is a "digital world", and aspiring

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communications engineers should be able to weed out the old information quickly and easily. However, what many people don't realize is that digital
methods are simply a subset of analog methods with more stringent requirements. Digital transmissions are done using techniques perfected in analog radio
and TV broadcasts. Digital computer modems are sending information over the old analog phone networks. Digital transmissions are analyzed using analog
mathematical concepts such as modulation, SNR (signal to noise ratio), Bandwidth, Frequency Domain, etc... For these reasons, we can simplify both
discussions by keeping them in the same book.

Signals and Systems

Perhaps we should divide the book in terms of the signals that are being sent, and the systems that are physically doing the sending. This makes some sense,
except that it is impossible to design an appropriate signal without understanding the restrictions of the underlying network that it will be sent on. Also, once
we develop a signal, we need to develop transmitters and receivers to send them, and those are physical systems as well.

Systems Approach

It is a bit confusing to be writing a book about Communication Systems and also considering the pedagogical Systems Approach. Although using the same
word, they are not quite the same thing.

This approach is almost identical to the description above (Signals & Systems) except that it is not limited to the consideration of signals (common in many
university texts), but can include other technological drivers (codecs, lasers, and other components).

In this case we give a brief overview of different communication systems (voice, data, cellular, satellite etc.) so that students will have a context in which to
place the more detailed (and often generic) information. Then we can then zoom in on the mathematical and technological details to see how these systems do
their magic. This lends itself quite well to technical subjects since the basic systems (or mathematics) change relatively slowly, but the underlying technology
can offen change rapidly and take unexpected terns.

I would like to suggest that the table of contents in this book be rearranged to reflect this pedagogical approach: Systems examples first, followed by the
details.

Why would anyone want to study (tele)communications?

Telecommunications is an alluring industry with a provocative history filled with eccentric personalities: Bell, Heavyside, Kelvin, Brunel and many others. It
is fraught with adventure and danger: adventure spanning space and time; danger ranging from the remote depths of the ocean floor to deep space, from the
boardrooms of AT&T to the Hong Kong stock exchange.

Telecommunications has been heralded as a modern Messiah and cursed as a pathetic sham. It has created and destroyed empires and institutions. It has
proclaimed the global village while sponsoring destructive nationalism. It has come to ordinary people, but has remained largely in the control of the ‘media’
and even 'big brother'. Experts will soon have us all traveling down a techno information highway, destination — unknown.

Telecommunications has become the lifeblood of modern civilization. Besides all that, there’s big bucks in it

About This Book

There are a few points about this book that are worth mentioning:

Real-world examples will appear in these boxes

1. The programming parts of this book will not use any particular language, although we may consider particular languages in dedicated chapters.

This page will attempt to show some of the basic history of electrical communication systems.

Chronology

1831 Samuel Morse invents the first repeater and the telegraph is born

1837 Charles Wheatstone patents "electric telegraph"

1849 England to France telegraph cable goes into service -- and fails after 8 days.

1850 Morse patents "clicking" telegraph.

1851 England-France commercial telegraph service begins. This one uses gutta-percha, and survives.

1858 August 18 - First transatlantic telegraph messages sent by the Atlantic Telegraph Co. The cable deteriorated quickly, and failed after 3 weeks.

1861 The first transcontinental telegraph line is completed

1865 The first trans-Atlantic cable goes in service

1868 First commercially successful transatlantic telegraph cable completed between UK and Canada, with land extension to USA. The message rate is 2
words per minute.

1870 The trans-Atlantic message rate is increased to 20 words per minute.

1874 Baudot invents a practical Time Division Multiplexing scheme for telegraph. Uses 5-bit codes & 6 time slots -- 90 bps max. rate. Both Western Union
and Murray would use this as the basis of multiplex telegraph systems.

1875 Typewriter invented.

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1876 Alexander Graham Bell and Elisa Grey independently invent the telephone (although it may have been invented by Antonio Meucci as early as 1857)

1877 Bell attempts to use telephone over the Atlantic telegraph cable. The attempt fails.

1880 Oliver Heaviside's analysis shows that a uniform addition of inductance into a cable would produce distortionless transmission.

1883 Test calls placed over five miles of under-water cable.

1884 - San Francisco-Oakland gutta-percha cable begins telephone service.

1885 Alexander Graham Bell incorporated AT&T

1885 James Clerk Maxwell predicts the existence of radio waves

1887 Heinrich Hertz verifies the existence of radio waves

1889 Almon Brown Strowger invents the first automated telephone switch

1895 Gugliemo Marconi invents the first radio transmitter/receiver

1901 Gugliemo Marconi transmits the first radio signal across the Atlantic 1901 Donald Murray links typewriter to high-speed multiplex system, later used
by Western Union

1905 The first audio broadcast is made

1910 Cheasapeake Bay cable is first to use loading coils underwater

1911 The first broadcast license is issued in the US

1912 Hundreds on the Titanic were saved due to wireless

1915 USA transcontinental telephone service begins (NY-San Francisco).

1924 The first video signal is broadcast

1927 First commercial transatlantic radiotelephone service begins

1929 The CRT display tube is invented

1935 Edwin Armstrong invents FM

1939 The blitzkrieg and WW II are made possible by wireless

1946 The first mobile radio system goes into service in St. Louis

1948 The transistor is invented

1950 Repeatered submarine cable used on Key West-Havana route.

1956 The first trans-Atlantic telephone cable, TAT-1, goes into operation. It uses 1608 vacuum tubes.

1957 The first artificial satellite, Sputnik goes into orbit

1968 The Carterphone decision allows private devices to be attached to the telephone

1984 The MFJ (Modification of Final Judgement) takes effect and the Bell system is broken up

1986 The first transAtlantic fiber optic cable goes into service

Claude Shannon
Harry Nyquist
Section 1: Communications Basics

It is important to know the difference between a baseband signal, and a broad band signal. In the Fourier Domain, a baseband signal is a signal that occupies
the frequency range from 0Hz up to a certain cutoff. It is called the baseband because it occupies the base, or the lowest range of the spectrum.

In contrast, a broadband signal is a signal which does not occupy the lowest range, but instead a higher range, 1MHz to 3MHz, for example. A wire may have
only one baseband signal, but it may hold any number of broadband signals, because they can occur anywhere in the spectrum.

Wideband vs Narrowband
Frequency Spectrum
A graphical representation of the various frequency components on a given transmission medium is called a frequency spectrum.

Consider a situation where there are multiple signals which would all like to use the same wire (or medium). For instance, a telephone company wants
multiple signals on the same wire at the same time. It certainly would save a great deal of space and money by doing this, not to mention time by not having
to install new wires. How would they be able to do this? One simple answer is known as Time-Division Multiplexing.

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Time Division Multiplexing

Time-Division Multiplexing (TDM) is a convenient method for combining various digital signals onto a single transmission media such as wires, fiber
optics or even radio. These signals may be interleaved at the bit, byte, or some other level. The resulting pattern may be transmitted directly, as in digital
carrier systems, or passed through a modem to allow the data to pass over an analog network. Digital data is generally organized into frames for transmission
and individual users assigned a time slot, during which frames may be sent. If a user requires a higher data rate than that provided by a single channel,
multiple time slots can be assigned.

Digital transmission schemes in North America and Europe have developed along two slightly different paths, leading to considerable incompatibility
between the networks found on the two continents.

BRA (basic rate access) is a single digitized voice channel, the basic unit of digital multiplexing.

North American TDM

The various transmission rates are not integral numbers of the basic rate. This is because additional framing and synchronization bits are required at every
multiplexing level.

In North America, the basic digital channel format is known as DS-0. These are grouped into frames of 24 channels each. A concatenation of 24 channels and

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a start bit is called a frame. Groups of 12 frames are called multiframes or superframes. These vary the start bit to aid in synchronizing the link and add
signaling bits to pass control messages.

S Bit Synchronization

The S bit is used to identify the start of a DS-1 frame. There are 8 thousand S bits per second. They have an encoded pattern, to aid in locating channel
position within the frame.

This forms a regular pattern of 1 0 1 0 1 0 for the odd frames and 0 0 1 1 1 0 for the even frames. Additional synchronization information is encoded in the
DS-1 frame when it is used for digital data applications, so lock is more readily acquired and maintained.

For data customers, channel 24 is reserved as a special sync byte, and bit 8 of the other channels is used to indicate if the remaining 7 bits are user data or
system control information. Under such conditions, the customer has an effective channel capacity of 56 Kbps.

To meet the needs of low speed customers, an additional bit is robbed to support sub-rate multiplexer synchronization, leaving 6 x 8 Kbps = 48 Kbps
available. Each DS-0 can be utilized as:

• 5 x 9.6 Kbps channels or

• 10 x 4.8 Kbps channels or
• 20 x 2.48 Kbps channels.

In the DS-2 format, 4 DS-1 links are interleaved, 12 bits at a time. An additional 136 Kbps is added for framing and control functions resulting in a total bit
rate of 6.312 Mbps.

Signaling

Signaling provides control and routing information. Two bits, called the A and B bits, are taken from each channel in frames 6 and 12 in the multiframe. The
A bit is the least significant bit in each channel in frame 6, and the B bit is the least significant bit in each channel in frame 12. This provides a signaling rate
of 666 2/3 bps per channel.

The quality of voice transmission is not noticeably affected when 2% of the signal is robbed for signaling. For data, it may be a different story. If the data is
encoded in an analog format such as FSK or PSK, then robbing bits is of no consequence, but if the data is already in digital form, then robbing bits results in
unacceptable error rates. It is for this reason that in North America, a 64 Kbps clear channel cannot readily be switched through the PSTN. This means that
data customers are limited to 56 Kbps clear channels. This simple condition has a profound effect on the development of new services such as ISDN. In most
facilities, the A and B bits represent the status of the telephone hook switch, and correspond to the M lead on the E&M interface of the calling party.

ESF

CCITT has modified the North American digital hierarchy for the deployment of ISDN, by means of recommendation G.704. ESF consists of 24 DS-0
channels in a frame, but groups them into a 24-frame multiframe instead of the usual 12-frame multiframe.

The S bit is renamed the F bit, but only 1/4 of them are used for synchronization. This is possible because of improvements in frame search techniques and
allows more signaling states to be defined.

Bit robbing is still used for signaling over an ESF link, but with the advent of ISDN, it will not be permitted. Instead, channel 24 is used to support a D
channel.

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Typical T1 CPE Application

The large telecom carriers are not the only ones who deploy high-speed TDM facilities. In many cases, heavy users of voice or data services can reduce their
transmission costs by concentrating their numerous low speed lines on to a high speed facility.

There are many types of T1 multiplexers available today. Some are relatively simple devices, while others allow for channel concatenation, thus supporting a
wide range of data rates. The ability to support multiple DS-0s allows for easy facilitation of such protocols as the video teleconferencing standard, Px64.

Multiplexers

Multiplexing units are often designated by the generic term Mab where a is input DS level and b is the output DS level. Thus, an M13 multiplexer combines
28 DS–1s into a single DS–3 and an M23 multiplexer combines 7 DS–2s into a single DS–3.

ZBTSI

ZBTSI (zero byte time slot interchange) is used on DS–4 links. Four DS-1 frames are loaded into a register, and renumbered 1–96. If there are any empty
slots [all zeros], the first framing bit is inverted and all blank slots are relocated to the front of the frame. Channel 1 is then loaded with a 7-bit number
corresponding to the original position of the first empty slot. Bit 8 used to indicate whether the following channel contains user information or another
address for an empty slot.

If there is a second vacancy, bit 8 in the previous channel is set, and the empty slot address is placed in channel 2. This process continues until all empty
positions are filled.

The decoding process at the receiver is done in reverse. Borrowing 1 in 4 framing bits for this system is not enough to cause loss of synchronization and
provides a 64 Kbps clear channel to the end-user.

European TDM Carriers

European systems were developed along slightly different principles. The 64 Kbps channel is still the basic unit, but signaling is not included in each channel.

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Instead, common channel signaling is used. In a level 1 carrier, channels 0 and 16 are reserved for signaling and control. This subtle difference means that
European systems did not experience the toll fraud and 56 k bottlenecks common to North American systems, and they experience a much larger penetration
of ISDN services.

Zero Substitutions
In order to prevent transmission systems from loosing lock on the data stream, it is necessary to avoid long strings of zeros. One of the most effective ways of
doing this is to replace the zeros with a predetermined code. This substitution must be done in such a way that the receiver can identify it and strip it off
before passing the data stream to the client.

AMI provides a simple means of detecting substitutions. In the normal course of events, alternate marks are inverted. Therefor, deliberately inducing a
bipolarvariation at the transmitter can alert the receiver of a substitution. However, a single violation is indistinguishable from a transmission error.
Consequently, some additional condition must also occur.

There are two common methods to create a second condition:

· Create a second bipolar violation in the opposite direction, within a specified time. This has the effect of keeping the average signal level at zero.
· Count the number of marks from the last substitution to predict the next type of violation

B6ZS

B6ZS (binary six zero substitution) is used on T2 AMI transmission links.

Synchronization can be maintained by replacing strings of zeros with bipolar violations. Since alternate marks have alternate polarity, two consecutive pulses
of the same polarity constitute a violation. Therefor, violations can be substituted for strings of zeros, and the receiver can determine where substitutions were
made.

Since the last mark may have been either positive (+) or negative (-), there are two types of substitutions:

These substitutions force two consecutive violations. A single bit error does not create this condition.

B8ZS

This scheme uses the same substitution as B6ZS.

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B3ZS

B3ZS is more involved than B6ZS, and is used on DS–3 carrier systems. The substitution is not only dependent on the polarity of the last mark, but also on
the number of marks since the last substitution.

HDB3

HDB3 (high density binary 3) introduces bipolar violations when four consecutive zeros occur. It can therefore also be called B4ZS. The second and thirds
zeros are left unchanged, but the fourth zero is given the same polarity as the last mark. The first zero may be modified to a one to make sure that successive
violations are of alternate polarity.

HDB3 is used in Europe. Violation, or V pulses are injected after three consecutive zeros. The fourth zero is given the same polarity as the last mark. In the
event of long strings of zeros occurring, a succession of single polarity pulses would occur, and a dc offset would build-up.

To prevent this, the first zero in a group of 4, may be modified to a 1. This B or balancing pulse assures that successive violations are of alternate polarity.

Block Code Substitution

These schemes operate on bytes rather than a bit at a time. Some transmit the signal as binary levels, but most use multi-level pulses. Some authors categorize
these as line codes.

A binary block code has the designation nBmB, where n input bits are encoded into m output bits. The most common of these is the 3B4B code.

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In Europe 4B3T, which encodes 4 binary bits into 3 ternary levels, has been selected as the BRA for ISDN. In North America, 2B1Q which encodes 2 binary
bits into 1 quaternary level has been selected for BRA.

Some block codes do not generate multilevel pulses. For example, 24B1P or 24B25B simply adds a P or parity bit to a 24 bit block.

Benefits of TDM
TDM is all about cost: fewer wires and simpler receivers are used to transmit data from multiple sources to multiple destinations. TDM also uses less
bandwidth than Frequency-Division Multiplexing (FDM) signals, unless the bitrate is increased, which will subsequently increase the necessary bandwidth of
the transmission.

Synchronous TDM
Synchronous TDM is a system where the transmitter and the receiver both know exactly which signal is being sent. Consider the following diagram:

Signal A ---> |---| |A|B|C|A|B|C| |------| ---> Signal A

Signal B ---> |TDM| --------------> |De-TDM| ---> Signal B
Signal C ---> |---| |------| ---> Signal C

In this system, starting at time-slice 0, every third time-slice is reserved for Signal A; starting at time-slice 1, every third time-slice is reserved for Signal B;
and starting at time-slice 2, every third time-slice is reserved for Signal C. In this situation, the receiver (De-TDM) needs only to switch after the signal on
each time-slice is received.

Statistical TDM
Synchronous TDM is beneficial because the receiver and transmitter can both cost very little. However, consider the most well-known network: the Internet.
In the Internet, a given computer might have a data rate of 1kbps when hardly anything is happening, but might have a data rate of 100kbps when
downloading a large file from a fast server. How are the time-slices divided in this instance? If every time slice is made big enough to hold 100Kbps, when
the computer isn't downloading any data, all of that time and electricity will be wasted. If every time-slice is only big enough for the minimum case, the time
required to download bigger files will be greatly increased.

The solution to this problem is called Statistical TDM, and is the solution that the Internet currently uses. In Statistical TDM, each data item, known as the
payload (we used time-slices to describe these earlier), is appended with a certain amount of information about who sent it, and who is supposed to receive it
(the header). The combination of a payload and a header is called a packet. Packets are like envelopes in the traditional "snail mail" system: Each packet
contains a destination address and a return address as well as some enclosed data. Because of this, we know where each packet was sent from and where it is
going.

The downside to statistical TDM is that the sender needs to be smart enough to write a header, and the receiver needs to be smart enough to read the header
and (if the packet is to be forwarded,) send the packet toward its destination.

Link Utilization

Statistical multiplexing attempts to maximize the use of a communication path. The study of this is often called queuing theory. A queue is simply a line of
customers or packets waiting to be served. Under most circumstances, the arrival rate is unpredictable and therefor follows a random or Poisson distribution
pattern, whereas the service time is constant.

The utilization or fraction of time actually used by a packet multiplexing system to process packets is given by:

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The queue length or average number of items waiting to be served is given by:

Example

A T1 link has been divided into a number of 9.6 Kbps channels and has a combined user data rate of 1.152 Mbps. Access to this channel is
offered to 100 customers, each requiring 9.6 Kbps data 20% of the time. If the user arrival time is strictly random find the T1 link utilization.

Solution

The utilization or fraction of time used by the system to process packets is given by:

A 24 channel system dedicated to DATA, can place five 9.6 Kbps customers in each of 23 channels, for a total of 115 customers. In the above
statistical link, 100 customers created an average utilization of 0.167 and were easily fitted, with room to spare if they transmit on the average
20% of the time. If however, the customer usage were not randomly distributed, then the above analysis would have to be modified.

This example shows the potential for statistical multiplexing. If channels were assigned on a demand basis (only when the customer had something to send), a
single T1 may be able to support hundreds of low volume users.

A utilization above 0.8 is undesirable in a statistical system, since the slightest variation in customer requests for service would lead to buffer overflow.
Service providers carefully monitor delay and utilization and assign customers to maximize utilization and minimize cost.

Packets
Packets will be discussed in greater detail once we start talking about digital networks (specifically the Internet). Packet headers not only contain address
information, but may also include a number of different fields that will display information about the packet. Many headers contain error-checking
information (checksum, Cyclic Redundancy Check) that enables the receiver to check if the packet has had any errors due to interference, such as electrical
noise.

Duty Cycles
Duty cycle is defined as " the time that is effectively used to send or receive the data, expressed as a percentage of total period of time." The more the duty
cycle , the more effective transmission or reception.

We can define the pulse width, τ, as being the time that a bit occupies from within it's total alloted bit-time Tb. If we have a duty cycle of D, we can define the
pulse width as:

τ = DTb

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Where:

The pulse width is equal to the bit time if we are using a 100% duty cycle.

Introduction
It turns out that many wires have a much higher bandwidth than is needed for the signals that they are currently carrying. Analog Telephone transmissions,
for instance, require only 3 000 Hz of bandwidth to transmit human voice signals. Over short distances, however, twisted-pair telephone wire has an available
bandwidth of nearly 100 000 Hz!

There are several terrestrial radio based communications systems deployed today. They include:

• Cellular radio
• Mobile radio
• Digital microwave radio

Mobile radio service was first introduced in the St. Louis in 1946. This system was essentially a radio dispatching system with an operator who was able
topatch the caller to the PSTN via a switchboard. Later, an improved mobile telephone system, IMTS, allowed customers to dial their own calls without the
need for an operator. This in turn developed into the cellular radio networks we see today.

The long haul PSTNs and packet data networks use a wide variety of transmission media including

• Terrestrial microwave
• Satellite microwave
• Fiber optics
• Coaxial cable

In this section, we will be concerned with terrestrial microwave systems. Originally, microwave links used FDM exclusively as the access technique, but
recent developments are changing analog systems to digital where TDM is more appropriate.

Fixed Access Assignment

Three basic methods can be used to combine customers on to fixed channel radio links:

• FDMA - (frequency division multiple access) analog or digital

• TDMA - (time division multiple access) three conversation paths are time division multiplexed in 6.7 mSec time slots on a single carrier.
• CDMA - (code division multiple access) this uses spread spectrum techniques to increase the subscriber density. The transmitter hops through a
pseudo-random sequence of frequencies. The receiver is given the sequence list and is able to follow the transmitter. As more customers are
added to the system, the signal to noise will gradually degrade. This is in contrast to AMPS where customers are denied access once all of the
frequencies are assigned code division multiple access [digital only]

What is FDM?

Frequency Division Multiplexing (FDM) allows engineers to utilize the extra space in each wire to carry more than one signal. By frequency-shifting some
signals by a certain amount, engineers can shift the spectrum of that signal up into the unused band on that wire. In this way, multiple signals can be carried
on the same wire, without having to divy up time-slices as in Time-Division Multiplexing schemes.In analog transmission, signals are commonly multiplexed
using frequency-division multiplexing (FDM), in which the carrier bandwidth is divided into subchannels of different frequency widths, each carrying a
signal at the same time in parallel

Broadcast radio and television channels are separated in the frequency spectrum using FDM. Each individual channel
occupies a finite frequency range, typically some multiple of a given base frequency.

Traditional terrestrial microwave and satellite links employ FDM. Although FDM in telecommunications is being reduced, several systems will continue to
use this technique, namely: broadcast & cable TV, and commercial & cellular radio.

Analog Carrier Systems

The standard telephony voice band [300 – 3400 Hz] is heterodyned and stacked on high frequency carriers by single sideband amplitude modulation. This is
the most bandwidth efficient scheme possible.

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The analog voice channels are pre-grouped into threes and heterodyned on carriers at 12, 16, and 20 KHz. The resulting upper sidebands of four such
pregroups are then heterodyned on carriers at 84, 96, 108, and 120 KHz to form a 12-channel group.

Since the lower sideband is selected in the second mixing stage, the channel sequence is reversed and a frequency inversion occurs within each channel.

This process can continue until the available bandwidth on the coaxial cable or microwave link is exhausted.

In the North American system, there are:

• 12 channels per group

• 5 groups per supergroup
• 10 super groups per mastergroup
• 6 master groups per jumbogroup

In the European CCITT system, there are:

• 12 channels per group

• 5 groups per supergroup
• 5 super groups per mastergroup
• 3 master groups per supermastergroup

There are other FDM schemes including:

• L600 - 600 voice channels 60–2788 KHz

• U600 - 600 voice channels 564–2084 KHz
• L3 - 1860 voice channels 312–8284 KHz, comprised of 3 mastergroups and a supergroup
• L4 - 3600 voice channels, comprised of six U600s

Benefits of FDM
FDM allows engineers to transmit multiple data streams simultaneously over the same channel, at the expense of bandwidth. To that extent, FDM provides a
trade-off: faster data for more bandwidth. Also, to demultiplex an FDM signal requires a series of bandpass filters to isolate each individual signal. Bandpass
filters are relatively complicated and expensive, therefore the receivers in an FDM system are generally expensive.

Examples of FDM
As an example of an FDM system, Commercial broadcast radio (AM and FM radio) simultaneously transmits multiple signals or "stations" over the airwaves.
These stations each get their own frequency band to use, and a radio can be tuned to receive each different station. Another good example is cable television,
which simultanously transmits every channel, and the TV "tunes in" to which channel it wants to watch.

Orthogonal FDM

Orthogonal Frequency Division Multiplexing (OFDM) is a more modern variant of FDM that uses orthogonal sub-carriers to transmit data that does not
overlap in the frequency spectrum and is able to be separated out using frequency methods. OFDM has a similar data rate to traditional FDM systems, but has
a higher resilience to disruptive channel conditions such as noise and channel fading.

Voltage Controlled Oscillators (VCO)

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A voltage-controlled oscillator (VCO) is a device that outputs a sinusoid of a frequency that is a function of the input voltage. VCOs are not time-invariant,
linear components. A complete study of how a VCO works will probably have to be relegated to a book on electromagnetic phenomena. This page will,
however, attempt to answer some of the basic questions about VCOs.

A basic VCO has input/output characteristics as such:

v(t) ----|VCO|----> sin(a[f + v(t)]t + O)

VCOs are often implemented using a special type of diode called a "Varactor". Varactors, when reverse-biased, produce a small amount of capacitance that
varies with the input voltage.

Phase-Locked Loops

If you are talking on your cellphone, and you are walking (or driving), the phase angle of your signal is going to change, as a function of your motion, at the
receiver. This is a fact of nature, and is unavoidable. The solution to this then, is to create a device which can "find" a signal of a particular frequency, negate
any phase changes in the signal, and output the clean wave, phase-change free. This device is called a Phase-Locked Loop (PLL), and can be implemented
using a VCO.

Purpose of VCO and PLL

VCO and PLL circuits are highly useful in modulating and demodulating systems. We will discuss the specifics of how VCO and PLL circuits are used in this
manner in future chapters.

Varactors
As a matter of purely professional interest, we will discuss varactors here. (VCO)

What is an Envelope Filter?

The envelope detector is a simple analog circuit that can be used to find the peaks in a quickly-changing waveform.
Envelope detectors are used in a variety of devices, specifically because passing a sinusoid through an envelope If anybody has some images that they
detector will supress the sinusoid. can upload, it would be much better
then these ASCII art things.
Circuit Diagram
In essence, an envelope filter has the following diagram:

o------+------+------o
+ | | +
\ (c)
vin /R | vout
\ |
- | | -
o------+------+------o

Where (c) represents a capacitor, and R is a resistor. Under zero input voltage (vin = 0), the capacitor carries no charge, and the resistor carries no current.
When vin is increased, the capacitor stores charge until it reaches capacity, and then the capacitor becomes an open circuit. At this point, all current in the
circuit is flowing through the resistor, R. As voltage decreases, the capacitor begins to discharge it's stored energy, slowing down the state change in the
circuit from high voltage to low voltage.

Positive Voltages
By inserting a diode at the beginning of this circuit, we can negate the effect of a sinusoid, dipping into negative voltage, and forcing the capacitor to
discharge faster:

diode
o-->|--+------+------o
+ | | +
\ (c)
vin /R | vout
\ |
- | | -
o------+------+------o

Purpose of Envelope Filters

Envelope filters help to find the outer bound of a signal that is changing in amplitude.

Envelope Filters are generally used with AM demodulation, discussed later.

(Envelope Detectors)

Modulation is a term that is going to be used very frequently in this book. So much in fact, that we could almost have renamed this book "Principals of
Modulation", without having to delete too many chapters. So, the logical question arises: What exactly is modulation?

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Definition
Modulation is a process of mixing a signal with a sinusoid to produce a new signal. This new signal, conceivably, will have certain benefits of an un-
modulated signal, especially during transmission. If we look at a general function for a sinusoid:

f(t) = Asin(ωt + φ)

we can see that this sinusoid has 3 parameters that can be altered, to affect the shape of the graph. The first term, A, is called the magnitude, or amplitude of
the sinusoid. The next term, ω is known as the frequency, and the last term, φ is known as the phase angle. All 3 parameters can be altered to transmit data.

The sinusoidal signal that is used in the modulation is known as the carrier signal, or simply "the carrier". The signal that is being modulated is known as the
"data signal". It is important to notice that a simple sinusoidal carrier contains no information of it's own.

Types of Modulation
There are 3 different types of modulation: Amplitude modulation, Frequency modulation, and Phase modulation. We will talk about the differences and uses
of all three in a later chapter.

Why Use Modulation?

Clearly the concept of modulation can be a little tricky, especially for the people who don't like trigonometry. Why then do we bother to use modulation at
all? To answer this question, let's consider a channel that essentially acts like a bandpass filter: The lowest frequency components and the highest frequency
components are attenuated or unusable, in some way. If we can't send low-frequency signals, then we need to shift our signal up the frequency ladder.
Modulation allows us to send a signal over a bandpass frequency range. If every signal gets it's own frequency range, then we can transmit multiple signals
simultaneously over a single channel, all using different frequency ranges. Another use of modulation is that it allows the efficient use of antenna, a baseband
signal to be transmitted will need an antenaa of huge length, whereas once modulated the antenna length will be reduced dramatically (antenna length almost
equal 1/10 of the signal wavelength

Examples
Think about your car radio. There are more then a dozen (or so) channels on the radio at any time, each with a given frequency: 100.1MHz, 102.5MHz etc...
Each channel gets a certain range (usually about 0.2MHz), and the entire station gets transmitted over that range. Modulation makes it all possible, because it
allows us to send voice and music (which are essentiall baseband signals) over a bandpass (or "Broadband") channel.

non-sinusoidal modulation
A sine wave at one frequency can separated from a sine wave at another frequency (or a cosine wave at the same frequency) because the two signals are
"orthogonal".

There are other sets of signals, such that every signal in the set is orthogonal to every other signal in the set.

A simple orthogonal set is time multiplexed division (TDM) -- only one transmitter is active at any one time.

Other more complicated sets of orthogonal waveforms -- Walsh codes and various pseudonoise codes such as Gold codes and maximum length sequences --
are also used in some communication systems.

The process of combining these waveforms with data signals is sometimes called "modulation", because it is so very similar to the way modulation combines
sine waves are with data signals.

further reading

Data Coding Theory/Spectrum Spreading

Wikipedia:Walsh code

Wikipedia:Gold code

Wikipedia:pseudonoise code

Wikipdia:maximum length sequence

There is lots of talk nowadays about buzzwords such as "Analog" and "Digital". Certainly, engineers who are interested in creating a new communication
system should understand the difference. Which is better, analog or digital? What is the difference? What are the pros and cons of each? This chapter will
look at the answers to some of these questions.

What are They?

What exactly is an analog signal, and what is a digital signal?

Analog
Analog signals are signals with continuous values. Analog signals are used in many systems, although the use of analog signals has declined with the
advent of cheap digital signals.

Digital
Digital signals are signals that are represented by binary numbers, "1" or "0". The 1 and 0 values can correspond to different discrete voltage values,
and any signal that doesnt quite fit into the scheme just gets rounded off.

What are the Pros and Cons?

Each paradigm has it's own benefits and problems.

Analog

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Analog systems are very tolerant to noise, make good use of bandwidth, and are easy to manipulate mathematically. However, analog signals require
hardware receivers and transmitters that are designed to perfectly fit the particular transmission. If you are working on a new system, and you decide to
change your analog signal, you need to completely change your transmitters and receivers.

Digital
Digital signals are very intolerant to noise, and digital signals can be completely corrupted in the presence of excess noise. In digital signals, noise
could cause a 1 to be interpreted as a 0 and vice versa, which makes the received data different than the original data. Imagine if the army transmitted a
position coordinate to a missile digitally, and a single bit was received in error? This single bit error could cause a missile to miss its target by miles.
Luckily, there are systems in place to prevent this sort of scenario, such as checksums and CRCs, which tell the receiver when a bit has been corrupted
and ask the transmitter to resend the data. The primary benefit of digital signals is that they can be handled by simple, standardized receivers and
transmitters, and the signal can be then dealt with in software (which is comparatively cheap to change).

The Difference between Digital and Discrete

Digital quantity may be either 0 or 1, but discrete may be any numerical value i.e 0,1....9.

Which is Better?
There is no way to say which type of signal is better or worse. Modern digital systems often require more expensive components (consider the difference in
price between old TV sets, and new HDTV sets), although lower-end digital systems can be moderately priced. Analog systems need to be built with a
complex array of Op Amps, resistors, capacitors, diodes, etc, while a digital system can be implemented with a generic microcontroller, and some quick
programming.

Sampling and Reconstruction

The process of converting from analog data to digital data is called "sampling". The process of recreating an analog signal from a digital one is called
"reconstruction". This book will not talk about either of these subjects in much depth beyond this, although other books on the topic of EE might, such as A-
level Physics (Advancing Physics)/Digitisation.

further reading

Electronics/Digital to Analog & Analog to Digital Converters

Signals need a channel to follow, so that they can move from place to place. These Communication Mediums, or "channels" are things like wires and
antennae that transmit the signal from one location to another. Some of the most common channels are listed below:

Twisted Pair Wire

Twisted Pair is a transmission medium that uses two conductors that are twisted together to form a pair. The concept for the twist of the conductors is to
prevent interference. Ideally, each conductor of the pair basically receives the same amount of interference, positive and negative, effectively cancelling the
effect of the interference. Typically, most inside cabling has four pairs with each pair having a different twist rate. The different twist rates help to further
reduce the chance of crosstalk by making the pairs appear electrically different in reference to each other. If the pairs all had the same twist rate, they would
be electrically identical in reference to each other causing crosstalk, which is also referred to as capacitive coupling. Twisted pair wire is commonly used in
telephone and data cables with variations of categories and twist rates.

Other variants of Twisted Pair are the Shielded Twisted Pair cables. The shielded types operate very similar to the non-shielded variety, except that Shielded
Twisted Pair also has a layer of metal foil or mesh shielding around all the pairs or each individual pair to further shield the pairs from electromagnetic
interference. Shielded twisted pair is typically deployed in situations where the cabling is subjected to higher than normal levels of interference.

Coaxial Cable
Another common type of wire is Coaxial Cable. Coaxial cable (or simply, "coax") is a type of cable with a single data line, surrounded by various layers of
padding and shielding. The most common coax cable, common television cable, has a layer of wire mesh surrounding the padded core, that absorbs a large
amount of EM interference, and helps to ensure a relatively clean signal is transmitted and received. Coax cable has a much higher bandwidth than a twisted
pair, but coax is also significantly more expensive than an equal length of twisted pair wire. Coax cable frequently has an available bandwidth in excess of
hundreds of megahertz (in comparison with the hundreds of kilohertz available on twisted pair wires).

Originally, Coax cable was used as the backbone of the telephone network because a single coaxial cable could hold hundreds of simultaneous phone calls by
a method known as "Frequency Division Multiplexing" (discussed in a later chapter). Recently however, Fiber Optic cables have replaced Coaxial Cable as
the backbone of the telephone network because Fiber Optic channels can hold many more simultaneous phone conversations (thousands at a time), and are
less susceptible to interference, crosstalk, and noise then Coaxial Cable.

Fiber Optics
Fiber Optic cables are thin strands of glass that carry pulses of light (frequently infrared light) across long distances. Fiber Optic channels are usually
immune to common RF interference, and can transmit incredibly high amounts of data very quickly. There are 2 general types of fiber optic cable: single
frequency cable, and multi-frequency cable. single frequency cable carries only a single frequency of laser light, and because of this there is no self-
interference on the line. Single-frequency fiber optic cables can atain incredible bandwidths of many gigahertz. Multi-Frequency fiber optics cables allow a
Frequency-Division Multiplexed series of signals to each inhabit a given frequency range. However, interference between the different signals can decrease
the range over which reliable data can be transmitted.

Wireless Transmission
In wireless transmission systems, signals are propagated as Electro-Magnetic waves through free space. Wireless signals are transmitted by a transmitter,
and received by a receiver. Wireless systems are inexpensive because no wires need to be installed to transmit the signal, but wireless transmissions are
susceptable not only to EM interference, but also to physical interference. A large building in a city, for instance can interfere with cell-phone reception, and a
large mountain could block AM radio transmissions. Also, WiFi internet users may have noticed that their wireless internet signals don't travel through walls
very well.

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There are 2 types of antennas that are used in wireless communications, isotropic, and directional.

Isotropic

People should be familiar with isotropic antennas because they are everywhere: in your car, on your radio, etc... Isotropic antennas are omni-directional in the
sense that they transmit data out equally (or nearly equally) in all directions. These antennas are excellent for systems (such as FM radio transmission) that
need to transmit data to multiple receivers in multiple directions. Also, Isotropic antennas are good for systems in which the direction of the receiver, relative
to the transmitter is not known (such as cellular phone systems).

Directional

Directional antennas focus their transmission power in a single narrow direction range. Some examples of directional antennas are satellite dishes, and wave-
guides. The downfall of the directional antennas is that they need to be pointed directly at the receiver all the time to maintain transmission power. This is
useful when the receiver and the transmitter are not moving (such as in communicating with a geo-synchronous satellite).

Receiver Design
It turns out that if we know what kind of signal to expect, we can better receive those signals. This should be intuitive, because it is hard to find something if
we don't know what precisely we are looking for. How is a receiver supposed to know what is data and what is noise, if it doesnt know what data looks like?

Coherent transmissions are transmissions where the receiver knows what type of data is being sent. Coherency implies a strict timing mechanism, because
even a data signal may look like noise if you look at the wrong part of it. In contrast, noncoherent receivers don't know exactly what they are looking for, and
therefore noncoherent communication systems need to be far more complex (both in terms of hardware and mathematical models) to operate properly.

This section will talk about coherent receivers, first discussing the "Simple Receiver" case, and then going into theory about what the optimal case is. Once
we know mathematically what an optimal receiver should be, we then discuss two actual implementations of the optimal receiver.

It should be noted that the remainder of this book will discuss optimal receivers. After all, why would a communication's engineer use anything that is less
then the best?

The Simple Receiver

A simple receiver is just that: simple. A general simple receiver will consist of a low-pass filter (to remove excess high-frequency noise), and then a sampler,
that will select values at certain points in the wave, and interpolate those values to form a smooth output curve. In place of a sampler (for purely analog
systems), a general envelope filter can also be used, especially in AM systems. In other systems, different tricks can be used to demodulate an input signal,
and acquire the data. However simple receivers, while cheap, are not the best choice for a receiver. Occcasionally they are employed because of their price,
but where performance is an issue, a better alternative receiver should be used.

The Optimal Receiver

Mathematically, Engineers were able to predict the structure of the optimal receiver. Read that sentence again: Engineers are able to design, analyze, and
build the best possible receiver, for any given signal. This is an important development for several reasons. First, it means that no more research should go
into finding a better receiver. The best receiver has already been found, after all. Second, it means any communications system will not be hampered (much)
by the receiver.

Derivation

here we will attempt to show how the coherent receiver is derived.

Matched Receiver

The matched receiver is the logical conclusion of the optimal receiver calculation. The matched receiver convolutes the signal with itself, and then tests the
output. Here is a diagram:

s(t)----->(Convolve with r(t))----->

This looks simple enough, except that convolution modules are often expensive. An alternative to this approach is to use a correlation receiver.

Correlation Receiver

The correlation receiver is similar to the matched receiver, instead with a simple switch: The multiplication happens first, and the integration happens second.

Here is a general diagram:

r(t)
|
v
s(t) ----->(X)----->(Integrator)--->

In a digital system, the integrator would then be followed by a threshold detector, while in an analog receiver, it might be followed by another detector, like
an envelope detector.

Conclusion
To do the best job of receiving a signal, we need to know the form of the signal that we are sending. This should seem obvious, we can't design a receiver
until after we've decided how the signal will be sent. This method poses some problems however, in that the receiver must be able to line up the received

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signal with the given reference signal to work the magic: If the received signal and the reference signal are out of sync with each other, either as a function of
an error in phase or an error in frequency, then the optimal receiver will not work.

further reading

Section 2: Analog Modulation

Analog Modulation Overview

Let's take a look at a generalized sinewave:

It consists of three components namely; amplitude, frequency and phase. Each of which can be decomposed to provide finer detail:

x(t) = As(t)sin(2π[fc + kfm(t)]t + αφ(t))

Types of Analog Modulation

We can see 3 parameters that can be changed in this sine wave to send information:

As(t). This term is called the "Amplitude", and changing it is called "Amplitude Modulation" (AM)

kfm(t) This term is called the "Frequency Shift", and changing it is called "Frequency Modulation"

αφ(t). this term is called the "Phase angle", and changing it is called "Phase Modulation".

The terms frequency and phase modulation are often combined into a more general group called "Angle Modulation".

The Breakdown
Each term consists of a coefficient (called a "scaling factor"), and a function of time that corresponds to the information that we want to send. The scaling
factor out front, A, is also used as the transmission power coefficient. When a radio station wants their signal to be stronger (regardless of whether it is AM,
FM, or PM), they "crank-up" the power of A, and send more power out onto the airwaves.

How we Will Cover the Material

We are going to go into separate chapters for each different type of modulation. This book will attempt to discuss some of the mathematical models and
techniques used with different modulation techniques. It will also discuss some practical information about how to construct a transmitter/receiver, and how
to use each modulation technique effectively.

This page will talk about Amplitude modulation.

Amplitude Modulation

Amplitude modulation (AM) occurs when the amplitude of a carrier wave is modulated, to correspond to a source signal. In AM, we have an equation that
looks like this:

Fsignal(t) = A(t)sin(ωt)

We can also see that the phase of this wave is irrelevant, and does not change (so we dont even include it in the equation).

AM Radio uses AM modulation

AM Double-Sideband (AM-DSB for short) can be broken into two different, distinct types: Carrier, and Suppressed Carrier varieties (AM-DSB-C and AM-
DSB-SC, for short, respectively). This page will talk about both varieties, and will discuss the similarities and differences of each.

Characteristics

Modulation Index

Amplitude modulation requires a high frequency constant carrier and a low frequency modulation signal.

A sine wave carrier is of the form

A sine wave modulation signal is of the form

Notice that the amplitude of the high frequency carrier takes on the shape of the lower frequency modulation signal, forming what is called a modulation
envelope.

The modulation index is defined as the ratio of the modulation signal amplitude to the carrier amplitude.

where

The overall signal can be described by:

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More commonly, the carrier amplitude is normalized to one and the am equation is written as:

In most literature this expression is simply written as:

If the modulation index is zero (mam = 0) the signal is simply a constant amplitude carrier.

If the modulation index is 1 (mam = 1), the resultant waveform has maximum or 100% amplitude modulation.

Sidebands

Expanding the normalized AM equation:

we obtain:

where:

sinωct represents the carrier

represents the lower sideband

represents the upper sideband

The sidebands are centered on the carrier frequency. The are the sum and difference frequencies of the carrier and modulation signals. In the above example,
they are just single frequencies, but normally the baseband modulation signal is a range of frequencies and hence two bands are formed.

As a side point, note that multiplication in the time domain causes addition and subtraction in the frequency domain.

AM Modulator

The standard amplitude modulation equation is:

From this we notice that AM involves a multiplication process. (Recall that multiplication in the time domain is equal to addition and subtraction in the
frequency domain). There are several ways to perform this function electronically. The simplest method uses a switch.

Switching Modulators

Switching modulators can be placed into two categories: unipolar and bipolar.

Bipolar Switching Modulator

The bipolar switch is the easiest to visualize. Note that an AM waveform appears to consist of a low frequency dc signal whose polarity is reversing at a
carrier rate.

The AM signal can be created by multiplying a dc modulation signal by ±1.

However, since the square wave contains lots of harmonics, the resulting
multiplication will contain lots of extraneous frequencies. Mathematically,
the spectrum of the square wave signal (given by the Fourier Transform) is
of the form:

This seems complicated but, if the square wave switching function has a 50% duty cycle, this simplifies to:

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This tells us that the square wave is actually composed of a series of cosines (phase shifted sines) at odd multiples of the fundamental switching
frequency. Therefore, using this signal to multiply the baseband signal results in AM signals being generated at each of the odd harmonics of the
switching (carrier) frequencies. Since the amplitude of the harmonics decreases rapidly, this technique is practical for only the first few harmonics, and
produces an enormous amount of unwanted signals (noise).

A band pass filter can be used to select any one of the AM signals. The number of different output frequencies
can be significantly reduced if the multiplier accepts sinewaves at the carrier input.

Removing the DC component from the input eliminates the carrier signal and creates DSBSC modulation.

Physically this is done by reversing the signal leads:

The process of reversing the polarity of a signal is easily accomplished by placing two switch pairs in the
output of a differential amplifier. The Mc1496 Balanced Modulator is an example of such a device.

Unipolar Switching Modulator

As previously mentioned, an AM signal can be created by multiplying a dc

modulation signal by 0 & 1.

The spectrum of this signal is defined by:

Physically this is done by turning the modulation signal on and off at the carrier rate:

A high amplitude carrier can be used to turn a diode on and off. A dc bias is placed on the modulation signal to
make certain that only the carrier (not the modulation signal) can reverse bias the diode.

It may not seem obvious, but the output of

this circuit contains a series of AM
signals. A bandpass filter is needed to
extract the desired one. Normally it is the
1st or 3rd harmonic of the fundamental.
(The 1st harmonic is the fundamental.)

Collector Modulator

The diode switching modulator is incapable of producing high power signals

since it is a passive device. A transistor can be used to overcome this limitation.

Square Law Modulator

The voltage-current relationship of a diode is nonlinear near the knee and is of the form:

The coefficient a and b are constants associated with the particular diode.

Amplitude modulation occurs if the diode is kept in the square law region
when signals combine.

Let the injected signals be of the form:

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The voltage applied across the diode and resistor is given by:

The current in the diode and hence in the resistor is given by:

Which expands to:

Modulation Index Measurement

It is sometimes difficult to determine the modulation index, particularly for complex signals. However, it is relatively easy to determine it by observation.

The trapezoidal oscilloscope display can be used to determine the modulation index.

AM modulation index:

The trapezoidal display makes it possible to quickly recognize certain types

of problems, which would reduce the AM signal quality.

The highest authorized carrier power for

AM broadcast in the US is 50 kilowatts,
although directional stations are permitted
52.65 kilowatts to compensate for losses
in the phasing system. The ERP can be much higher

C-QUAM

The basic idea behind the C-Quam modulator is actually quite simple. The output stage is an ordinary AM modulator however; the carrier signal has been
replaced by an amplitude limited vector modulator. Therefore, the limiter output is really a phase-modulated signal.

A standard AM receiver will detect the amplitude variations as L+R. A stereo receiver will also detect the
phase variations and to extract L-R. It will then process these signals to separate the left and right channels.

To enable the stereo decoder, a 25 Hz pilot tone is added to the L-R channel.

AM Receivers

The most common receivers in use today are the super heterodyne type. They consist of:

Antenna

RF amplifier

Local Oscillator and Mixer

IF Section

Detector and Amplifier

The need for these subsystems can be seen when one considers the much simpler and inadequate TRF or tuned radio frequency amplifier.

TRF Amplifier

It is possible to design an RF amplifier to accept only a narrow range of frequencies, such as one radio station on the AM band.

By adjusting the center frequency of the tuned circuit, all other input signals can be excluded.

The AM band ranges from about 500 KHz to 1600 kHz. Each station
requires 10 kHz of this spectrum, although the baseband signal is only 5
KHz.

Recall that for a tuned circuit: . The center or resonant frequency in

an RLC network is most often adjusted by varying the capacitor value.

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However, the Q remains approximately constant as the center frequency is adjusted. This suggests that as the bandwidth varies as the circuit is tuned.

For example, the Q required at the lower end of the AM band to select only one radio station would be approximately:

As the tuned circuit is adjusted to the higher end of the AM band, the resulting bandwidth is:

A bandwidth this high could conceivably pass three adjacent stations, thus making meaningful reception impossible.

To prevent this, the incoming RF signal is heterodyned to a fixed IF or intermediate frequency and passed through a constant bandwidth circuit.

Superheterodyne Receiver

The RF amplifier boosts the RF signal into the mixer. It has broad tuning and amplifies not just one RF station,
but many of them simultaneously. It also amplifies any input noise and even contributes some of its own.

The other mixer input is a high frequency sine wave created by a local oscillator. In AM receivers, it is always
455 kHz above the desired station carrier frequency. An ideal mixer will combine the incoming carrier with the
local oscillator to create sum and difference frequencies. .

A real mixer combines two signals and creates a host of new frequencies:

• A dc level
• The original two frequencies
• The sum and difference of the two input frequencies
• Harmonics of the two input frequencies
• Sums and differences of all of the harmonics

Since the RF amplifier passes several radio stations at once, the mixer output can be very complex. However, the only signal of real interest is the difference
between the desired station carrier frequency and the local oscillator frequency. This difference frequency, also called the IF (intermediate frequency) will
alway be 455 kHz. By passing this through a 10 kHz BPF (band pass filter) centered at 455 kHz, the bulk of the unwanted signals can be eliminated.

Local Oscillator Frequency

Since the mixer generates sum and difference frequencies, it is possible to generate the 455 KHz IF signal if the local oscillator is either above or below the
IF. The inevitable question is which is preferable.

Case I The local Oscillator is above the IF. This would require that the oscillator tune from (500 + 455) kHz to (1600 + 455) kHz or
approximately 1 to 2 MHz. It is normally the capacitor in a tuned RLC circuit, which is varied to adjust the center frequency while the inductor is
left fixed.

Since

solving for C we obtain

When the tuning frequency is a maximum, the tuning capacitor is a minimum and vice versa. Since we know the range of frequencies to be
created, we can deduce the range of capacitance required.

Making a capacitor with a 4:1 value change is well within the realm of possibility.

Case II The local Oscillator is below the IF. This would require that the oscillator tune from (500 - 455) kHz to (1600 - 455) kHz or
approximately 45 kHz to 1145 kHz, in which case:

It is not practical to make a tunable capacitor with this type of range. Therefore the local oscillator in a standard AM receiver is above the radio band.

Image Frequency

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Just as there are two oscillator frequencies, which can create the same IF, two different station frequencies can create the IF. The undesired station frequency
is known as the image frequency.

If any circuit in the radio front end exhibits non-linearities, there is a possibility that other combinations may
create the intermediate frequency. Once the image frequency is in the mixer, there is no way to remove it since
it is now heterodyned into the same IF band as the desired station.

AM Detection

There are two basic types of AM detection, coherent and non-coherent. Of these two, the non-coherent is the
simpler method.

Non-coherent detection does not rely on regenerating the carrier signal. The information or modulation envelope can be removed or
detected by a diode followed by an audio filter.

Coherent detection relies on regenerating the carrier and mixing it with the AM signal. This creates sum and difference frequencies. The
difference frequency corresponds to the original modulation signal.

Both of these detection techniques have certain drawbacks. Consequently, most radio receivers use a combination of both.

Envelope Detector

An envelope detector is simply a half wave rectifier followed by a low pass filter. In the case of commercial AM
radio receivers, the detector is placed after the IF section. The carrier at this point is 455 kHz while the maximum
envelope frequency is only 5 kHz. Since the ripple component is nearly 100 times the frequency of the highest
baseband signal and does not pass through any subsequent audio amplifiers.

An AM signal where the carrier frequency is only 10 times the envelope frequency would have considerable
ripple:

Synchronous Detector

In a synchronous or coherent detector, the incoming AM signal is mixed with the original carrier frequency.

If you think this looks suspiciously like a mixer, you are absolutely right! A
synchronous detector is one where the difference frequency between the two
inputs is zero Hz. Of in other words, the two input frequencies are the same.
Let's check the math.

Recall that the AM input is mathematically defined by:

At the multiplier output, we obtain:

The high frequency component can be filtered off leaving only the original modulation signal.

This technique has one serious drawback. The problem is how to create the exact carrier frequency. If the frequency is not exact, the entire baseband signal
will be shifted by the difference. A shift of only 50 Hz will make the human voice unrecognizable. It is possible to use a PLL (phase locked loop), but making
one tunable for the entire AM band is not trivial.

As a result, most radio receivers use an oscillator to create a fixed intermediate frequency. This is then followed by an envelope detector or a fixed frequency
PLL.

Squaring Detector

The squaring detector is also a synchronous or coherent detector. It avoids the problem of having to recreate the carrier by simply squaring the input signal. It
essentially uses the AM signal itself as a sort of wideband carrier.

The output of the multiplier is the square of the input AM signal:

Since the input is being multiplied by the sinωct component, one of the resulting difference terms is the original modulation signal. The principle difficulty
with this approach is trying to create a linear, high frequency multiplier.

AM-DSBSC
AM-DSB-SC is characterized by the following transmission equation:

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v(t) = As(t)cos(2πfct)

It is important to notice that s(t) can contain a negative value. AM-DSB-SC requires a coherent receiver, because the modulation data can go negative, and
therefore the receiver needs to know that the signal is negative (and not just phase shifted). AM-DSB-SC systems are very susceptable to frequency shifting
and phase shifting on the receiving end. In this equation, A is the transmission amplitude.

Double side band suppressed carrier modulation is simply AM without the broadcast carrier. Recall that the AM signal is defined by:

The carrier term in the spectrum can be eliminated by removing the dc offset from the modulating signal:

Double Balanced Ring Modulator

One of the circuits which is capable of creating DSBSC is the double balance ring modulator.

If the carrier is large enough to cause the diodes to switch states, then the circuit acts like a diode switching
modulator:

The modulation signal is inverted at the carrier rate. This is essentially

multiplication by ±1. Since the transformers cannot pass dc, there is no term
which when multiplied can create an output carrier. Since the diodes will
switch equally well on either cycle, the modulation signal is effectively
being multiplied by a 50% duty cycle square wave creating numerous
DSBSC signals, each centered at an odd multiple of the carrier frequency.
Bandpass filters are used to extract the frequency of interest.

Some IC balanced modulators use this technique, but use transistors instead of diodes to perform the switching.

Push Pull Square Law Balanced Modulator

This circuit uses the same principles as the diode square law modulator. Since dc cannot pass through the
transformer, it would be expected that there would be no output signal at the carrier frequency.

The drain current vs. gate-source voltage is of the form:

The net drain current in the output transformer is given by:

By applying KVL around the gate loops we obtain:

Putting it all together we obtain:

inet = aem + 2becem

From this we note that the first term is the originating modulation signal and can easily be filtered off by a high pass filter. The second term is of the
form:

which is AM DSBSC.

AM-DSB-C

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In contrast to AM-DSB-SC is AM-DSB-C, which is categorized by the following equation:

v(t) = A[s(t) + c]cos(2πfct)

Where c is a positive term representing the carrier. If the term [s(t) + c] is always non-negative, we can receive the AM-DSB-C signal non-coherently, using a
simple envelope detector to remove the cosine term. The +c term is simply a constant DC signal and can be removed by using a blocking capacitor.

It is important to note that in AM-DSB-C systems, a large amount of power is wasted in the transmission sending a "boosted" carrier frequency. since the
carrier contains no information, it is considered to be wasted energy. The advantage to this method is that it greatly simplifies the receiver design, since there
is no need to generate a coherent carrier signal at the receiver. For this reason, this is the transmission method used in conventional AM radio.

AM-DSB-SC and AM-DSB-C both suffer in terms of bandwidth from the fact that they both send two identical (but reversed) frequency "lobes", or bands.
These bands (the upper band and the lower band) are exactly mirror images of each other, and therefore contain identical information. Why can't we just cut
one of them out, and save some bandwidth? The answer is that we can cut out one of the bands, but it isn't always a good idea. The technique of cutting out
one of the sidebands is called Amplitude Modulation Single-Side-Band (AM-SSB). AM-SSB has a number of problems, but also some good aspects. A
compromise between AM-SSB and the two AM-DSB methods is called Amplitude Modulation Vestigial-Side-Band (AM-VSB), which uses less
bandwidth then the AM-DSB methods, but more than the AM-SSB.

Transmitter

A typical AM-DSB-C transmitter looks like this:

c cos(...)
| |
Signal ---->(+)---->(X)----> AM-DSB-C

which is a little more complicated than an AM-DSB-SC transmitter.

Receiver

An AM-DSB-C receiver is very simple:

AM-DSB-C ---->|Envelope Filter|---->|Capacitor|----> Signal

The capacitor blocks the DC component, and effectively removes the +c term.

AM-SSB
To send an AM-SSB signal, we need to remove one of the sidebands from an AM-DSB signal. This means that we need to pass the AM-DSB signal through
a filter, to remove one of the sidebands. The filter, however, needs to be a very high order filter, because we need to have a very agressive roll-off. One
sideband needs to pass the filter almost completely unchanged, and the other sideband needs to be stopped completely at the filter.

To demodulate an AM-SSB signal, we need to perform the following steps:

1. Low-pass filter, to remove noise

2. Modulate the signal again by the carrier frequency
3. Pass through another filter, to remove high-frequency components
4. Amplify the signal, because the previous steps have attenuated it significantly.

AM-SSB is most efficient in terms of bandwidth, but there is a significant added cost involved in terms of more complicated hardware to send and receive
this signal. For this reason, AM-SSB is rarely seen as being cost effective.

Single sideband is a form of AM with the carrier and one sideband removed. In normal AM broadcast, the transmitter is rated in terms of the carrier power.
SSB transmitters attempt to eliminate the carrier and one of the sidebands. Therefore, transmitters are rated in PEP (peak envelope power).

With normal voice signals, an SSB transmitter outputs 1/4 to 1/3 PEP.

There a numerous variations of SSB:

SSB - Single sideband - amateur radio

SSSC - Single sideband suppressed carrier - a small pilot carrier is transmitted

ISB - Independent sideband - two separate sidebands with a suppressed carrier. Used in radio telephony.

VSB - Vestigial sideband - a partial sideband. Used in broadcast TV.

ACSSB - Amplitude companded SSB

There are several advantages of using SSB:

• More efficient spectrum utilization

• Less subject to selective fading
• More power can be placed in the intelligence signal
• 10 to 12 dB noise reduction due to bandwidth limiting

Filter Method

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The simplest way to create SSB is to generate DSBSC and then use a bandpass filter to extract one of the sidebands.

This technique can be used at relatively low carrier frequencies. At high frequencies, the Q of the filter
becomes unacceptably high. The required Q necessary to filter off one of the sidebands can be approximated
by:

where:

Several types of filters are used to suppress unwanted sidebands:

LC - Maximum Q = 200

Ceramic - Maximum Q = 2000

Mechanical - Maximum Q = 10,000

Crystal - Maximum Q = 50,000

In order to reduce the demands placed upon the filter, a double heterodyne technique can be used.

The first local oscillator has a relatively low frequency thus enabling the removal of one of the sidebands
produced by the first mixer. The signal is then heterodyned a second time, creating another pair of sidebands.
However, this time they are separated by a sufficiently large gap that one can be removed by the band limited
power amplifier or antenna matching network.

Example

Observe the spectral distribution under the following conditions:

• Audio baseband = 100 HZ to 5 KHz

• LO1 = 100 kHz
• LO2 = 50 MHz

The spectral output of the first mixer is:

If the desired sideband suppression is 80 dB, the Q required to filter off one of the sidebands is approximately:

It is evident that a crystal filter would be needed to remove the unwanted sideband.

After the filter, only one sideband is left. In this example, we’ll retain the USB. The spectrum after the second mixer is:

The Q required to suppress one of the side bands by 80 dB is approximately:

Thus, we note that the required Q drops in half.

This SSB filter technique is used in radiotelephone applications.

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Phase Shift Method

The output from the top mixer is given by:

The output from the bottom mixer is given by:

The output of the sumer is:

which corresponds to the lower sideband.

The major difficulty with this technique is the need to provide a constant 90o phase shift over the entire input audio band. To overcome this obstacle, the
Weaver or third method uses an audio sub carrier, which is phase shifted.

Weaver Method

The Weaver or ‘third’ method places the baseband signal on a low frequency quadrature carrier.

This has the advantage of not requiring a broadband phase shifter however; the use of four mixers makes it
awkward and seldom used.

SSB Transmitter

AM-SSB transmitters are a little more complicated:

cos(...)
|
Signal ---->(X)---->|Low-Pass Filter|----> AM-SSB

The filter must be a very high order, for reasons explained in that chapter.

SSB Receiver

An AM-SSB receiver is a little bit complicated as well:

cos(...)
|
AM-SSB ---->(X)---->|Low-Pass Filter|---->|Amplifier|----> Signal

This filter doesnt need to be a very high order, like the transmitter has.

These receivers require extremely stable oscillators, good adjacent channel selectivity, and typically use a double conversion technique. Envelope detectors
cannot be used since the envelope varies at twice the frequency of the AM envelope.

Stable oscillators are needed since the detected signal is proportional to the difference between the untransmitted carrier and the instantaneous side band. A
small shift of 50 Hz makes the received signal unusable.

SSB receivers typically use fixed frequency tuning rather than continuous tuning as found on most radios. Crystal oscillators are often used to select the fixed
frequency channels.

AM-VSB
As a compromise between AM-SSB and AM-DSB is AM-VSB. To make an AM-VSB signal, we pass an AM-DSB signal through a lowpass filter. Now, the
trick is that we pass it through a low-order filter, so that some of the filtered sideband still exists. This filtered part of the sideband is called the "Vestige" of
the sideband, hence the name "Vestigial Side Band".

AM-VSB signals then can get demodulated in a similar manner to AM-SSB. We can see when we remodulate the input signal, the two vestiges (the positive
and negative mirrors of each other) over-lap each other, and add up to the original, unfiltered value!

AM-VSB is less expensive to implement then AM-SSB because we can use lower-order filters.

Broadcast television in North America uses AM-VSB

Transmitter

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here we will talk about an AM-VSB transmitter circuit.

Receiver

here we will talk about an AM-VSB receiver circuit.

AM Demodulation
When trying to demodulate an AM signal, it seems like good sense that only the amplitude of the signal needs to be examined. By only examining the
amplitude of the signal at any given time, we can remove the carrier signal from our considerations, and we can examine the original signal. Luckily, we have
a tool in our toolbox that we can use to examine the amplitude of a signal: The Envelope Detector. (AM)

Frequency Modulation

If we make the frequency of our carrier wave a function of time, we can get a generalized function that looks like this:

sFM = Acos(2π[fc + ks(t)]t + φ)

We still have a carrier wave, but now we have the value ks(t) that we add to that carrier wave, to send our data.

As an important result, ks(t) must be less then the carrier frequency always, to avoid ambiguity and distortion.

Deriving the FM Equation

Recall that a general sinusoid is of the form:

Frequency modulation involves deviating a carrier frequency by some amount. If a sine wave is used to deviate the carrier, the expression for the frequency at
any instant would be:

where:

ωi = instantaneous frequency
ωc = carrier frequency
∆ω = carrier deviation
ωm = modulation frequency

This expression describes a signal varying sinusoidally about some average frequency. However, we cannot simply substitute this expression into the general
equation for a sinusoid to get the FM equation. This is because the sine operator acts on angles, not frequency. Therefore, we must define the instantaneous
frequency in terms of angles.

It should be noted that the modulation signal amplitude governs the amount of carrier deviation while the modulation signal frequency governs the rate of
carrier deviation.

The term ω is an angular velocity (radians per second) and is related to frequency and angle by the following relationship:

To find the angle, we must integrate ω with respect to time:

We can now find the instantaneous angle associated with the instantaneous frequency:

This angle can now be substituted into the general carrier signal to define FM:

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The FM modulation index is defined as the ratio of the carrier deviation to modulation frequency:

Consequently, the FM equation is often written as:

Bessel's Functions

This is a very complex expression and it is not readily apparent what the sidebands of this signal are like. The solution to this problem requires a knowledge
of Bessel's functions of the first kind and order p. In open form, it resembles:

where:

Magnitude of the frequency component

p = Side frequency number (not to be confused with sidebands)
x = Modulation index

As a point of interest, Bessel's functions are a solution to the following equation:

Bessel's functions occur in the theory of cylindrical and spherical waves, much like sine waves occur in the theory of plane waves.

It turns out that FM generates an infinite number of side frequencies (in both the upper and lower sidebands). Each side frequency is an integer multiple of the
modulation signal frequency. The amplitude of higher order side frequencies decreases rapidly and can generally be ignored.

The amplitude of the carrier signal is also a function of the modulation index and under some conditions, its amplitude can actually go to zero. This does not
mean that the signal disappears, but rather that all of the broadcast energy is redistributed to the side frequencies.

A plot of the carrier and first five side frequency amplitudes as a function of modulation index resembles:

The Bessel coefficients have several interesting properties including:

One very useful interpretation of this is: J0 represents the voltage amplitude of the carrier, J1 represents the
amplitude of the 1st side frequency, J2 the 2nd side frequency etc. Note that the sum of the squares (power)
remains constant.

FM Bandwidth

FM generates upper and lower sidebands, each of which contain an infinite number of side frequencies. However, the FM bandwidth is not infinite because
the amplitude of the higher order side frequencies decreases rapidly. Carson's Rule is often used to calculate the bandwidth, since it contains more than 90%
of the FM signal.

Carson's Rule

In commercial broadcast applications, the maximum modulation index (mfm) = 5, the maximum, carrier deviation (∆f) = 75 kHz, and maximum modulation
frequency (fm) = 15 kHz. The total broadcast spectrum according to Carson's rule is 180 kHz, but an additional 20 kHz guard band is used to separate adjacent
radio stations. Therefore, each FM radio station is allocated 200 kHz.

Noise

In AM systems, noise easily distorts the transmitted signal however, in FM systems any added noise must create a frequency deviation in order to be
perceptible.

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The maximum frequency deviation due to random noise occurs when the noise is at right angles to the resultant signal. In the worst case the signal frequency
has been deviated by:

δ = θfm

This shows that the deviation due to noise increases as the modulation frequency increases. Since noise power is the square of the noise voltage, the signal to
noise ratio can significantly degrade.

To prevent this, the amplitude of the modulation signal is increased to keep the S/N ratio constant over the entire broadcast band. This is called pre-emphasis.

Pre & De-emphasis

Increasing the amplitude of high frequency baseband signals in the FM modulator (transmitter) must be compensated for in the FM demodulator (receiver)
otherwise the signal would sound quite tinny (too much treble).

The standard curves resemble:

In commercial FM broadcast, the emphasis circuits consist of a simple RC network with a time constant of 75
µSec and a corner frequency of 2125 Hz.

The magnitude of the pre-emphasis response is defined by:

FM Transmission Power
Since the value of the amplitude of the sine wave in FM does not change, the transmitted power is a constant. As a
general rule, for a sinusoid with a constant amplitude, the transmitted power can be found as follows: The equation for the transmitted
power in a sinusoid is a fundamental
equation. Remember it.

Where A is the amplitude of the sine wave, and RL is the resistance of the load. In a normalized system, we set RL to 1.

The Bessel coefficients can be used to determine the power in the carrier and any side frequency:

PC is the power in the unmodulated carrier.

PT is the total power and is by definition equal to the unmodulated carrier power.

As the modulation index varies, the individual Bessel coefficients change and power is redistributed from the carrier to the side frequencies.

FM Transmitters
FM Transmitters can be easily implemented using a VCO (see why we discussed Voltage Controlled Oscillators, in the first section?), because a VCO
converts an input voltage (our input signal) to a frequency (our modulated output).

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Signal ----->|VCO|-----> FM Signal

FM Receivers
Any angle modulation receiver needs to have several components:

1. A limiter, to remove abnormal amplitude values

2. bandpass filter, to separate the out-of-band noise.
3. A Discriminator, to change a frequency back to a voltage
4. A lowpass filter, to remove noise added by the discriminator.

A discriminator is essentially a differentiator in line with an envelope detector:

FM ---->|Differentiator|---->|Envelope Filter|----> Signal

Also, you can add in a blocking capacitor to remove any DC components of the signal, if needed. (FM)

Phase Modulation
Phase modulation is most commonly used to convey digital signals. All high performance modems today use phase modulation.

Similar to FM (frequency modulation), is Phase modulation. (We will show how they are the same in the next chapter.) If we alter the value of the phase
according to a particular function, we will get the following generalized PM function:

sPM = Acos(2πfct + αs(t))

It is important to note that the fact that for all values of t. If this relationship is not satisfied, then the phase angle is said to be wrapped.

BPSK Modulator

The binary phase shift keyed modulator is the simplest of app PSK modulators since it has only two output phase states. It is generally a multiplier which can
either be an IC (integrated circuit) or ring modulator.

The output has two phase states:

In the above illustration, the duration of each of the phase states corresponds to one signaling element or baud. The baud rate is therefor equal to the bit rate.

The spectrum of the BPSK signal will depend upon the data being transmitted, but it is very easy to sketch it for the highest data rate input.

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The resultant BPSK spectrum is:

QPSK Modulators (4-PSK)

Quadrature modulation uses two data channels denoted I (in phase) and Q (quadrature phase) displaced by 90o with respect to each other. It may seem
somewhat paradoxical, that although these two channels are combined prior to transmission, they do not interfere with each other.

The receiver is quite capable of separating them because of their quadrature or orthogonal nature.

In the most basic configuration, there are 4 possible output phases. This suggests that each output symbol correspond to 2 bits of binary information. Since
several bits can be encoded into a baud, the bit rate exceeds the baud rate.

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The first thing that happens in this circuit is that the incoming bits are organized into groups of 2 called dibits. They are separated into 2 data streams and kept
constant over the dibit period.

Each data stream is fed to a BPSK modulator. However, orthogonal carriers feed the two modulators. The output of the I channel modulator resembles:

The output of the Q channel modulator resembles

Combining the I and Q channels has the effect of rotating the output state by 45o.

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Rotating the output reference to 45o for the sake of clarity, the transmitted output for this particular data sequence is therefor:

8-PSK

This process of encoding more bits into each output baud or phase state can be continued. Organizing binary bits into 3 bytes corresponds to 8 different
conditions.

The output constellation diagram for the 8 different phase states is:

From this diagram it is readily apparent that two different amplitudes are needed on the I and Q channels. If the A bit is used to control the polarity of the I
channel and the B bit the polarity of the Q channel, then the C bit can be used to define the two different amplitudes. In order to evenly space the phase states;
the amplitudes must be ± 0.38 and ± 0.92. The magnitude of the I and Q channel signals must always be different. An inverter can be used to assure this
condition.

The input bit stream is organized into 3 bit bytes. Each bit is sent to a different location to control a certain aspect of the modulator. The inputs to the 2 - 4
level converter are 0’s or 1’s but the output is ± 0.38 or ± 0.92, depending on the C bit.

Wrapped/Unwrapped Phase
The phase angle is a circular quantity, with the restriction 0 = 2π. Therefore, if we wrap the phase a complete 360 degrees around, the receiver will not know
the difference, and the transmission will fail. When the phase exceeds 360 degrees, the phase value is said to be wrapped. It is highly difficult to construct a
communication system that can detect and decode a wrapped phase value.

PM Transmitter

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PM signals can be transmitted using a technique very similar to FM transmitters. The only difference is that we need to add a differentiator to it:

Signal ---->|Differentiator|---->|VCO|----> PM Signal

PM Receiver
PM receivers have all the same parts as an FM receiver, except for the 3rd step:

1. A limiter, to remove abnormal amplitude values

2. bandpass filter, to separate the out-of-band noise.
3. A Phase detector, to convert a phase back into a voltage
4. A lowpass filter, to remove noise added by the discriminator.

Phase detectors can be created using a Phase-Locked-Loop (again, see why we discussed them first?). (PM)

Concept
We can see from our initial overviews that FM and PM modulation schemes have a lot in common. Both of them are altering the angle of the carrier sinusoid
according to some function. It turns out that we can go so far as to generalize the two together into a single modulation scheme known as angle modulation.
Note that we will never abbreviate "angle modulation" with the letters "AM", because Amplitude modulation is completely different from angle modulation.

Instantaneous Phase
Let us now look at some things that FM and PM have of common:

sFM = Acos(2π[fc + ks(t)]t + φ)

sPM = Acos(2πfct + αs(t))

What we want to analyze is the argument of the sinusoid, and we will call it Psi. Let us show the Psi for the bare carrier, the FM case, and the PM case:

Ψcarrier(t) = 2πfct + φ

ΨFM(t) = 2π[fc + ks(t)]t + φ

ΨPM(t) = 2πfct + αs(t)

s(t) = Acos(Ψ(t))

This Psi value is called the Instantaneous phase of the sinusoid.

Instantaneous Frequency
Using the Instantaneous phase value, we can find the Instantaneous frequency of the wave with the following formula:

We can also express the instantaneous phase in terms of the instantaneous frequency:

Where the greek letter "lambda" is simply a dummy variable used for integration. Using these relationships, we can begin to study FM and PM signals
further.

Determining FM or PM
If we are given the equation for the instantaneous phase of a particular angle modulated transmission, is it possible to determine if the transmission is using
FM or PM? it turns out that it is possible to determine which is which, by following 2 simple rules:

1. In PM, instantaneous phase is a linear function.

2. In FM, instantaneous frequency minus carrier frequency is a linear function.

For a refresher course on Linearity, there is a chapter on the subject in the Signals and Systems book worth re-reading.

FM radio uses generalized "Angle Modulation"

Bandwidth
In a PM system, we know that the value αs(t) can never go outside the bounds of ( − π,π]. Since sinusoidal functions oscillate between [-1, 1], we can use
them as a general PM generating function. Now, we can combine FM and PM signals into a general equation, called angle modulation:

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v(t) = Asin(2πfct + βsin(2πfmt))

If we want to analyze the spectral components of this equation, we will need to take the Fourier transform of this. But, we can't integrate a sinusoid of a
sinusoid, much less find the transform of it. So, what do we do?

It turns out (and the derivation will be omitted here, for now) that we can express this equation as an infinite sum, as such:

But, what is the term Jn(β)? J is the Bessel function, which is a function that exists only as an open integral (it is impossible to write it in closed form).
Fortunately for us, there are extensive tables tabulating Bessle function values.

The Bessel Function

The definition of the Bessel function is the following equation:

The bessel function is a function of 2 variables, N and β.

Bessel Functions have the following properties:

If n is even:

J − n(β) = Jn(β)

If n is odd:

J − n(β) = − Jn(β)

.

The bessel function is a relatively advanced mathematical tool, and we will not analyze it further in this book.

Carson's Rule
If we have our generalized function:

v(t) = Asin(2πfct + βsin(2πfmt))

We can find the bandwidth BW of the signal using the following formula:

BW = 2(β + 1)fm = 2(∆f + fm)

where ∆f is the maximum frequency deviation, of the transmitted signal, from the carrier frequency. It is important to note that Carson's rule is only an
approximation (albeit one that is used in industry frequently).

what?

Demodulation: First Step

Now, it is important to note that FM and PM signals both do the same first 2 steps during demodulation:

1. Pass the signal through a limiter, to remove amplitude peaks

2. Pass the signal through a bandpass filter to remove low and high frequency noise (as much as possible, without filtering out the signal).

Once we perform these two steps, we no longer have white noise, because we've passed the noise through a filter. Now, we say the noise is colored.

here is a basic diagram of our demodulator, so far:

channel
s(t) ---------> r(t) --->|Limiter|--->|Bandpass Filter|---->z(t)

Where z(t) is the output of the bandpass filter.

Filtered Noise
To denote the new, filtered noise, and new filtered signal, we have the following equation:

z(t) = γAcos(Ψ(t)) + n0(t)

Where we call the additive noise n0(t) because it has been filtered, and is not white noise anymore. n0(t) is known as narrow band noise, and can be denoted
as such:

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Now, once we have it in this form, we can use a trigonometric identity to make this equation more simple:

Where

Here, the new noise parameter R(t) is a rayleigh random variable, and is discussed in the next chapter.

Noise Analysis
R(t) is a noise function that affects the amplitude of our received signal. However, our receiver passes the signal through a limiter, which will remove
amplitude fluctuations from our signal. For this reason, R(t) doesnt affect our signal, and can be safely ignored for now. This means that the only random
variable that is affecting our signal is the variable , "Theta". Theta is a uniform random variable, with values between pi and -pi. Values outside this
range "Wrap around" because phase is circular.

Section 3: Transmission

This page will discuss some of the fundamental basics of EM wave propagation.

Electromagnetic Spectrum

Radio Waves

Maxwell first predicted the existence of electromagnetic waves in the 19th century. He came to this conclusion by careful examination of the equations
describing electric and magnetic phenomenon. It was left up to Hertz to create these waves, and Marconi to exploit them.

In spite of one hundred years of study, exactly what radio waves are and why they exist, remain somewhat of a mystery.

Electromagnetic waves in free space, or TEM waves, consist of electric and magnetic fields, each at right angles to each other and the direction of
propagation.

The relationship between wavelength and frequency is give by:

c = λf

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where c is the speed of light (approximately 300,000,000 m/s in vacuum), f is the frequency of the wave, and λ is the wavelength of the wave.

Radio waves can be reflected and refracted in a manner similar to light. They are affected by the ground terrain, atmosphere and other objects.

Maxwell’s equations state that a time varying magnetic field produces an electric field and a time varying electric field produces a magnetic field. This is kind
of a chicken and egg situation.

Radio waves propagate outward from an antenna, at the speed of light. The exact nature of these waves is determined by the transmission medium. In free
space, they travel in straight lines, whereas in the atmosphere, they generally travel in a curved path. In a confined or guided medium, radio waves do not
propagate in the TEM mode, but rather in a TE or TM mode.

Radio waves interact with objects in three principle ways:

Reflection – A radio wave bounces off an object larger than its wavelength.
Diffraction – Waves bend around objects.
Scattering – A radiowave bounces off an object smaller than its wavelength.

Because of these complex interactions, radio wave propagation is often examined in three distinct regions in order to simplify the analysis:

Surface (or ground) waves are located very near the earth’s surface.
Space waves occur in the lower atmosphere (troposphere).
Sky waves occur in the upper atmosphere (ionosphere).

The boundaries between these regions are somewhat fuzzy. In many cases, it is not possible to examine surface waves without considering space waves.

Common RF Band Designations

Surface Waves

These are the principle waves used in AM, FM and TV broadcast. Objects such as buildings, hills, ground conductivity, etc. have a significant impact on their
strength. Surface waves are usually vertically polarized with the electric field lines in contact with the earth.

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Refraction

Because of refraction, the radio horizon is larger than the optical horizon by about 4/3. The typical maximum direct wave transmission distance (in km) is
dependent on the height of the transmitting and receiving antennas (in meters):

However, the atmospheric conditions can have a dramatic effect on the amount of refraction.

Super Refraction

In super refraction, the rays bend more than normal thus shortening the radio horizon. This phenomenon occurs when temperature increases but moisture
decreases with height. Paradoxically, in some cases, the radio wave can travel over enormous distances. It can be reflected by the earth, rebroadcast and super
refracted again.

Sub refraction

In sub refraction, the rays bend less than normal. This phenomenon occurs when temperature decreases but moisture increases with height. In extreme cases,
the radio signal may be refracted out into space.

Space Waves

These waves occur within the lower 20 km of the atmosphere, and are comprised of a direct and reflected wave.

Direct Wave

This is generally a line of sight transmission, however, because of atmospheric refraction the range extends slightly beyond the horizon.

Ground Reflected Wave

Radio waves may strike the earth, and bounce off. The strength of the reflection depends on local conditions. The received radio signal can cancel out if the
direct and reflected waves arrive with the same relative strength and 180o out of phase with each other.

Horizontally polarized waves are reflected with almost the same intensity but with a 180o phase reversal.

Vertically polarized waves generally reflect less than half of the incident energy. If the angle of incidence is greater than 10o there is very little change in
phase angle.

Sky Waves

These waves head out to space but are reflected or refracted back by the ionosphere. The height of the ionosphere ranges from 50 to 600 km.

Radio waves are refracted by the ionized gas created by solar radiation. The amount of ionization depends on the time of day, season and the position in the
11-year sun spot cycle. The specific radio frequency refracted is a function of electron density and launch angle.

A communication channel thousands of kilometers long can be established by successive reflections at the earth’s surface and in the upper atmosphere. This
ionospheric propagation takes place mainly in the HF band.

The ionosphere is composed of several layers, which vary according to the time of day. Each layer has different propagation characteristics:

D layer – This layer occurs only during the day at altitudes of 60 to 90 km. High absorption takes place at frequencies up to 7 MHz.

E layer – This layer occurs at altitudes of 100 to 125 km. In the summer, dense ionization clouds can form for short periods. These clouds called
sporadic E can refract radio signals in the VHF spectrum. This phenomenon allows amateur radio operators to communicate over enormous
distances.

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F layer - This single nighttime layer splits into two layers (F1 and F2) during the day. The F1 layer forms at about 200 km and F2 at about 400
km. The F2 layer propagates most HF short-wave transmissions.

Because radio signals can take many paths to the receiver, multipath fading can occur. If the signals arrive in phase, the result is a stronger signal. If they
arrive out of phase with each other, they tend to cancel.

Deep fading, lasting from minutes to hours over a wide frequency range, can occur when solar flares increase the ionization in the D layer.

The useful transmission band ranges between the LUF (lowest usable frequency) and MUF (maximum usable frequency). Frequencies above the MUF are
refracted into space. Below the LUF, radio frequencies suffer severe absorption. If a signal is near either of these two extremes, it may be subject to fading.

Meteors create ionization trails that reflect radio waves. Although these trails exist for only a few seconds, they have been successfully used in
communications systems spanning 1500 Km.

The Aurora Borealis or Northern Lights cause random reflection in the 3 - 5 MHz region. Aurora causes signal flutter at 100 Hz to 2000 Hz thus making
voice transmission impossible.

Fading and Interference

Radio signals may vary in intensity for many reasons.

Flat Earth Reflections (Horizontal Polarization)

There are at least two possible paths for radio waves to travel when the antennas are near the earth: direct path and reflected path. These two signals interact
in a very complex manner. However, ignoring polarization and assuming a flat earth can produce some interesting mathematical descriptions.

p1 = direct wave path length

p2 = reflected wave path length

∆p = p2 - p1 difference in path lengths

d = distance

From the geometry we can observe:

But and

therefore

If the difference in the two paths ∆p, is 1/2 λ long, the two signals tend to cancel. If ∆p is equal to λ, the two signals tend to reinforce. The path difference ∆p
therefore corresponds to a phase angle change of:

The resultant received signal is the sum of the two components. The situation is unfortunately made more complex by the fact that the phase integrity of the
reflected wave is not maintained at the point of reflection.

If we limit the examination of reflected waves to the horizontally polarized situation, we obtain the following geometry:

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Applying the cosine rule to this diagram, we obtain a resultant signal of:

The signal strength of the direct wave is the unit distance value divided by the distance: Therefore, the received signal can be written as:

For small angles this can be approximated by:

Multipath Fading

The received signal is generally a combination of many signals, each coming over a different path. The phase and amplitude of each component are related to
the nature of the path. These signals combine in a very complex manner. Some multipath fading effects are characterized by delay spread, Rayleigh and
Ricean fading, doppler shifting, etc. Fading is the most significant phenomenon causing signal degradation. There are several different categories of fading:

Flat fading: the entire pass band of interest is affected equally (also known as narrow or amplitude varying channels).

Frequency selective fading: certain frequency components are affected more than others (also known as wideband channels). This
phenomenon tends to introduce inter-symbol interference.

Slow fading: the channel characteristics vary at less than the baud rate.

Fast fading: the channel characteristics vary faster than the baud rate.

Time Dispersion

Time dispersion occurs when signals arrive at different times. Signals traveling at the speed of light move about 1 foot in 1 nSec. This spreading tends to limit
the bit rate over RF links.

Rayleigh Fading

The Rayleigh distribution can be used to describe the statistical variations of a flat fading channel. Generally, the strength of the received signal falls off as
the inverse square of the distance between the transmitter and receiver. However, in cellular systems, the antennas are pointed slightly down and the signal
falls of more quickly.

Ricean Fading

The Ricean distribution is used to describe the statistical variations of signals with a strong direct or line-of-sight component and numerous weaker reflected
ones. This can happen in any multipath environment such as inside buildings or in an urban center.

A received signal is generally comprised of several signals, each taking a slightly different path. Since some may add constructively in-phase and others out
of phase, the overall signal strength may vary by 40 dB or more if the receiver is moved even a very short distance.

Doppler Shift

A frequency shift is caused by the relative motion of the transmitter and receiver, or any object that reflects/refracts signal. This movement creates random
frequency modulation. Doppler frequency shift is either positive or negative depending on whether the transmitter is moving towards or away from the
receiver.

This Doppler frequency shift is given by:

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vm is the relative motion of the transmitter with respect to the receiver, c is the speed of light and fc is the transmitted frequency. In the multipath
environment, the relative movement of each path is generally different. Thus, the signal is spread over a band of frequencies. This is known as the Doppler
spread.

Atmospheric Diffraction

Radio waves cannot penetrate very far into most objects. Consequently, there is often a shadow zone behind objects such as buildings, hills, etc.

The radio shadow zone does not have a very sharp cutoff due to spherical spreading, also called Huygens’ principle. Each point on a wavefront acts as it were
a point source radiating along the propagation path. The overall wavefront is the vector sum of all the point sources or wavelets. The wavelet magnitude is
proportional to 1 + cosθ where θ is measured from the direction of propagation. The amplitude is a maximum in the direction of propagation and zero in the
reverse direction.

Reflection
Diffraction
Path Loss
Rayleigh Fading
Rician Fading
Doppler Shift
This page is going to talk about the effect of noise on transmission systems.

Types of Noise

Most man made electro-magnetic noise occurs at frequencies below 500 MHz. The most significant of these include:

• Hydro lines
• Ignition systems
• Fluorescent lights
• Electric motors

Therefore deep space networks are placed out in the desert, far from these sources of interference.

There are also a wide range of natural noise sources which cannot be so easily avoided, namely:

•Atmospheric noise - lighting < 20 MHz

•Solar noise - sun - 11 year sunspot cycle
•Cosmic noise - 8 MHz to 1.5 GHz
•Thermal or Johnson noise. Due to free electrons striking vibrating ions.
•White noise - white noise has a constant spectral density over a specified range of frequencies. Johnson noise is an example of white noise.
•Gaussian noise - Gaussian noise is completely random in nature however, the probability of any particular amplitude value follows the normal
distribution curve. Johnson noise is Gaussian in nature.
•Shot noise - bipolar transistors

where q = electron charge 1.6 x 10-17 coulombs

•Excess noise, flicker, 1/f, and pink noise < 1 KHz are Inversely proportional to frequency and directly proportional to temperature and dc current
•Transit time noise - occurs when the electron transit time across a junction is the same period as the signal.

Of these, only Johnson noise can be readily analysed and compensated for. The noise power is given by:

Pn = kTB

Where:

k = Boltzman's constant (1.38 x 10-23 J/K)

T = temperature in degrees Kelvin
B = bandwidth in Hz

This equation applies to copper wire wound resistors, but is close enough to be used for all resistors. Maximum power transfer occurs when the source and
load impedance are equal.

Combining Noise Voltages

The instantaneous value of two noise voltages is simply the sum of their individual values at the same instant.

This result is readily observable on an oscilloscope. However, it is not particularly helpful, since it does not result in a single stable numerical value such as
one measured by a voltmeter.

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If the two voltages are coherent [K = 1], then the total rms voltage value is the sum of the individual rms voltage values.

If the two signals are completely random with respect to each other [K = 0], such as Johnson noise sources, the total power is the sum of all of the individual
powers:

A Johnson noise of power P = kTB, can be thought of as a noise voltage applied through a resistor, Thevenin equivalent.

An example of such a noise source may be a cable or transmission line. The amount of noise power transferred from the source to a load, such as an amplifier
input, is a function of the source and load impedances.

If the load impedance is 0 Ω, no power is transferred to it since the voltage is zero. If the load has infinite input impedance, again no power is transferred to it
since there is no current. Maximum power transfer occurs when the source and load impedances are equal.

The rms noise voltage at maximum power transfer is:

Observe what happens if the noise resistance is resolved into two components:

From this we observe that random noise resistance can be added directly, but random noise voltages add vectorially:

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If the noise sources are not quite random, and there is some correlation between them [0 < K < 1], the combined result is not so easy to calculate:

where

K = correlation [0 < K < 1]

R0 = reference impedance

Noise Temperature
The amount of noise in a given transmission medium can be equated to thermal noise. Thermal noise is well-studied, so it makes good sense to reuse the same
equations when possible. To this end, we can say that any amount of radiated noise can be approximated by thermal noise with a given effective
temperature. Effective temperature is measured in Kelvin. Effective temperature is frequently compared to the standard temperature, To, which is 290
Kelvin.

In microwave applications, it is difficult to speak in terms of currents and voltages since the signals are more aptly described by field equations. Therefore,
temperature is used to characterize noise. The total noise temperature is equal to the sum of all the individual noise temperatures.

Noise Figure

The terms used to quantify noise can be somewhat confusing but the key definitions are:

Signal to noise ratio: It is either unitless or specified in dB. The S/N ratio may be specified anywhere within a system.

Noise Factor (or Noise Ratio): (unit less)

Noise Figure: NF = 10logF = SNRin − SNRout dB

This parameter is specified in all high performance amplifiers and is measure of how much noise the amplifier itself contributes to the total noise. In a perfect
amplifier or system, NF = 0 dB. This discussion does not take into account any noise reduction techniques such as filtering or dynamic emphasis.

Friiss' Formula & Amplifier Cascades

It is interesting to examine an amplifier cascade to see how noise builds up in a large communication system.

Amplifier gain can be defined as:

Therefore the output signal power is: Sout = GSin

and the noise factor (ratio) can be rewritten as:

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The output noise power can now be written: Nout = FGNin

From this we observe that the input noise is increased by the noise ratio and amplifier gain as it passes through the amplifier. A noiseless amplifier would
have a noise ratio (factor) of 1 or noise figure of 0 dB. In this case, the input noise would only be amplified by the gain since the amplifier would not
contribute noise.

The minimum noise that can enter any system is the Johnson Noise:

Therefore the minimum noise that can appear at the output of any amplifier is:

The output noise of a perfect amplifier would be (F = 1):

The difference between these two values is the noised created (added) by the amplifier itself:

This is the additional (created) noise, appearing at the output.

The total noise out of the amplifier is then given by:

If a second amplifier were added in series, the total output noise would consist the first stage noise amplified by the second stage gain, plus the additional
noise of the second amplifier:

If we divide both sides of this expression by the common term: G1G2kTB

we obtain:

Recall:

Then:

This process can be extended to include more amplifiers in cascade to arrive at:

Friiss' Formula

This equation shows that the overall system noise figure is largely determined by the noise figure of the first stage in a cascade since the noise contribution of
any stage is divided by the gains of the preceding stages. This is why the 1st stage in any communication system should be an LNA (low noise amplifier).

Receiver Sensitivity
In a given bandwidth, W, we can show that the noise power N equals:

N = F(kT0)W

From N, we can show that the sensitivity of the receiver is equal to

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Cascaded Systems
This page will discuss the topic of signal propagation through physical mediums, such as wires.

This section of the Communication Systems book is a stub. You can help by expanding this section.

Transmission Line Equation

Many kinds of communication systems require signals at some point to be conveyed over copper wires.

The following analysis requires two assumptions:

• A transmission line can be decomposed into small, distributed passive electrical elements
• These elements are independent of frequency (i.e. although reactance is a function of frequency, resistance, capacitance and inductance are not)

These two assumptions limit the following analysis to frequencies up to the low MHz region. The second assumption is particularly difficult to defend since it
is well known that the resistance of a wire increases with frequency because the conduction cross-section decreases. This phenomenon is known as the skin
effect and is not easy to evaluate.

The purpose behind the following mathematical manipulation is to obtain an expression that defines the voltage (or current) at any time (t) along any portion
(x) of the transmission line. Later, this analysis will be extended to include the frequency domain.

Recall the characteristic equations for inductors and capacitors:

and

Kirchoff's Voltage Law

Kirchoff's voltage law (KVL) simply states that the sum of all voltage potentials around a closed loop equal zero. Or in other words, if you walked up a hill
and back down, the net altitude change would be zero.

Applying KVL in the above circuit, we obtain:

Rearranging:

But the LHS (left hand side) of the above equation, represents the voltage drop across the cable element ∆v, therefor:

Dividing through by ∆x, we obtain:

The LHS is easily recognized as a derivative. Simplifying the notation:

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This expression has both current and voltage in it. It would be convenient to write the equation in terms of current or voltage as a function of distance or time.

Simplifying the Equation (trust me)

The first step in separating voltage and current is to take the derivative with respect to the position x (Equation 1):

The next step is to eliminate the current terms, leaving an expression with voltage only. The change in current along the line is equal to the
current being shunted across the line through the capacitance C and conductance G. By applying KCL in the circuit, we obtain the necessary
information (Equation 2):

Taking the derivative with respect to time, we obtain (Equation 3):

Substituting (Equation 2) and (Equation 3) into (Equation 1), we obtain the desired simplification:

Collecting the terms, we obtain:

The Transmission Line Equation for Voltage

This equation is known as the transmission line equation. Note that it has voltage at any particular location x as a function of time t.

Similarly for current, we obtain:

The Transmission Line Equation for Current

But we're not quite done yet.

Solving the Transmission Line Equation

Historically, a mathematician would solve the transmission line equation for v by assuming a solution for v, substituting it into the equation, and observing
whether the result made any sense. An engineer would follow a similar procedure by making an “educated guess” based on some laboratory experiments, as
to what the solution might be. Today there are more sophisticated techniques used to find solutions. In this respect, the engineer may lag behind the
mathematician by several centuries in finding applications for mathematical tools.

To solve the transmission line equation, we shall guess that the solution for the voltage function is of the form:

The first term represents a unity vector rotating at an angular velocity of ω radians per second, in other words, a sine wave of some frequency. The second
term denotes the sinusoid being modified by the transmission line, namely its amplitude decaying exponentially with distance. If we let γ be a complex
quantity, we can also include any phase changes which occur as the signal travels down the line.

The sine wave is used as a signal source because it is easy to generate, and manipulate mathematically. Euler’s Identity shows the relationship between
exponential notation and trigonometric functions:

Euler's Identity

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Going back to our educated guess, we will let γ = α + jβ, therefore:

The term e − αx represents the exponential amplitude decay as this signal travels down the line. α is known as the attenuation coefficient and is
expressed in Nepers per meter.

The term represents the frequency of the signal at any point along the line. β component is known as the phase shift coefficient, and is
expressed in radians per meter.

Substituting our educated guess

into the transmission line equation for voltage, we obtain:

This looks pretty intimidating, but if you can do basic differentials and algebra, you can do this!

Simplifying the Equation (trust me)

The idea now is to work through the math to see if we come up with a reasonable solution. If we arrive at a contradiction or an unreasonable result, it means
that our educated guess was wrong and we have to do more experimenting and come up with a better guess as to how voltage and current travel down a
transmission line.

Let's look at this equation one term at a time:

LHS = RHS Term 1 + RHS Term 2 + RHS Term 3

Starting with the left hand side (LHS) we get the following simplification:

Believe it or not, the RHS Term 1 does not need simplifying.

Simplifying the RHS Term 2, we obtain:

Simplifying the RHS Term 3, we obtain:

Let's put it all back together:

Note that each of the four terms contain the expression .

Therefore we end up with:

And this can be further simplified to:

Attenuation and Phase Shift Coefficients

This result is not self contradictory or unreasonable. Therefore we conclude that our educated guess was right and we have successfully found an expression

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for attenuation and phase shift on a transmission line as a function of its distributed electrical components and frequency.

Lossless Transmission Line

Signal loss occurs by two basic mechanisms: signal power can be dissipated in a resistor [or conductance] or signal currents may be shunted to an AC ground
via a reactance. In transmission line theory, a lossless transmission line does not dissipate power. Signals, will still gradually diminish however, as shunt
reactances return the current to the source via the ground path. For the power loss to equal zero, R = G = 0. This condition occurs when the transmission line
is very short. An oscilloscope probe is an example of a very short transmission line. The transmission line equation reduces to the voltage equation:

and the current equation:

To determine how sinusoidal signals are affected by this type of line, we simply substitute a sinusoidal voltage or current into the above expressions and solve
as before, or we could take a much simpler approach. We could start with the solution for the general case:

Let R = G = 0, and simplify:

Equating the real and imaginary parts:

α=0

This expression tells us that a signal travelling down a lossless transmission line, experiences a phase shift directly proportional to its frequency.

Phase Velocity

A new parameter, known as phase velocity, can be extracted from these variables:

meters per second

Phase velocity is the speed at which a fixed point on a wavefront, appears to move. In the case of wire transmission lines, it is also the velocity of
propagation., typically: 0.24c < Vp < 0.9c .

The distance between two identical points on a wavefront is its wavelength (λ) and since one cycle is defined as 2π radians:

and ω = 2πf

therefore:

Vp = λf

In free space, the phase velocity is 3 x 108 meters/sec, the speed of light. In a cable, the phase velocity is somewhat lower because the signal is carried by
electrons. In a waveguide transmission line, the phase velocity exceeds the speed of light.

Distortionless Transmission Line

A distortionless line does not distort the signal phase, but does introduce a signal loss. Since common transmission lines are not super conductors, the signal
will decrease in amplitude but retain the same shape as the input. This characteristic is essential for long cable systems.

Phase distortion does not occur if the phase velocity Vp is constant at all frequencies.

By definition, a phase shift of 2π radians occurs over one wavelength λ.

Since

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Then:

This tells us that in order for phase velocity Vp to be constant, the phase shift coefficient β, must vary directly with frequency ω.

Recall

The problem now is to find β. This can be done as follows:

It may seem that we have lost β, but do not give up. The 2nd and 3rd roots can be expanded by means of the Binomial Expansion.

Recall:

In this instance n = 1/2. Since the contribution of successive terms diminishes rapidly, γ is expanded to only 3 terms:

This may seem complex, but remember it is only algebra and it will reduce down to simple elegance. Expanding the terms we obtain:

Since γ = α + jβ, we merely have to equate the real and imaginary terms to find β.

Or

Note that if then

From this we observe that β is directly proportional to ω.

Therefore the requirement for distortionless transmission is:

RC = LG

This is one of the essential design characteristics for a broadband coax cable network.

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If we equate the real terms, we obtain:

So there is a reason to study algebra after all!

The Frequency Domain

Signal analysis is often performed the frequency domain. This tells us how the transmission line affects the spectral content of the signals they are carrying.

To determine this, it is necessary to find the Fourier Transform of the transmission line equation. Recall:

and recall (hopefully) the Fourier Transform (which converts the time domain to the frequency domain):

To prevent this analysis from ‘blowing up’, we must put a stipulation on the voltage function namely, that it vanishes to zero at an infinite distance down the
line. This comprises a basic boundary condition.

This stipulation is in agreement with actual laboratory experiments. It is well known that the signal magnitude diminishes as the path lengthens.

Likewise, a time boundary condition, that the signal was zero at some time in the distant past and will be zero at some time in the distant future, must be
imposed.

Although engineers have no difficulty imposing these restrictions, mathematical purists, are somewhat offended. For this and other reasons, other less
restrictive transforms have been developed. The most notable in this context, is the Laplace transform, which does not have the same boundary conditions.

Having made the necessary concessions in order to continue our analysis, we must find the Fourier Transform corresponding to the following terms:

Then applying the transform on the derivative, we obtain:

This equation can be solved by using integration by parts:

Applying the boundary conditions when t goes to infinity makes the 1st term disappear.

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Note that the resulting integral is simply the Fourier Transform. In other words:

similarly:

We can now write the transmission line equation in the frequency domain:

where:

Rearranging the terms, we obtain:

or

since:

then

or

This represents the most general form of the transmission line equation in the frequency domain. This equation must now be solved for V to observe how
voltage (or current) varies with distance and frequency. This can be done by assuming a solution of the form:

These terms represent an exponential decay as the signal travels down the transmission line. If we ignore any reflections, assuming that the cable is infinitely
long or properly terminated, this simplifies to:

V = V0e − γx

To verify whether this assumption is correct, substitute it into the equation, and see if a contradiction occurs. If there is no contradiction, then our assumption
constitutes a valid solution.

γ2V0e − γx − γ2V0e − γx = 0

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0=0

Thus we validate the assumed solution. This tells us that in the frequency domain, the voltage or current on a transmission line decays exponentially:

V = V0e − γx

where:

In exponential notation, a sinusoid may be represented by a rotating unity vector, of some frequency:

ejωt = cosωt + jsinωt

Note that the magnitude of this function is 1, but the phase angle is changing as a function of t.

If we let: V0 = ejωt

Then:

This result is quite interesting because it is the same solution for the transmission line equation in the time domain. The term e − αx represents an exponential
decay. The signal is attenuated as length x increases. The amount of attenuation is defined as:

Attenuation in Nepers:

Attenuation in dB:

This allows us to determine the attenuation at any frequency at any point in a transmission line, if we are given the basic line parameters of R, L, G, & C.

The term represents a rotating unity vector since:

The phase angle of this vector is βx radians.

Characteristic Impedance
The characteristic impedance of a transmission line is also known as its surge impedance, and should not be confused with its resistance. If a line is infinitely
long, electrical signals will still propagate down it, even though the resistance approaches infinity. The characteristic impedance is determined from its AC
attributes, not its DC ones.

Recall from our earlier analysis:

This page will discuss Wireless EM wave propagation, and some basics about antennas.

Isotropic Antennas

In communication we talk about 'antennas'; insects have 'antennae'

An isotropic antenna radiates it's transmitted power equally in all directions. This is an ideal model; all real antennas have at least some directionality
associated with them. However, it is mathematically convenient, and good enough for most purposes.

A radio antenna is an example of an isotropic antenna

Power Flux Density

If the transmitted power is spread evenly across a sphere of radius R from the antenna, we can find the power per unit area of that sphere, called the Power
Flux Density using the greek letter Φ (capital phi) and the following formula:

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Where PT is the total transmitted power of the signal.

Effective Area

The effective area of an antenna is the equivalent amount of area of transmission power, from a non-ideal isotropic antenna that appears to be the area from
an ideal antenna. For instance, if our antenna is non-ideal, and 1 meter squared of area can effectively be modeled as .5 meters squared from an ideal antenna,
then we can use the ideal number in our antenna. We can relate the actual area and the effective area of our antenna using the antenna efficiency number, as
follows:

The area of an ideal isotropic antenna can be calculated using the wavelength of the transmitted signal as follows:

Received Power

The amount of power that is actually received by a receiver placed at distance R from the isotropic antenna is denoted PR, and can be found with the
following equation:

PR = ΦRAe

Where ΦR is the power flux density at the distance R. If we plug in the formula for the effective area of an ideal isotropic antenna into this equation, we get
the following result:

Where LP is the path-loss, and is defined as:

The amount of power lost across freespace between two isotropic antenna (a transmitter and a receiver) depends on the wavelength of the transmitted signal.

Directional Antennas
A directional antenna, such as a parabolic antenna, attempts to radiate most of it's power in the direction of a known receiver.

A "satellite dish" is an example of a parabolic antenna

Here are some definitions that we need to know before we proceed:

Azimuth Angle
The Azimuth angle, often denoted with a θ (greek lower-case Theta), is the angle that the direct transmission makes with respect to a given reference
angle (often the angle of the target receiver) when looking down on the antenna from above.
Elevation Angle
The elevation angle is the angle that the transmission direction makes with the ground. Elevation angle is denoted with a φ (greek lower-case phi)

Directivity

Given the above definitions, we can define the transmission gain of a directional antenna as a function of θ and φ, assuming the same transmission power:

Effective Area

The effective area of a parabolic antenna is given as such:

Transmit Gain

If we are at the transmit antenna, and looking at the receiver, the angle that the transmission differs from the direction that we are looking is known as Ψ
(greek upper-case Psi), and we can find the transmission gain as a function of this angle as follows:

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Where denotes the first-order bessel function.

Friis Equation

The Friis Equation is used to relate several values together when using directional antennas:

The Friis Equation is the fundamental basis for link-budget analysis.

Link-Budget Analysis
If we express all quantities from the Friis Equation in decibels, and divide both sides by the noise-density of the transmission medium, N0, we get the
following equation:

C / N0 = EIRP − LP + (GR / Te) − k

Where C/N0 is the received carrier-to-noise ratio, and we can decompose N0 as follows:

N0 = kTe

k is Boltzmann's constant, (-228.6dBW) and Te is the effective temperature of the noise signal (in degrees Kelvin). EIRP is the "Equivalent Isotropic
Radiated Power", and is defined as:

EIRP = GTPT

To perform a link-budget analysis, we add all the transmission gain terms from the transmitter, we add the receive gain divided by the effective temperature,
and we subtract boltzman's constant and all the path losses of the transmission.

This page is all about Space-Division Multiplexing (SDM).

What is SDM: When we want to transmit multiple messages, the goal is maximum reuse of the given resources: time and frequency. Time-Division
Multiplexing (TDM), operates by dividing the time up into time slices, so that the available time can be reused. Frequency-Division Multiplexing
(FDM), operates by dividing up the frequency into transmission bands, so that the frequency spectrum can be reused. However, if we remember our
work with directional antennas, we can actually reuse both time and frequency, by transmitting our information along parallel channels. This is known
as Space-Division Multiplexing.

Technical categorisations
Spatial beamforming

Spactial Coding

Multipathing
Application systems
MIMO Systems

Smart antenna

Section 4: Digital Modulation

Definition
What is PAM? Pulse-Amplitude Modulation is "pulse shaping". Essentially, communications engineers realize that the shape of the pulse in the time domain
can positively or negatively affect the characteristics of that pulse in the frequency domain. There is no one way to shape a pulse, there are all sorts of
different pulse shapes that can be used, but in practice, there are only a few pulse shapes that are worth the effort. These chapters will discuss some of the
common pulses, and will develop equations for working with any generic pulse.

Square Wave
The most logical way to transmit a digital signal is through a stream of pulses. One distinct pulse for a digital "1", and another distinct pulse for a digital "0".
Intuitively, a square pulse will transmit this data, and there are a number of different ways to transmit the data using

The square wave is a basic choice for transmitting digital data because it is easy to transmit, and is generally easy to receive. If we take the fourier transform
of a square wave, we get a sinc function. A sinc function is a never-ending function, which means that a square wave in the time domain has a very wide
bandwidth. When using a square wave, there will always be a trade-off, because high-frequency components of the square wave will be attenuated by the
channel, and the resultant waveform will be more prone to error on the other end.

Unipolar Square Wave

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A unipolar square wave is a wave where a logical 1 is transmitted using a square pulse of height A. Then a logical 0 is transmitted with a 0 voltage.

Bipolar Square Wave

A bipolar square wave is a square wave where a 1 is transmitted with a pulse of height A/2, and a 0 is transmitted with a pulse of -A/2.

Other pulses
It turns out that simply by changing the shape of the pulse (changing it away from a square wave), we can create a signal with better properties: lower
bandwidth, lower error rate, etc... This section will talk about a few different pulses.

Sinc
By the property of duality, however, we can see that if we have a sinc wave in the time domain, we will have a square-shape in the frequency domain. This is
an interesting result, because it means that we can transmit a sinc shape with definite bandwidth bounds, and it can travel through a channel in it's entirety
without being attenuated, or losing any harmonics!

Comparison
Here we will show a basic comparison between square pulses and sinc pulses:

Metric Square Pulse Sinc Pulse

Bandwidth Large bandwidth Small bandwidth
Jitter Not susceptable to Jitter Very susceptable to Jitter
Noise Very susceptable to Noise Susceptable to Noise
ISS Not affected by ISS Affected heavily by ISS

slew-rate-limited pulses
Most systems that use RS232, RS485, or high speed clocks use slew-rate-limited drivers. Some use driver chips with integrated slew-rate limiting; other
systems use digital driver chips that put sharp square pulses on their output pins, then an external slew-rate-limiting resistor[1] between that output pin and the
rest of the system.

Raised-Cosine Rolloff
Sinc pulses use less bandwidth, but they are very susceptable to jitter. Conversely, Square pulses have a large bandwidth, but are very resistant to jitter. The
decision seems hopeless, but the intrepid communications engineer will try to find a 3rd option. The 3rd option is called the "Raised-Cosine Rolloff" pulse,
and has the best characteristics of each wave. Raised Cosine Rolloff (RCR) pulses are resistant to jitter, and at the same time have reasonably good
bandwidth. Granted the bandwidth of an RCR pulse is wider than a sinc pulse, and the jitter resistance isn't as good as with a square wave, but this is a
compromise, after all.

Pulse Bandwidth Jitter Resistance

Rectangular rb/2 Excellent
Sinc Bad
RCR Good

Binary symmetric pulses

A system uses binary symmetric pulses if it transmits only 2 kinds of pulse, s1 and s0, and s1(t) = -s0(t). In a symmetric case, our comparator circuits are very
easy: If the correlator outputs a positive value, it's a binary 1. If it outputs a negative value, it's a binary 0.

Asymmetric Pulses
Assymetric pulses are more difficult for many reasons:

1. The threshold where the comparator should test the value may not be zero
2. The correlation receiver needs to correlate two different signals.
3. The different pulses may have different power, and they may be different susceptable to noise.

Asymmetric Correlation Receiver

When asymmetric pulses are used, the receiver system must actually employ 2 coherent optimal receivers, each one tuned to receive one of the pulses. When
two coherent optimal receivers are used, the receiver that is outputting the highest voltage value at the end of time T has received it's pulse.

... why not employ only 1 coherent optimal receiver, tuned to receiver neither s0 nor s1, but instead tuned to receive the difference (s1 - s0) ? ...

Some PAM systems, such as Ethernet 100BASE-T2, use PAM-5 ... (PAM)

This page discusses the binary modulation schemes and "keying".

What is "Keying?"
Square waves, sinc waves, and raised-cosine rolloff waves are all well and good, but all of them have drawbacks. If we use an optimal, matched filter, we can

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eliminate the effect of jitter, so frankly, why would we consider square waves at all? Without jitter as a concern, it makes no sense to correct for jitter, or even
take it into consideration. However, since the matched filter needs to look at individual symbols, the transmitted signal can't suffer from any intersymbol
interference either. Therefore, we aren't using the sinc pulse.

Since the raised-cosine roll-off wave suffers from both these problems (in smaller amounts, however), we dont want to use that pulse either.

So the question is, what other types of pulses can we send?

It turns out that if we use some of the techniques we have developed using analog signal modulation, and implement a sinusoidal carrier wave, we can create
a signal with no inter-symbol interference, very low bandwidth, and no worries about jitter. Just like analog modulation, there are 3 aspects of the carrier
wave that we can change: the amplitude, the frequency, and the phase angle. Instead of "modulation", we call these techniques keying techniques, because
they are operating on a binary-number basis.

There is one important point to note before continuing with this discussion: Binary signals are not periodic signals. Therefore, we cannot expect that a binary
signal is going to have a discrete spectra like a periodic squarewave will have. For this reason, the spectral components of binary data are continuous spectra.

Amplitude Shift Keying

In an ASK system, we are changing the amplitude of the sine wave to transmit digitial data. We have the following cases:

Binary 1: A1sin(fct)

Binary 0: A0sin(fct)

The simplest modulation scheme sets A0 = 0V (turning the transmitter off), and setting A1 = +5V (any random non-zero number turns the transmitter on).
This special case of ASK is called OOK (On-Off keying). Morse code uses OOK.

Another common special case of ASK sets A1 to some positive number, and A0 to the corresponding negative number A0 = -A1. We will mention this case
again later.

In ASK, we have the following equation:

a(t)sin(ωt)

by the principal of duality, multiplication in the time domain becomes convolution in the frequency domain, and vice-versa. Therefore, our frequency
spectrum will have the following equation:

A(jω) * δ(t − ω)

where the impulse function is the fourier-transform of the sinusoid, centered at the frequency of the wave. the value for A is going to be a sinc wave, with a
width dependant on the bitrate. We remember from the Signals and Systems book that convolution of a signal with an impulse is that signal centered where
the impulse was centered. Therefore, we know now that the frequency domain shape of this curve is a sinc wave centered at the carrier frequency.

Frequency Shift Keying

In Frequency Shift Keying (FSK), we can logically assume that the parameter that we will be changing is the frequency of the sine wave. FSK is unique
among the different keying methods in that data is never transmitted at the carrier frequency, but is instead transmitted at a certain offset from the carrier
frequency. If we have a carrier frequency of fc, and a frequency offset of ∆f, we can transmit binary values as such:

Binary 1: Asin((fc + ∆f)t)

Binary 0: Asin((fc − ∆f)t)

Similar to ASK, we have FSK, which uses 2 different frequencies to transmit data. For now we will call them ω1,ω2. Using the same logic that we used
above, the fourier representations of these waves will be (respectively):

A1(jω) * δ(t − ω1)

A0(jω) * δ(t − ω2)

With one sinc wave centered at the first frequency, and one sinc wave centered at the second frequency. Notice that A1 and A0 are the half-square waves
associated with the 1s and the 0s, respectively. These will be described later.

Error Rate

The BER of coherent BPSK in the presence of gaussian and Rayleigh noise is as follows:

Gaussian Noise Rayleigh Fading

Phase Shift Keying

PSK systems are slightly different then ASK and FSK systems, and because of this difference, we can exploit an interesting little trick of trigonometry. PSK
is when we vary the phase angle of the wave to transmit different bits. For instance:

Binary 1: Asin(fct + φ1)

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Binary 0: Asin(fct + φ0)

If we evenly space them out around the unit-circle, we can give ourselves the following nice values:

Binary 1: Asin(fct + 0)

Binary 0: Asin(fct + π)

Now, according to trigonometry, we have the following identity:

sin(fct + π) = − sin(fct)

So in general, our equations for each signal (s) is given by:

s1(t) = Asin(fct)

s0(t) = − Asin(fct)

Which looks awfully like an ASK signal. Therefore, we can show that the spectrum of a PSK signal is the same as the spectrum of an ASK signal.

There are two commonally used forms of Phase Shift keying Modulation:

Binary Phase Shift Keying (BPSK)

Quadrature Phase Shift Keying (QPSK)

Binary Phase Shift keying is set out above.

QPSK

Quadrature Phase Shift Keying utilises the fact that a cosine wave is in quadrature to a sine wave, alowing 2 bits to be simultaniously represented.

Binary 11: Asin(fct + 0) + cos(fc + π / 2)

Binary 10: Asin(fct + 0) + cos(fc − π / 2)

Binary 01: Asin(fct + π) + + cos(fc + π / 2)

Binary 00: Asin(fct + π) + + cos(fc − π / 2)

QPSK has the advantage over BPSK of requiring half the transmission band width for the same data rate, and error probability.

Error Rate

The BER of coherent BPSK in the presence of gaussian and Rayleigh noise is as follows:

Gaussian Noise Rayleigh Fading

Binary Transmitters
Binary Receivers
Now what if try to cram more information into a single bittime? If we take 2 bits at a time, and arrange them together, we can assign each set of 2 bits to a
different symbol, and then we can transmit the different symbols.

Pronunciation
First off, "M-ary" is not pronounced like the female name "mary". If you say "mary" to a boardroom filled with engineers, they will laugh at you, and your
boss will politely tell you later to start cleaning up your desk. "M-ary" is pronounced more like the word "memory", without the leading 'm'. Try pronouncing
it like "emery".

Example: 4-ASK
Let us use the following scheme:

"00" = +5V

"01" = +1.66V

"10" = -1.66V

"11" = -5V

we can see now that we can transmit data twice as fast using this scheme, although we need to have a more complicated receiver, that can decide between 4
different pulses (instead of 2 different pulses, like we have been using).

Bits Per Symbol

All popular communication systems transmit an integer number of bits per symbol. We can relate the number of bits ("k") and the number of different
symbols ("m") with the following equation:

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m = 2k

This causes the number of symbols to be a power of two.

With M-ary modulation techniques, the "symbols per second" rate can be much slower than the "bits per second" data rate.

QPSK
CPFSK (MSK)
[MSK]minimum shift keying

DPSK
For further reading

Wikipedia:Constellation_diagram

Wikipedia:Quadrature amplitude modulation

The quadrature amplitude modulation (QAM) system of modulation is the most popular M-ary scheme.

Definition
Consider the case of a system with two carrier waves instead of a single carrier wave as we have considered with modulation schemes so far. One is a sine
wave, and the other is a cosine wave of the same frequency. Since these two waves are orthogonal we can use them simultaneously in a single channel
without losing the information of either. If both waves have the same frequency f we can write out the equation for a generic symbol, s:

s(t) = Aksin(ft) + Bkcos(ft)

In this way, we can create multiple symbols by simply changing the values of A and B. This equation can be broken up into two parts:

Aksin(ft) Which is called the "in-phase" component of the equation.

Bkcos(ft) Which is called the "quadrature" component of the equation.

An equation which is written as a sum of a sine plus a cosine is said to be in "quadrature form". If we combine the two components into a single waveform as
such:

This form is called the "Polar Form" of the equation.

56K modems and Digital TV use QAM

Constellation Plots
If we make a graph with the X axis being the values for A, and the Y axis being the values for B, we get what is called a Constellation Plot. These plots are
called constellation plots due to the similiarity in shape and layout with astronomical star charts. The A and B values for each symbol are plotted (the "stars")
and various measurements between them are used to determine information from the system. On a constellation plot, we can see a number of rules:

1. The further apart the points are on the constellation plot, the less likely they are to be mistaken for each other in the presence of noise.
2. The closer the points are to the origin, the less power it takes to send.
3. The more points there are, the faster the data rate (bit rate) at a fixed symbol rate (more symbols)
4. The fewer points there are, the simpler and cheaper the hardware necessary to distinguish between them (fewer symbols, fewer thresholds in the
receiver).

For these reasons there is no single "best" constellation plot, but it is up to the engineer to pick the points that are best for the system. In other words, trade
offs need to be made between speed, performance, and cost of hardware. These tradeoffs can be made by placing the constellation points at different locations
on the constellation plot.

Benefits of QAM
For further reading
(QAM)

This page will discuss the mathematical basis, and the design of noncoherent receivers.

This section of the Communication Systems book is a stub. You can help by expanding this section.

Line Codes

In addition to pulse shaping, there are a number of useful line codes that we can use to help reduce our errors or to have other positive effects on our signal.

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Line coding consists of representing the digital signal to be transported, by an amplitude- and time-discrete signal, that is optimally tuned for the specific
properties of the physical channel (and of the receiving equipment). The waveform pattern of voltage or current used to represent the 1s and 0s of a digital
signal on a transmission link is called line encoding. The common types of line encoding are unipolar, polar, bipolar and Manchester encoding.

Line codes are used commonly in computer communication networks over short distances.

There are numerous ways digital information can be coded onto a transmission medium. Some of the more common include:

Each of the various line formats has a particular advantage and disadvantage. It is not possible to select one, which will meet all needs. The format may be
selected to meet one or more of the following criteria:

• Minimize transmission hardware

• Facilitate synchronization
• Ease error detection and correction
• Minimize spectral content
• Eliminate a dc component

The Manchester code is quite popular. It is known as a self-clocking code because there is always a transition during the bit interval. Consequently, long

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strings of zeros or ones do not cause clocking problems.

Non-Return to Zero Codes (NRZ)

Non-Return to Zero (NRZ) codes are a bit awkwardly named, especially considering that the unipolar NRZ code does return to a zero value. In essence, an
NRZ code is just a simple square wave, assigning one value to a binary 1, and another amplitude to a binary 0.

NRZ codes are more bandwidth efficient than bipolar ones. However, their spectral components go all the way down to 0 Hz. This prevents them from being
used on transmission lines which are transformer coupled, or for some other reason cannot carry DC.

Unipolar NRZ

Unipolar NRZ is simply a square wave with +AV being a binary 1, and 0V being a binary 0. NRZ is convenient because computer circuits use unipolar NRZ
internally, and it requires little effort to expand this system outside the computer. Unipolar NRZ has a DC term, but a relatively narrow bandwidth.

Bipolar NRZ

... This definition conflicts with Wikipedia:Non-return-to-zero#Bipolar_Non-Return-to-Zero_Level and Wikipedia:bipolar encoding. What to do? ...

Bipolar NRZ operates in the following way: "0" is always denoted by 0V, but "1" is denoted by an alternating sequence of +5V and -5V. For instance, let's
say we have the digital signal 101011011:

+5V 0V -5V 0V +5V -5V 0V +5V -5V

1 0 1 0 1 1 0 1 1

Bipolar NRZ has better bandwidth than unipolar NRZ, and the bipolar case is guaranteed to never have a DC component.

AMI

AMI (alternate mark inversion) is an example of a bipolar line code. Each successive mark is inverted and the average or DC level of the line is therefor zero.
This system cannot readily be used on fiber optic links.

AMI is usually implemented as RZ pulses.

One of the weaknesses of this approach is that long strings of zeros cause the receivers to loose lock. It is therefor necessary to impose other rules on the
signal to prevent this.

CMI

The CMI bipolar line code is a subset of AMI. Marks are encoded as alternate polarity full period pulses. Spaces are encoded by half a period at the negative
voltage and half period at the positive voltage. This coding scheme has the advantage that it uses only two voltage levels instead of 3, and requires less logic
to implement than HDB3.

Manchester

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Manchester codes were an invention that allows for timing information to be sent along with the data. In an NRZ code, if there is a long stream of ones or
zeros, the receiver could conceivably suffer so much compound jitter that it would either lose or gain an entire bit time, and then be out of sync with the
transmitter. This is because a long stream of 1s or 0s would not "change state" at all, but instead would simply stay at a single value. Manchester codes say
that every single bittime will have a transition in the middle of the bit time, so that a receiver could find that transition, and "lock on" to the signal again, if it
started to stray away from center. Because there are more transitions, however, manchester codes also require the highest bandwidth of all the line codes.

Ethernet LAN uses manchester codes

Differential Codes
Some communication channels (such as phase-modulated sine waves and differential twisted pairs) have the characteristic that transitions between the 2
symbols used can be easily distinguished, but when first starting to receive it is difficult to tell which of the 2 states it is in. For example, full-speed USB uses
a twisted pair and transmits +3.3 V on one line and 0 V on the other line for "1", but 0 V on the one line and +3.3 V on the other line for "0". Because some
cables have an extra half-twist in them, it is impossible for a device that was just plugged in to tell whether the symbol it is currently receiving is a "1" or a
"0".

Differential codes still work, not even noticing when the 2 wires get switched.

Differential codes, in general, look exactly the same on a oscilloscope or spectrum analyzer as the non-differential code they are based on, and so use exactly
the same bandwidth and have exactly the same bitrate.

Differential codes that work when the 2 wires get switched include:

Differential Manchester encoding -- based on Manchester encoding

Non-Return-to-Zero Inverted (NRZI) -- based on non-return-to-zero (NRZ)

(A few non-differential codes also work even when the 2 wires get switched -- such as bipolar encoding, and MLT-3 encoding).

Differential NRZ

Differential Manchester

Comparison
Code bandwidth timing DC value
Unipolar NRZ low bandwidth no timing information High DC component
Bipolar NRZ lower bandwidth no timing information no DC component
Differential NRZ lower bandwidth no timing information little or no DC component
Manchester high bandwidth Good clock recovery no DC component
Differential Manchester moderate bandwidth Good clock recovery No DC Component

further reading

Wikipedia:differential coding

Section 5: Analog Networks

Old folks may very well remember the first incarnation of the telephone networks, where an operator sitting at a desk would physically connect different
wires to transmit a phone call from one house to another house. The days however when an operator at a desk could handle all the volume and all the
possibilities of the telephone network are over. Now, automated systems connect wires together to transmit calls from one side of the country to another
almost instantly.

== What is Circuit-Switching? == Circuit switching is a mechanism of assigning a predefined path from source node to destination node during the entire
period of connection. Plain old telephone system (POTS) is a well known example of analogue circuit switching..

Strowger Switch
Cross-Bar Switch
Telephony
Telephone Network

Rotary vs Touch-Tone

Cellular Network Introduction

further reading

We will go into more details of the telephone network in a later chapter, Communication Networks/Analog and Digital Telephony.

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The cable television network is something that is very near and dear to the hearts of many people, but few people understand how cable TV works. The
chapters in this section will attempt to explain how cable TV works, and later chapters on advanced television networks will discuss topics such as cable
internet, and HDTV.

coax cable has a bandwidth in the hundreds of megahertz, which is more then enough to transmit multiple streams of video and audio simultaneously. Some
people mistakenly think that the television (or the cable box) sends a signal to the TV station to tell what channel it wants, and then the TV station sends only
that channel back to your home. This is not the case. The cable wire transmits every single channel, simultaneously. It does this by using frequency division
multiplexing.

TV Channels
Each TV channel consists of a frequency range of 6MHz. Of this, most of it is video data, some of it is audio data, some of it is control data, and the rest of it
is unused buffer space, that helps to prevent cross-talk between adjacent channels.

Scrambled channels, or "locked channels" are channels that are still sent to your house on the cable wire, but without the control signal that helps to sync up
the video signal. If you watch a scrambled channel, you can still often make out some images, but they just don't seem to line up correctly. When you call up
to order pay-per-view, or when you buy another channel, the cable company reinserts the control signal into the line, and you can see the descrambled
channel.

A descrambler, or "cable black box" is a machine that artificially recreates the syncronization signal, and realigns the image on the TV. descrambler boxes
are illegal in most places.

NTSC
PAL
SECAM
HDTV
Everybody has a radio. Either it is in your house, or it is in your car. The pages in this chapter will discuss some of the specifics of radio transmission, will
discuss the differences between AM and FM radio.

AM Radio
FM Radio
Amateur Radio
Other Modulated Audio
Section 6: Digital Networks

In a digital communications system, there are 2 methods for data transfer: parallel and serial. Parallel connections have multiple wires running parallel to
each other (hence the name), and can transmit data on all the wires simultaneously. Serial, on the other hand, uses a single wire to transfer the data bits one at
a time.

Parallel Data
The parallel port on modern computer systems is an example of a parallel communications connection. The parallel port has 8 data wires, and a large series of
ground wires and control wires. IDE harddisk connectors are another good example of parallel connections in a computer system.

Serial Data
The serial port on modern computers is a good example of serial communications. Serial ports have a single data wire, and the remainder of the wires are
either ground or control signals. USB and FireWire are good examples of other serial communications standards.

Which is Better?
It is a natural question to ask which one of the two transmission methods is better. At first glance, it would seem that parallel ports should be able to send data
much faster than serial ports. Let's say we have a parallel connection with 8 data wires, and a serial connection with a single data wire. Simple arithmetic
seems to show that the parallel system can transmit 8 times as fast as the serial system.

However, parallel ports suffer extremely from inter-symbol interference (ISI) and noise, and therefore the data can be corrupted over long distances. Also,
because the wires in a parallel system have small amounts of capacitance and mutual inductance, the bandwidth of parallel wires is much lower then the
bandwidth of serial wires. We all know by now that an increased bandwidth leads to a better bit rate. We also know that less noise in the channel means we
can successfully transmit data reliably with a lower Signal-to-Noise Ratio, SNR.

If, however, we bump up the power in a serial connection by using a differential signal with 2 wires (one with a positive voltage, and one with a negative
voltage), we can use the same amount of power, have twice the SNR, and reach an even higher bitrate without suffering the effects of noise. USB cables, for
instance, use sheilded, differential serial communications, and the USB 2.0 standard is capable of data transmission rates of 480Mbits/sec!

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In addition, because of the increased potential for noise and interference, parallel wires need to be far shorter then serial wires. Consider the standard parallel
port wire to connect the PC to a printer: those wires are between 3 and 4 feet long. Now consider ethernet wires (which are serial): they can be bought in
lengths of 50 feet or more!

UART, USART
This page will discuss the OSI Reference Model

OSI Model
Layer What It Does
Application The application layer is what the user of the computer will see and interact with. This layer is the "Application" that the programmer
Layer develops.
Presentation
The Presentation Layer is involved in formatting the data into a human-readable format, and translating different languages, etc...
Layer
The Session Layer will maintain different connections, in case a single application wants to connect to multiple remote sites (or form
Session Layer
multiple connections with a single remote site).
Transport The Transport Layer will handle data transmissions, and will differentiate between Connection-Oriented transmissions (TCP) and
Layer connectionless transmissions (UDP)
The Network Layer allows different machines to address each other logically, and allows for unreliable data transmission between
Network Layer
computers (IP)
Data-Link The Data-Link Layer is the layer that determines how data is sent through the physical channel. Examples of Data-Link protocols are
Layer "Ethernet" and "PPP".
The Physical Layer consists of the physical wires, or the antennas that comprise the physical hardware of the transmission system.
Physical Layer
Physical layer entities include WiFi transmissions, and 100BaseT cables.

What It Does

The OSI model allows for different developers to make products and software to interface with other products, without having to worry about how the layers
below are implemented. Each layer has a specified interface with layers above and below it, so everybody can work on different areas without worrying about
compatability.

Packets
Higher level layers handle the data first, so higher level protocols will touch packets in a descending order. Let's say we have a terminal system that uses TCP
protocol in the transport layer, IP in the network layer, and Ethernet in the Data Link layer. This is how the packet would get created:

1. Our application creates a data packet

|Data|

2. TCP creates a TCP Packet:

|TCP Header|Data|

3. IP creates an IP packet:

|IP Header|TCP Header|Data|CRC|

4. Ethernet Creates an Ethernet Frame:

|Ethernet Header|IP Header|TCP Header|Data|CRC|

On the receiving end, the layers receive the data in the reverse order:

1. Ethernet Layer reads and removes Ethernet Header:

|IP Header|TCP Header|Data|CRC|

2. IP layer reads the IP header and checks the CRC for errors

|TCP Header|Data|

3. TCP Layer reads TCP header

|Data|

4. Application reads data.

It is important to note that multiple TCP packets can be squeezed into a single IP packet, and multiple IP packets can be put together into an Ethernet Frame.

Common Alterations
Other Reference Models

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Channels
A channel is a communication medium, the path that data takes from source to destination. A channel can be comprised of so many different things: wires,
free space, and entire networks. Signals can be routed from one type of network to another network with completely different characteristics. In the Internet, a
packet may be sent over a wireless WiFi network to an ethernet lan, to a DSL modem, to a fiber-optic backbone, et cetera. The many unique physical
characteristics of different channels determine the three characteristics of interest in communication: the latency, the data rate, and the reliability of the
channel.

Bandwidth and Bitrate

Bandwidth is the difference between the upper and lower cutoff frequencies of, for example, a filter, a communication channel, or a signal spectrum.
Bandwidth, like frequency, is measured in hertz (Hz). The bandwidth can be physically measured using a spectrum analyzer.

Bandwidth, given by the variables Bw or W is closely related to the amount of digital bits that can be reliably sent over a given channel:

rb = 2W

where rb is the bitrate. If we have an M-ary signaling scheme with m levels, we can expand the previous equation to find the maximum bit rate for the given
bandwidth.

rb = 2Wm

Example: Bandwidth and Bitrate

Let's say that we have a channel with 1KHz bandwidth, and we would like to transmit data at 5000 bits/second. We would like to know how many
levels of transmission we would need to attain this data rate. Plugging into the second equation, we get the following result:

However, we know that in M-ary transmission schemes, m must be a multiple of 2. Rounding up to the nearest multiple of 2, we find that m = 4.

Channel Capacity
The "capacity" of a channel is the theoretical upper-limit to the bit rate over a given channel that will result in negligible errors. Channel capacity is measured
in bits/s.

Shannon's channel capacity is an equation that determines the information capacity of a channel from a few physical characteristics of the channel. A
communication systems can attempt to exceed the Shannon's capacity of a given channel, but there will be many errors in transmission, and the expense is
generally not worth the effort. Shannon's capacity, therefore, is the theoretical maximum bit rate below which information can be transmitted with negligible
errors.

The Shannon channel capacity, C, is measured in units of bits/sec and is given by the equation:

C = Wlog2(1 + SNR)

C is the maximum capacity of the channel, W is the available bandwidth in the channel, and SNR is the signal to noise ratio, not in DB.

Because channel capacity is proportional to analog bandwidth, some people call it "digital bandwidth".

Channel Capacity Example

The telephone network has an effective bandwidth less then 3000Hz (but we will round up), and transmitted signals have an average SNR less
than 40dB (10,000 times larger). Plugging those numbers into Shannon's equation, we get the following result:

C = 3000log2(1 + 10000) = 39863.569...

we can see that the theoretical maximum channel capacity of the telephone network (if we generously round up all our numbers) is approximately
40Kb/sec!. How then can some modems transmit at a rate of 56kb/sec? it turns out that 56k modems use a trick, that we will talk about in a later
chapter.

Acknowledgement
Digital information packets have a number of overhead bits known as a header. This is because most digital systems use statistical TDM (as discussed in the
Time-Division Multiplexing chapter). The total amount of bits sent in a transmission must be at least the sum of the data bits and the header bits. The total
number of bits transmitted per second (the "throughput") is always less than the theoretical capacity. Because some of this throughput is used for these header
bits, the number of data bits transmitted per second (the "goodput") is always less than the throughput.

In addition, since we all want our information to be transmitted reliably, it makes good sense for an intelligent transmitter and an intelligent receiver to
communicate directly to each other, to ensure reliable transmission. This is called acknowledgement, and the process is called hand-shaking.

An essential part of acknowledgement is forward error correction, a subject that we will talk about more in depth later. Forward error correction (FEC) is
the process of embedding some sort of checksum (called a CRC sum in IP communications) into the packet header. The receiver then reads this checksum,
determines if there is an error in the transmission, and then sends back an acknowledgement packet.

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ARQ: ACK and NAK

In an acknowledgement request (ARQ) scheme, the transmitter sends out data packets, and the receiver will then send back an acknowledgement. A positive
acknowledgement (called "ACK") means that the packet was received without any detectable errors. A negative acknowledgement (called "NAK") means
that the packet was received in error. Generally, when a NAK is received by the transmitter, the transmitter will send the packet again.

Streaming Packets

In some streaming protocols, such as RTP, the transmitter is sending time-sensitive data, and it can therefore not afford to wait for acknowledgement packets
and checksum calculations. In these types of systems, the receiver will attempt to detect errors in the received packets, and if an error is found, the bad packet
is simply deleted.

The Internet
The Internet has become, arguably, the most important and pervasive example of a network on the planet. The Internet connects people from across a street,
and from around the globe at nearly the speed of light. The sections in this chapter will discuss some fundamentals about the internet, and some more
advanced chapters on the subject will be discussed later in the book.

Client-Server
The exact relationship between a client and a server in the traditional Client-Server relationship can be complicated, and in today's world, the distinctions are
more hazy still. This page will attempt to define what a client and a server are, how they are related, and how they communicate.

Client
A client is a communications entity that requests information
Server
A server is a communications entity that supplies information, or responds to a request.

Packet-Switching Networks
In the page on Time-Division Multiplexing (TDM), we talked about how we can use a method of breaking information up into chunks, and prefixing that data
with address information to send multiple streams of data over a single channel. These chunks of data with their headers are called packets, and are the basis
for communication in the internet.

In local area networks (LAN), packets are sent over baseband channels for very short distances. These networks use Statistical TDM (packets) to control
access to the channel, and to control the process of moving information from it's source to it's address. This process of gettings things where they need to go is
called routing. Devices that route packets are called (surprise!) routers.

Over larger networks such as Wide Area Networks (WAN), high-capacity broadband channels such as fiber optic cables connect different local LANs. Over
a WAN network, packets are then Frequency Division Multiplexed (FDM) to flow simultaneously over these broad channels. At the other end, these packets
are moved back down to a baseband system, and routed using TDM techniques again.

When talking about the different components in a computer network, a lot of different words fly around: Routers, Hubs, Switches, and Gateways, are some
example. This page will talk about some of the different peices of hardware that make the Internet possible. This page will only offer brief explanations of
each component, opting instead to save complex discussions on the workings of each for later chapters (or even later books!).

Hubs
An Ethernet hub, normally just called a hub, is a networking device used to connect multiple Ethernet segments in order to create a primitive LAN. They are
primarily connected using unshielded twisted pairs/shielded twisted pairs (UTP/STP) or Fiber Optic wires, and require no special administration to function.
Hubs operate on the physical layer (Layer 1) of the OSI model, and indiscriminately forward frames to every other user in the domain.

Hubs perform a variety of tasks, including:

Acting as a multi-port repeater,

Forwarding frames to all users,

Allowing for connectivity to networks using 10base2 or 10base5 media,

Partitioning ports when excessive collisions occur, which isolates the affected segments,

And broadcasting a "jam" signal when collisions take place on Ethernet media.

Hubs also extend but do not control collision domains, absorbing bandwidth and allowing excessive collisions to occur and hinder performance, when
switches or bridges can effectively break up a network.

Despite the increase of switches as the connection medium for workgroups, hubs can still operate in a number of situations:

Small computer clusters,

Careless users/saboteurs cannot interfere with a hub's function,

Connecting outdated networks to newer methods of data transmission,

And when budget is a priority over functionality.

Switches and Routers

Wikipedia:Routers

Wikipedia:Network Switch

Gateways

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Repeaters

Proxys

What is Ethernet?
Ethernet was invented in 1973 at Xerox Corporation's Palo Alto Research Center(PARC) by a researcher named Bob Metcalfe. Bob Metcalfe was asked to
build a networking system for the computers at PARC. Xerox wanted such a network because they were designing the world's first laser printer and wanted
all of the computers there to be able to print using it. There were two challenges he faced, he had to make it fast enough for the laser printer as well as be able
to connect hundreds of computers in the same building together.

Ethernet is U.S. Patent #4,063,220

Physical Ethernet Connections

Ethernet Communications
CSMA/CD
Ethernet uses the carrier sense multiple access with collision detection (CSMA/CD) algorithm for arbitration. CSMA/CD logic helps prevent collisions and
also defines how to act when a collision does occur. The CSMA/CD algorithm works like this:

a. A device with a frame to send listens until Ethernet is not busy.

b. When the Ethernet is not busy, the sender begins sending the frame.

c. The sender listens to make sure that no collision occurred.

d. Once the sender hears the collision, they each send a jamming signal, to ensure that all stations recognize the collision.

e. After the jamming is complete, each sender randomizes a timer and waits that long.

f. When each timer expires, the process starts over with Step 1.

So, all devices on the Ethernet need to use CSMA/CD to avoid collisions and to recover when inadvertent collisions occur.

Additional Specifications
http://inventors.about.com/library/weekly/aa111598.htm (IEEE 802.3)

There are 2 different protocols that are prominent in the field of wireless internet: WiFi and WiMAX.

WiFi (IEEE 802.11)

Wifi simply stands for Wireless Fidelity.

802.11a

What is 802.11a and history of 802.11a?

It is a Wireless LAN standard from the IEEE(Institute of Electronics and Electrical Engineers). It was released on October 11th in 1999.

Speed:

It can achieve a maximum speed of 54Mbps. Although the typical data rate transfer is at 22Mbps. If there is a need the data rate will be reduced to 48, 36, 24,
18, 12, 9 then 6Mbps respectively. This usually occurs as the distance between the access point or the wireless router and the computer gets further and
further away.

Frequency:

It operates under the 5Ghz frequency band. The advantage of this is that it has lesser interference compared to the 802.11b and 802.11g standards, which
operate at 2.4Ghz. It means that quite a number of electronic equipment use this frequency band such as microwaves, cordless phones, bluetooth devices etc.
Therefore, the more electronic equipment that use the same frequency band, the more interferences it will cause among the equipment that are using that
frequency band.

Interoperability:

802.11a will not operate readily with 802.11b or 802.11g due to the different frequency bands unless the equipment implements the both standards. E.g.
Equipment that use both 802.11a and 802.11g standards.

Number of Channels

It has 12 non-overlapping channels. 8 are for indoor(within the area) and the other 4 are for point to point.

802.11b

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What is 802.11b and its history

It is also something like 802.11a. It is of course a wireless standard made by IEEE and guess what it was implemented on the same month and year as
802.11a which was in 0ctober 1999.

Speed

802.11b has the lowest speed after 802.11 legacy. It can reach a maximum speed of only 11 Mbps.

Frequency

802.11g

802.11n

WiMax (IEEE 802.16)

Section 7: Hybrid Networks

Coaxial Cable
Coaxial cable has an incredibly high bandwidth (compared to twisted pair), and it distorts very little over long distances. For this reason, coax is able to carry
a large number of analog television channels to a very large audience.

Bi-Directional Cable
The original implementation of the television network only needed to move data in one direction: from the station to the homes. For this reason, a number of
amplifiers were installed in the network that take the signal from the base station, and amplify that signal towards the homes. However, a problem arises
when cable internet users want to transmit data back to the base station (upload).

The original cable TV network had a very large amount of available bandwidth, but it wasnt designed to transmit data from the user back to the network.
Instead, the entire network was set up with directional amplifiers, that would amplify data going to the user, but wouldnt affect data coming back from the
user.

HDTV
HDTV is the next generation of television, and actually allows better resolution, larger frame size, and lower bandwidth then the traditional analog signals.
Also, digital signals are less prone to cross-talk between channels, so channels don't need to be spaced as far apart in the frequency domain as analog signals
are.

This chapter will discuss the next generations of the cable TV network.

Channels
The TV channels in an analog TV scheme carry channels spaced every 6MHz from about 150MHz to 500MHz. Below 150MHz was considered originally to
be too susceptable to noise, and there was simply no need to expand above 500MHz. However, with the advent of cable internet, the system needed to be
revamped.

Bandwidth
A new band was set aside, from 55MHz to 75MHz, to allow traffic to be uploaded from the user. Also, another band was set aside, from 550 to 750MHz to
allow for cable internet downloads. A cable modem would be able to demodulate these two bands of data, without interfereing with the TV signal.

Problems
200MHz of download bandwidth seems like a lot, but every household on a given line (and there could be 100 or more) all need to share this bandwidth,
which can slow down the system, especially in heavily populated areas.

Satellite TV
Modems Introduction
The telephone network was originally designed to carry voice data. The human ear can only really hear sounds up to the 15KHz range, and most of that is just
high-frequency fluff noise that isn't needed to transmit human voice signals. Therefore, the decision was made to limit the telephone network to a maximum
transmission frequency of 3400Hz, and a minimum frequency of 400Hz (to limit the passage of DC signals, which could damage the circuit). This gives the
telephone network an effective bandwidth of 3000Hz, which is perfect for transmitting voice, but which isn't that good for transmitting anything else.

Original telephone modems would use the existing telephone network to carry internet signals to a remote ISP. However, new DSP modems use a much
larger frequency band, and this information is separated from the phone network almost as soon as it leaves your house. New voice technologies, such as
VoIP completely bypass the old telephone infrastructure, and instead transmit voice signals over the internet.

The chapters in this section will talk about the analog and digital hybrid nature of the telephone network.

Modems
Modems were the original wide-spread method for home users to connect to the internet. Modems modulated digital data according to different schemes (that

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changed as time passed), and transmitted that data through the telephone network.

The telephone network was originally designed to only transmit voice data, so most of the network installed a series of low-pass filters on the different lines,
to prevent high-frequency data or noise from damaging the circuits. Because of this, the entire telephone network can be seen as having a hard bandwidth of
3000Hz. In reality, the lines used have a much higher bandwidth, but the telephone network cuts out all the high-frequency signals. DSL modems make use of
that "lost bandwidth", but the original modems had to work within the 3000Hz limit.

If we take the shannon channel capacity of a telephone line (assuming a signal SNR of 40db, which is nearly impossible), we can get the following result:

If we then plug this result into Nyquist's equation, we can find how many levels of transmission we need to use to get this bitrate:

rb = 2Wlog2(m)

which gives

Therefore, using a 128-level transmission scheme, we can acheive a theoretical maximum bitrate of 40Kb/sec through a modem.

56k Modems
If the theoretical Shannon capacity of the telephone network is 40Kbps, how can modern modems achieve a speed of 56Kb/sec? The V.42 modem standard
(which is what a 56k modem is) utilizes a standard implementation of the Lemple-Ziv compression algorithm, to shrink the size of transmitted data, and
therefore apparently increase the transmission speed. The telephone companies aren't magically breaking the Shannon bound, they are just finding an
interesting path around it.

DSL
A single strand of twisted-pair telephone wire has a bandwidth of nearly 100KHz, especially over short distances. Over longer distances, noise will play a
much bigger role in the received signal, and the wire itself will attenuate the signal more with greater distance. This is why DSL is only offered in locations
that are close to the telephone office, and not in remote areas.

DSL signals require the addition of 2 new peices of hardware: The DSL modem, and the DSL splitter, which is located at the telephone company, and splits
the DSL signal (high frequencies) from the voice signal (low frequencies). Also, some houses may require the installation of additional filters, to prevent
cross-talk between DSL and voice signals.

VoIP
With the advent of modems and DSL technology, telephone companies have become an integral part of the internet. It's no surprise then, when phone calls
start getting digitized, and sent through the internet, instead of the old telephone network. Voice over IP (VoIP) is the logical conclusion to this train of
thought.

Further Reading

Voice over IP

Section 8: Advanced Internet

ARP Protocol and ARPANET Internet Backbone (Fiber Optics and SONET, ATM Networks)

It will be mentioned here, but also probably in every sub-chapter of this section that the intention of these chapters, much less of the entire book, that the
purpose of this is not to teach network programming. What these chapters do aim to do is provide a fast and dirty listing of available functions, and
demonstrate how they coincide with our previous discussions on networking. For further information on the subject, the reader is encouraged to check out
networking concepts on the programming bookshelf.

This page is not designed to be an in-depth discussion of C socket programming. Instead, this page would like to be a quick-and-dirty overview of C, in the
interests of reinforcing some of the networking concepts discussed earlier.

C and Unix
This section will (briefly) discuss how to program socket applications using the C programming language in a UNIX environment. The next section will then
discuss the differences between socket programming under Windows, and will explain how to port socket code from UNIX to Windows platforms.

C and Windows (Winsock)

Programming sockets in Windows is nearly identical to programming sockets in UNIX, except that windows requires a few different additions:

1. Use <Winsock.h>
2. Link to ws2_32.dll
3. Initialize Winsock with WSAStartup( )

The first 2 points are self-explanatory, and are actually dependant on your compiler, which we will not go into here. However, the 3rd point needs a little
explaining. We need to initialize winsock before we can use any of the socket functions (or else they will all return errors). To initialize, we must create a new
data object, and pass it to the initialization routine:

WSADATA wd;

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And we must pass a pointer to this structure, along with the requested version number for winsock to the function:

WSAStartup(MAKEWORD(2, 0), &wd);

The MAKEWORD macro takes two numbers, a major version number (the 2, above), and a minor version number (the 0, above). For instance, to use
Winsock2.0, we use MAKEWORD(2, 0). To use Winsock1.1, we use MAKEWORD(1, 1).

Also, it is important to note that Windows does not allow sockets to be read and written using the generic unix read( ) and write( ) functions. In windows,
you should instead use the recv( ) and send( ) functions. Also, people who are familiar with windows should know that windows treats sockets as an I/O
handle, so they can also be accessed with the Windows generic read/write functions ReadFile( ) and WriteFile( ).

Winsock, and windows in general also have a number of other functions available in winsock.dll, but also in other dlls (like wininet.dll) that will facilitate
higher-level internet operations.

Other Socket Implementations

Further Reading
For a comprehensive examination of the berkeley socket API in C, see the C Programming wikibook page on UNIX networking:
Programming:C_Networking_in_UNIX. DNS

IP Protocol
We've talked so far about TDM (Time-Division Multiplexing) techniques, and we've also talked about how different packets in a given network can contain
address information, that will help the routers along the way move the data to it's destination. This page will talk about one of the most important parts of the
internet: the IP Protocol.

What is IP?
IP stands for Internet Protocol, and is essentially what makes the internet different from other digital networks (ARPANET, for instance). IP protocol
assigns a unique address, called the "IP Address" to each computer in a network, and these IP addresses are used to route packets of information from a
source to a destination. IP protocol calls for each device in the network to make the best effort possible to transmit the data, but IP doesn't guarantee that the
data will arrive. If you are looking for a guarantee, you will have to implement a higher-level protocol (such as TCP).

From the OSI model, the IP Protocol is a Network-Layer Protocol.

The IP address is a different number from the "MAC Address" that is also found inside a computer. The IP address is a 32bit value that is unique among
computers in a given local network. A MAC address is a larger number that is unique in the entire world. However, it is very difficult to route packets
according to the MAC address.

IP also specifies the header that packets must have when traveling across the internet. This header is called the IP header, and will be discussed in the next
chapter.

The IP Protocol also specifies that each IP packet must have an error-checking code attached to the end of the packet. This error-checking code, called the
"Cyclic Redundency Check" or CRC Checksum is capable of helping the receiving computer determine if the packet has had any bit errors during
transmission. The CRC code is much more powerful at detecting errors than a single parity bit is, but CRC can be time consuming to calculate.

ICMP
ICMP, the Internet Control Message Protocol is a counter-part to the IP Protocol that allows for control messages to be sent across the network.

The Ping command in Unix and Windows uses ICMP

IPv4 and IPv6

IP Header
The IP header is a large field of information that is appended to the beginning of the packet. The IP header includes a large amount of information about the
packet, including the source IP address, and the destination IP address. Also, the IP header (IPv6 and up) includes information about the local area networks
for both the source and the destination terminals.

CRC Checksum
The CRC checksum is a 16bit data item appended to the end of an internet IP packet. The CRC contains a number that the receiver runs through a particular
algorithm, and can then determine if the packet is correct, or if there is an error.

Modulo-2 Arithmetic

This section will be a short primer on Modulo-2 Arithmetic

Calculating the CRC

Using the CRC to Find Errors

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Classful Address Scheme

Subnet Masks
Classless Interdomain Routing (CIDR)
Further Reading

Wikipedia:Internet Protocol Suite

Wikipedia:IPv4

Wikipedia:IPv6

IPv6 Info & resources

TCP and UDP

The TCP and UDP protocols are 2 different protocols that handle data communications between terminals in an IP network (the internet). This page will talk
about what TCP and UDP are, and what the differences are between them.

In the OSI model, TCP and UDP are "Transport Layer" Protocols.

Connection-Oriented vs Connectionless
TCP
TCP stands for Transmission Control Protocol, and is one of the most popular protocols on the internet. TCP is a "Connection-Oriented" transmission
protocol, which means that it establishes a dedicated connection between 2 computers before sending data. TCP assigns each connection on a given computer
a locally unique number called a "Port" number. Each TCP connection can be uniquely defined in the whole world by its combination of IP address and Port
information for both the client and the server. In TCP, addresses are usually written as:

IPAddress:Port

When a terminal receives a TCP transmission, it sends back a receipt called an "Acknowledgment" (ACK) to the sender. The sender waits for the ACK, and if
it doesn't receive an ACK after a certain amount of time, the sender will resend the information. In this way, TCP tries to provide for a guarantee: If it doesn't
succeed the first time, try try again.

Common Examples of TCP in action are web-browsers, and the HTTP protocol.

TCP Headers

UDP
Unlike TCP, UDP doesn't establish a connection before sending data, it just sends. Because of this, UDP is called "Connectionless". UDP packets are often
called "Datagrams". An example of UDP in action is the DNS service. DNS servers send and receive DNS requests using UDP.

UDP Headers

HTTP
The Hypertext Transport Protocol (HTTP) is an application layer protocol that transmits text over the internet. HTTP stands for (Hyper-Text Transfer
Protocol). HTTP is based on the TCP/IP protocols, and is used commonly on the internet for transmitting webpages from servers to browsers.

What is HTTP
Applications that use HTTP protocol to send and receive text documents are called "Web browsers", and there is a good chance that you are reading this page
using a web browser that uses HTTP protocol.

HTTP protocol uses TCP protocol to create an established, reliable connection between the client (the web browser) and the server (wikibooks.org). All
HTTP commands are in plain text, and almost all HTTP requests are sent using TCP port 80, of course any port can be used. HTTP protocol asks that each
request be in IP address form, not DNS format. So if we want to load www.wikibooks.org, we need to first resolve the wikibooks.org IP address from a DNS
server, and then send out that request. Let's say (and this is impossible) that the IP address for wikibooks.org is 192.168.1.1. Then, to load this very page, we
would create a TCP packet with the following text:

GET 192.168.1.1/wiki/Communication_Systems/HTTP_Protocol HTTP/1.1

The first part of the request, the word "GET", is our HTTP command. The middle part of the request is the URL (Universal Resource Locator) of the page we
want to load, and the last part of the request ("HTTP/1.1") tells the server which version of HTTP the request is going to use.

When the server gets the request, it will reply with a status code, that is defined in the HTTP standard. For instance:

HTTP/1.1 200 OK

or the infamous

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HTTP/1.1 404 Not Found

The first part of the reply is the version of HTTP being used, the second part of the reply is the error code number, and the last part of the reply is the message
in plain, human-readable text.

HTTP Methods
HTTP Status Codes
HTTP Success Codes

2** Success

200 OK

HTTP Error Codes

4** Client Error

403 Forbidden

404 NOT FOUND

5** Server Error

500 INTERNAL ERROR

HTML Markup
FTP Protocol
Like HTTP, FTP (File-Transfer Protocol) is an application-level protocol that handles the sending and receiving of binary files. It is based on the TCP/IP
protocols.

FTP Header
FTP Commands
Type ftp at windows or unix shell, to enter ftp> prompt

? : gives list the command avaliable

commandname ? : echos the usage of that particular command

FTP Error Codes

Routing is the process of getting information packets where they need to go. Routing is a surprisingly complicated task, and there are a number of different
algorithms used to find the shortest route between two points.

Routing Tables
Routing Algorithms
Bellman-Ford Algorithm

Dijkstra's Algorithm

Wikipedia:Dijkstra Algorithm

Routing Types

Hop-by-Hop Routing

Link-State Routing

Wikipedia:Link State Routing

Distance-Vector Routing

Wikipedia:Distance Vector Routing

Section 9: Coding and Security

Basic Coding
It turns out that we can actually improve the peformance of our system simply by changing the bits in our system. We can change the order of bits, or we can
change the number of bits that we send, to help decrease error rates. This chapter will talk about transmission codes that can help increase system

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performance.

Gray Codes
Let us look at a basic QASK system:

1. "00" = +5V
2. "01" = +1.66V
3. "10" = -1.66V
4. "11" = -5V

We will call these "symbol 1" (for 00), "symbol 2" (for 01), "symbol 3" (for 10) and "symbol 4" (for 11).

Now, let us make a basic assumption that the only errors we get will be errors between adjacent symbols. For instance, Symbol 1 can be mistaken for symbol
2, but symbol 1 can never be mistaken for symbol 3. In reality, there is always a very small chance that this could happen, but we will simply ignore it for our
purposes.

Now, we have 6 error conditions:

1. Symbol 1 looks like Symbol 2

2. Symbol 2 looks like Symbol 1
3. Symbol 2 looks like Symbol 3
4. Symbol 3 looks like Symbol 2
5. Symbol 3 looks like Symbol 4
6. Symbol 4 looks like Symbol 3

We can also see that in error conditions 1, 2, 5, and 6, a symbol error only produces 1 bit error. The other bit is actually not transmitted in error. The only time
where we see 2 bits transmitted in error is when there is a mixup between symbols 2 and 3. We can calculate our probability of bit error as such:

Pone bit error is the probability that 1 bit from the symbol will be in error. Ptwo bits error is the probability that both symbols will be received in error.

To alleviate this problem, we can reorder our symbols, and map them the following way:

1. "00" = +5V
2. "01" = +1.66V
3. "11" = -1.66V
4. "10" = -5V

Now, when any symbol is received in error, there is only one bit error. This small trick is called "Gray Coding", and is a useful (but simple) method to
improve system performance in terms of net bit errors.

Using Gray Codes

Gray codes are a simple example of how using an encoding/decoding mechanism in a communications system can have an effect on the error rate of the
system. In future chapters we will discuss error correction codes (ECC) that can be used to find and fix single-bit errors in a signal. If we use Gray codes to
ensure that the majority of errors are single-bit errors, we can then use more advanced ECC codes to resolve these errors.

Gray codes are typically not used on their own, but are combined with more robust coding methods, like described above.

For further reading

Wikipedia:Manchester codes

Wikipedia:MLT-3 Encoding

Spread Spectrum
There are multiple ways in which multiple clients can share a single transmission medium. Time-Division Multiplexing (TDM) is the system where different
clients get "time slices", periods of time where one client gets to transmit and other clients cannot. Since nobody else is transmitting, each client gets to use
the entire spectrum, which means high bandwidth and a fast data rate. There is also Frequency-Division Multiplexing (FDM), where each client transmits at
the same time, but on different frequencies. In FDM there is less available bandwidth, but each client gets to transmit for more time.

There is one more type that we need to discuss: Code-Division Multiplexing (CDM). CDM is also known as Spread-Spectrum multiplexing. In CDM, all
clients can transmit at all times on all frequencies.

Other benefits to CDM, that we will discuss later, is it's use in encryption and the inability for another client to jam or interfere with a transmission.

Why CDM?
CDM, and the related CDMA, are technologies that were originally designed for use in the military. Many of the techniques described earlier in this book are
designed to make optimal use of a transmission medium, in terms of higher bitrate while simultaneously minimizing the necessary bandwidth. CDM,
however, takes the opposite approach of maximizing bandwidth in order to pursue other valuable qualities. CDM techniques specifically widen and flatten the
transmitted signal, in some instances the signal becomes so wide and so short that it actually falls below the noise floor, and is impossible to discern. In
addition, CDM signals rely on a special type of code called a PN code to decipher. These two qualities together mean that CDM transmissions are virtually
immune to eavesdropping or interception.

Another benefit to CDM is that since the bandwidth being used is spread out, it is nearly impossible for another person to jam the transmission with
interference. Also, since the transmission energy is spread out across a wide frequency range, the data signal can be indistinguishable from the background
noise, which can prevent eavesdropping.

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A third benefit, as if we need a third, is that CDM transmissions have the nearly magical property that we can simultaneously transmit multiple transmissions
in the same time slice, and in the same frequency band, without ISI problems. This is how modern cellphone networks operate: multiple cell phones can talk
at the same time in the same frequency band.

Direct CDM
Let's start out with an exercise. Let's say that we have a digital code, for instance 1011. What we will do is combine this code with another code, called the
spreading code that is 3 times as long, and traveling 3 times as fast.

But wait a minute, if the spreading code is moving 3 times as fast (3 times the bit rate), then it will require 3 times the bandwidth to transmit! Well, this is
correct. CDM requires more bandwidth to transmit a single signal then either TDM or FDM, but we will see in a minute why this doesnt matter to us.

let's say that our spreading code is as follows: 101100111000. Now, we will combine our information signal (1011) with the spreading code as follows: We
will logically XOR each digit of the information signal with 3 consecutive digits of the spreading code (because the spreading code is moving 3 times as fast.
We will use the "X" symbol to denote the XOR operation:

1 0 1 1
X 101 X 100 X 111 X 000
----- ----- ----- -----
010 100 011 100

And now our resulting code is the value 010100011100. This will be called our transmission code. Now, we can pick one of our binary transmission
methods(ASK, M-ary PSK, QAM, etc...) to transmit this resultant digital signal over the channel to the receiver.

To demultiplex a CDM signal, we make use of the following mathematical rule:

a X b X b = a

Remember, the "X" symbol here is the XOR operation. It would seem that if we XOR the transmission code with the spreading code, we will get back our
original information signal! The only problem is that we need to have the same exact spreading code on the receiving end, to demultiplex the signal.

Benefits of CDM
So what exactly is the benefit of CDM? First, let's take a look at the spreading code:

Spreading Code

The spreading code is selected to have a number of different properties, but the most important is that the spreading code must be a pseudo-random number
(PN). PN numbers look random, but since we must be able to generate the same signal on the receving end, the spreading code can't be completely random.
Because of this property, it can be very difficult for an eavesdropper to intercept our data, because other people do not necessarily know our spreading code.
Even if other people did know our spreading code, they would not necessarily know what point in the code we were at. Some spreading codes can be very
long.

Increased Bitrate

Our spreading code is being generated at a much higher bitrate then our data code is. According to Shannon's channel capacity, for the same SNR and a larger
bit rate, our signal will be spread out over a much larger bandwidth.

Security

Now, since we know that the Spreading code looks like noise, it is safe to assume that when we XOR in our information signal, the result will look like noise
as well. In fact, the transmitted signal looks so much like noise, that unless the receiver has a perfectly synchronized spreading code to use in demultiplexing,
the signal is impossible to recover. This makes CDM relatively secure.

Multiplexing

Using CDM, we can transmit multiple signals simultaneously over the exact same frequency band. This is important because TDM allows us to use the same
frequency band but it separates out time slices. FDM doesnt need time slices, but it separates out signals into different frequency bands. CDM allows us to
transmit signals at the exact same time, in the exact same frequency band.

Now, when we receive the multiplexed signals simultaneously, each receiver can XOR with the spreading code, to get back the original signal. Remember,
each transmission signal looks like noise, so the receivers will simply ignore the signals that don't correlate with the spreading code.

direct CDM (DSS)

Direct CDM, also called direct spread spectrum (DSS), multiplexes data using a trick with the digital XOR operation.

Examples of Direct CDM

Direct CDM techniques are used in Cellular phone transmissions in north america. This is because there are far too many cellphones to give each phone it's
own frequency range, or it's own time slice. Using CDM, all the phones can simultaneously talk over the same frequency band at the same time.

Frequency Hopping CDM (FHSS)

Frequency Hopping CDM, also called frequency hopping spread spectrum (FHSS), transmits on only one narrow frequency band for a very short amount of
time, then "hops" to some other frequency.

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Let's say we have our PN generator, which is creating a spreading code in real time. We take a certain number of bits from that PN generator at a time, and
use them to pick a frequency range. We then transmit a short burst of data on that range, and then hop to the next frequency range for the next burst of data.

If we have the same PN generator on the receiving end, we can then use that spreading code to pick which frequency bands to listen to at any given time.
Using frequency hopping, our signal will take more bandwidth then normal (because we are using multiple bands to send information on. Frequency Hopping
has several advantages including the fact that it allows us to make better use of more bandwidth simultaneously, and--if we hop fast enough--it provides some
protection against reflections ("fading") and noise sources obliterating a few of the narrow bands.

Frequency hopping has the disadvantage the the receiver cannot use PSK, QPSK, or M-ary PSK techniques, because it is just too hard to synchronize the
different phase angles at different times on different frequency bands. However M-ary FSK is frequently used, with great success.

Fast and Slow Hops

A system is called a "Slow Hopping System" if the hop rate is slower than the symbol rate (an integer number of symbols transmitted per hop). A system is
called a "Fast Hopping System" if the symbol rate is slower then the hop rate (an integer number of hops per each symbol). Fast hop systems are more
complicated to implement, but are more immune to channel distortion.

COFDM
A orthogonal frequency division multiplexing (OFDM) system is similar to a frequency hopping system in that part of the information is transmitted at one
narrowband frequency, and other information is transmitted at a different narrowband frequency. However, rather than transmitting at only one narrowband
frequency at a time, a OFDM system transmits at all of its frequencies all the time. Because any one narrow band it transmitted constantly, it's fairly easy to
synchronize phase angles and use the best modulation technique for that band (QPSK). Because all the frequencies are generated by one transmitter, it is
possible to pack them far more tightly together without interference than a system that dedicated each frequency to a different transmitter.

Reflections ("fading") and noise sources often obliterate a few of the narrow bands. To overcome this, the bits are "coded" (COFDM). A forward error
correction (Data Coding Theory/Forward Error Correction) code is spread across every channel, such that even if 2 channels are completely obliterated, only
2 bits of the code word are lost. Then the receiver uses the FEC to reconstruct the obliterated data.

In practice, practically all OFDM systems are coded orthogonal frequency division multiplexing (COFDM) systems.

Data Compression
When transmitting digital data, we find that frequently we can't send our information as quickly as we would like. Specific limits, such as Shannon's Channel
Capacity, restrict the amount of digital information that can be transmitted over a given channel. Therefore, if we want to transmit more data, we need to find
a way to make our data smaller.

To make data smaller, we implement one of a number of techniques known as data compression. There are a number of different compression algorithms, but
they can all be broken down into two categories: Lossless algorithms, and lossy algorithms.

The fundamental function of a compression is to remove redundancy, where redundancy is all that could be removed or expressed in a different way, whilst
not removing it's meaning.

Lossy Algorithms
Lossy algorithms are techniques that can be used to transmit a "pretty good" version of the data. With a lossy algorithm, there is always going to be a certain
amount of data lost in the conversion. Lossy algorithms provide much higher compression rates then lossless algorithms, but the downfall is that information
must be lost to attain those high rates. An example of a lossy compression algorithm is JPEG image files, or MP3 music files. If the lossy algorithm is good
enough, the loss might not be noticeable by the recipient.

Lossless Algorithms
Lossless algorithms decrease the size of a given signal, while at the same time not losing any information from the original. For this reason, Lossless
compression algorithms are preferable to lossy algorithms, especially when the data needs to arrive at the recipient intact. Examples of lossless compression
algorithms are ZIP files, and GIF images.

Run-Length Encoding
Run-length encoding (RLE) is probably one of the best known compression techniques. Here is how it works:

Let's assume we have some input data: 'aaaabbbc' Now RLE compresses this by expressing the amount of times each symbol occurs, if it occurs more than 2
times.

This 'compresses' to '4a3bc' which means as much as 4 x a, 3 x b, 1 x c.

We don't express a data item explicitly if it occurs twice or just once. We would just lose space.

Huffman Coding
Huffman coding is a very powerful compression technique that can be used as an optimal lossless encoding technique.

Huffman at wikipedia: w:Huffman coding

Variable-Length Encoding
Notes

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Data compression, while a related field to coding theory, is not strictly in the scope of this book, and so we will not cover it any further here.

Introduction
Transfer of information from one place to another faces many difficulties. Principal among them is noise. For example suppose 01010101 is sent from one
end. Due to noise, it may be received as 11010101, with the first digit changed by noise. Clearly if what is sent is not what is received, communication can be
problematic. Error correcting codes have been developed to solve this type of problem.

In practical applications it makes sense to send messages in blocks, i.e. a series of digits one after another, e.g. 11111100. It is well known that electronic data
are represented using 0's and 1's. Each digit (0 or 1) is called a bit. A byte is made up of 8 bits. A byte allows us to represent 28 = 256 symbols. Let's suppose
for the moment that data is sent a byte at time. As before, due to noise what is sent may not always be what is received.

Computer scientists came up with a simple error detection method called parity check. With this method, we represent data using only the first 7 bits. The last
bit is always chosen so that together with the other seven there are an even number of 1's in the byte. When the data arrives at the other end, the receiver
counts the number of 1's in the byte. If it's odd, then the byte must have been contaminated by noise so the receiver may ask for retransmission.

This method only detects errors but it can not correct them other than ask for retransmission. If retransmission is expensive (e.g. satellite), parity check is not
ideal. Also its error detection ability is very low. If two bits were changed by noise, then the receiver will assume the message is correct. More sophisticated
error correction codes address these problems.

Hamming Code
Hamming code is an improvement on the parity check method. It can correct 1 error but at a price. In the parity check scheme, the first 7 bits in a byte are
actual information so 27 = 128 different symbols may be represented using a byte. But for Hamming code each block of data contains 7 bits (not 8) and only 4
bits in a block are used to represent data, so only 24 = 16 symbols may be represented in a block. Therefore, to send the same amount of info in Hamming
code, we will need to send a lot more bits. Anyhow let's see how Hamming Code works.

In this section we will see that Hamming code has some amazing properties, although we will not discuss why it works just yet. In fact, if only one error is
introduced in transmission, i.e. only one bit got changed, then the decoding method employed by the receiver will definitely be able to correct it. It is easy to
appreciate then, that if 2 or more errors are made, correction and even detection may not be possible.

For now, we will describe how Hamming code works, but only later do we develop the mathematics behind it. Therefore, this section can be skipped if one
wishes.

We let each of a, b, c and d take the value 0 or 1, and these are called information bits. A Hamming code is a block of 7 bits in the form of

(a + b + d, a + c + d, a, b + c + d, b, c, d) (mod 2)

..give matrix repn as can be seen, the digits a, b, c and d appear on their own in component 3, 5, 6, and 7. All other components are a combination of a, b, c
and d. So we call the 3rd, 5th, 6th and 7th component the information component, all other components are check components, and these components carry
extra information that allows us to detect and correct single errors. We will explain the funny notation above in turn. The (mod 2) notation means that we take
each of the values in the bracket, separated by commas, and look at its value modular 2 (we will see an example later).

We have represented the block of 7 bits in vector form, where each component corresponds to a bit. E.g. let a = b= c = 1, then we have

(1 + 1 + 1, 1 + 1 + 1, 1 , 1 + 1 + 1, 1, 1, 1) (mod 2)

which is

(1, 1, 1, 1, 1, 1, 1)

therefore 1111111 is the block of bits we are sending.

To detect single errors, it's also very simple. Suppose the codeword received is a1a2...a7, we compute 3 values

b0 = a1 + a3 + a5 + a7(mod 2)
b1 = a2 + a3 + a7 + a8(mod 2)
b2 = a5 + a6 + a7 + a8(mod 2)

then we declare the error is at the 22b2 + 2b1 + b0th position.

Suppose 1111111 is sent but 1111011 is received. The receiver computes

b0 = 1 + 1 + 0 + 1 = 1(mod 2)
b1 = 1 + 1 + 1 + 1 = 0(mod 2)
b2 = 0 + 1 + 1 + 1 = 1(mod 2)

so the error is at the 22b2 + 2b1 + b0 = 4 + 1 = 5th position as predicted!

If no error is made in transmission, then 22b2 + 2b1 + b0 = 0.

Summary
If one wishes to send a block of information consisting of 4 bits. Let these 4 bits be abcd, say.

To send: abcd

we compute and send (a + b + d, a + c + d, a, b + c + d, b, c, d) (mod 2)

To decode:

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b0 = a1 + a3 + a5 + a7(mod 2)

b1 = a2 + a3 + a7 + a8(mod 2)

b2 = a5 + a6 + a7 + a8(mod 2)
2

e = 2 b + 2b + b
2 1 0
if e = 0 then assume no error, otherwise declare a single error occurred at the eth position.

Exercise

...compute some codewords and decode them

Basics
The mathematical theory of error correcting codes applies the notion of a finite field, called also Galois field (after the famous French mathematician, Evariste
Galois, 1811-1832). In particular, the 2-element set {0,1} supports the structure of a finite field of size 2. In general, a finite field may have q elements, where
q is a prime power (no other number of elements is possible in a finite field; any two finite fields with the same number of elements are essentially alike, i.e.
isomorphic). E.g. a 7-element field may consist of elements 0,1,2,3,4,5,6, and its arithmetic operations + - * are performed modulo 7.

We denote the finite field of size q as or GF(q). GF stands for Galois Field.

Some Definitions

Code & Codeword

Let A be a finite set, called alphabet; it should have at least two different elements. Let n be a natural number. A code of length n over alphabet A is any
set C of n-long sequences of elements from A; the sequences from C are called codewords of C.

If alphabet A = {0,1} then codes over A are called binary.

For example, let C be a code over alphabet A := GF(5) := {0,1,2,3,4}. Let C := {000,111,222,333,444}. It has codewords 000, 111, 222, 333 and 444.

Now we should discuss some properties of a code. Firstly, we can have the notion of distance between two codewords.

Hamming Distance

Let C be a code, and x and y (bold to signify that each codeword is like a vector) are codewords of C. The Hamming distance of x and y denoted
d(x,y)
is the number places in which x and y differ.

E.g. d(000,111) = 3.

Hamming distance enjoys the following three fundamental metric properties:

1. d(x,y) = 0 <==> 'x' = 'y'

2. d(x,y) = d(y,x)
3. d(x,y) ≤ d(x,z) + d(z,y); triangle inequality

Minimum distance

The minimum distance of a code C denoted d(C) is the minimum distance possible between two different codewords of C

E.g. Let C = {000,111,110,001}, then d(C) = d(000,001) = 1, as the distance between any other codewords are greater than or equal to 1.

The significance of minimum distance

The minimum distance of a code C is closely related to its ability to correct errors. Let's illustrate why, using a hypothetical code C. Let say this code has
minimum distance 5, i.e. d(C) = 5. If a codeword, x is sent and only up to 2 errors were introduced in transmission, then it can be corrected.

Suppose x is sent but x + e is received, where e corresponds to some vector with up to 2 non-zero components. We see that x + e is closer to x than any other
codeword! This is due to the fact that d(C) = 5.

E.g. let C = {00000,11111,22222} and 00000 is sent but 00012 is received. It is easy to see that 00000 is the closet codeword to 00012. So we decode 00012
as 00000, we have in effect corrected 2 errors. But if 3 or more errors are made and we decode using the closest codeword, then we might be in trouble. E.g.
if 11111 is sent but 11222 is received. We decode 11222 as 22222, but this is wrong!

No error correcting code is perfect (although we call some perfect codes). No code can correct every possible error vector. But it is also reasonable to assume
that only a small number of errors are made each transmission and so we only need codes that can correct a small number of errors.

Nearest Neighbour decoding

If m > n, then it is reasonable to assume that it is more likely that n errors were made than m errors. In any communication channel, it is reasonable to assume
that the more errors, the less likely. Therefore it is very reasonable to decode a received block using nearest neighbour decoding, i.e. if y is received, we look
for a codeword x (of C) so that d(x,y) is minimum.

Using the above scheme, it is easy to see that if a code C has minimum distance d(C) = 2t + 1, then up to t errors can be corrected.

Exercises

If a code C has minimum distance d(C) = 2t + 2. How many errors can it correct using nearest neighbour decoding?

Linear Codes

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Basic linear algebra assumed from here on.

Notation

Let and GF(q)n both denote the n-dimensional vector space with components coming from {1,2,..,q - 1} with arithmetic performed modulo q

A linear code C is a q-ary [n,k,d] code, if

1. C is a set of vectors of length n,

2. each component (of a codeword) takes a value from GF(q),
3. all the codewords of C are spanned by k linearly indepedent vectors, and
4. d(C) = d

Note that if x and y are codewords then so is x + y. In other words, we can think of C as a vector-subspace of , of dimension k. So C can be
completely determined by providing a basis of k vectors that span the codewords. Let {vi | i = 1, 2, ..., k} be a basis for C, we call the matrix

the generator matrix of C.

E.g. let C be a 5-ary [5,3,3] code spanned by {10034,01013,00111} then the generator matrix is

Information rate
A q-ary [n,k,d] code can have qk different codewords as each codeword c is of the form

c = a1v1 + a2v2 + .. + akvk

where ai may take values 0, 1, .., q - 1 (as arithmetic is performed modulo q). So this code can represent qk symobols.
We see that the row span of G are all the codewords, so suppose one wants to send c1c2..ck, we can compute the corresponding codeword c by

E.g. let C and G be as above and we wish to send 012 to the receiver, we compute the codeword

Notice how the first 3 digits of the codeword is actually the message we want to send, so the last 2 digits are not needed if we don't want any error-correction
ability in the code.

A linear code is in standard form if it's generator matrix is of the form (I|N) (has an identiy matrix at the left end of the generator matrix). The matrix G above
is in standard form. It turns out that if G is in standard then it's easy for the receiver to read the intended message. It has another advantage which will be
discussed in the next section.

Decoding

One advantage of using linear code is that detection of errors is easy. Actually, we can find a matrix H such that if and only if x is a codeword.
So if y is received and then we can confidently say that y has been contaminated by noise.

To find such a H, let's suppose C is a q-ary [n,k,d] code and C is spanned by vi i = 1, 2, .., k.

Definition -- Inner Product & Orthongonality

Define the inner product of any two vectors to be

we say two vectors v and w are orthogonal if <v,w> = 0.

E.g. let C be a 7-ary [7,4,3] code, then

<(0,1,2,4,5,6,3), (1,4,5,0,3,2,0)> = 4 + 10 + 15 + 12 = 41 ≡ 6 (mod 7)

Firstly note that H must be a n by j matrix for some j. Think of H as a linear transformation in GF(q)n. By definition kerH = C, and by the rank-nullity
theorom

n = dim(imH) + k

so H has rank n - k. In fact, the row span of H is the span of n - k linearly independent vectors, wi i = 1,2,..,n - k where wi are orthogonal to each codeword in

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C.

Notice that C = imH and kerH are vectors subspaces (exercise). In fact we denote

where means the vector subspace where any vector is orthogonal to every vector in C.

So we need to find a basis for and let H be the matrix whose rows the the basis vectors! If the generator matrix G of C is in standard form, then H is very
easy to compute. Indeed if

then

For example, let G be as above i.e.

then we can conclude

look at the values modulo 5 (recall that G generates a 5-ary code), we get

We call H the parity check matrix as H can tell us whether a code has been contamniated with.

To see that each codeword HxT = 0, all we need to do if multiply H by each row of G (transposed) since the rows of G span C (exercise).

Exercises

1. Let H be the parity check matrix of a code C, i.e. HxT = 0 for all codewords x of C. Think of H as a linear transformation. Prove that C = imH and kerH are
vectors subspaces.

2. If G (the generator matrix) is in standard form, prove that H as constructed above is spanned by all the vectors orthogonal to the row span of G.

Why Hamming code works

The binary Hamming code is a [7,4,3] code. Since the minimum distance is 3, so it can correct 1 error. The Hamming code is special as it tells the receiver
where the error is. To construct a Hamming code, it is easier to construct the parity check matrix first

We will not discuss how to find G, it is left as an exercise. Notice the columns of H are just the binary representation of the number 1,2,.., and 7 in increasing
order. This is how Hamming code can tell us where the error is.

Let x be a codeword of the Hamming Code, and suppose x + ej is received, where ej is the vector where only the jth position is 1. In other words, one error is
made in the jth position.

Now notice that

but is just the j column of H which is the binary representation of j. Network Security (HTTPS and SSN) WEP MD5 Checksum DES and RSA

Appendices

The 802 portion of the IEEE is responsible for publishing standards on a number of different protocols. Any chapter with a note such as (IEEE 802.3) is
refering to the group below that has published the currently accepted standard for that topic.

Working Group Task Status

802.1 Higher Layer LAN Protocols Working Group Active
802.2 Logical Link Control Working Group Inactive
802.3 Ethernet Working Group Active
802.4 Token Bus Working Group Disbanded

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802.5 Token Ring Working Group Inactive

802.6 Metropolitan Area Network Working Group Disbanded
802.7 Broadband TAG Disbanded
802.8 Fiber Optic TAG Disbanded
802.9 Isochronous LAN Working Group Disbanded
802.10 Security Working Group Disbanded
802.11 Wireless LAN Working Group Active
802.12 Demand Priority Working Group Inactive
802.14 Cable Modem Working Group Disbanded
802.15 Wireless Personal Area Network (WPAN) Working Group Active
802.16 Broadband Wireless Access Working Group Active
802.17 Resilient Packet Ring Working Group Active
802.18 Radio Regulatory TAG Active
802.19 Coexistence TAG Active
802.20 Mobile Broadband Wireless Access (MBWA) Working Group Active
802.21 Media Independent Handoff Working Group Active
802.22 Wireless Regional Area Networks Active

Communication Systems/Errata

Further Reading
Wikibooks

Communication Theory

Voice over IP

Internet Technologies

Networking:Ports and Protocols

Internet Engineering

Serial Communications Bookshelf

Analog and Digital Conversion

Wireless Mesh Networks

Wikipedia Articles

Wikipedia:History Of Internet

Books

Garcia and Widjaja, "Communication Networks: Fundamental Concepts and Key Architectures", Second Edition, McGraw Hill, 2004. ISBN
007246352X

Stern and Mahmoud, "Communication Systems: Analysis and Design", Prentice Hall, 2004. ISBN 0130402680

Silage, Dennis, "Digital Communication Systems Using SystemVue", DaVinci Engineering Press, 2006. ISBN 1584508507

Haykin and Moher, "Modern Wireless Communications", Prentice Hall, 2005. ISBN 0130224723

Gibson, Jerry D. "Principles of Digital and Analog Communications, Second Edition", Macmillan Publishing Company, 1989. ISBN 0023418605

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