UNIT III PART-A

Transcription

UNIT III PART-A

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UNIT 1
PART-A
1. What is communication and mention the three essential things necessary or any
communication?
Communication is the process of establishing connection (or link) between two points
for information exchange
2. Define signal to noise ratio?
Signal to noise power ratio is the ratio of the signal power level to the noise power
level
S/N = Ps/Pn
Where Ps = Signal power(watts)
Pn= Noise power(watts)
Signal to noise power ratio is often expressed as a logarithmic function with the
decimal unit
S/N(db) = 10log Ps/Pn
3. What are the three primary characteristics of a transmission line?
Wave velocity
Frequency
Wavelength
4. What is transmission line?
Transmission line is a metallic conductor system used to transfer electrical energy
from one point to another using electrical flow.
5. Mention the disadvantage of an unbalanced transmission line?
The primary disadvantage is its reduced immunity to common mode signals such as
noise and other interference.
6. Mention the disadvantage of an open wire transmission line?
There is no shielding so the radiation losses are high
The cable is susceptible to picking up signals through mutual induction, which
produces cross talk.
7. What are the different types of losses involved in a transmission line?
Conductor loss
Dielectric heating loss
Radiation loss
Coupling loss
Corona
8. What are standing waves?
With mismatched line there are two electromagnetic waves travelling in an opposite
direction, present on the line at the same time. These waves are in fact called
travelling waves. Two travelling wave setup an interference pattern known as
standing wave.
9. Define critical frequency?
Critical frequency is defined as the highest frequency that can be propagated directly
upward and still be returned to the earth by the ionosphere.
10. What is Maximum Usable Frequency?
MUF is the highest frequency that can be used for sky wave propagation between two
specific points on earth’s surface.
PART-B
1. What are the types of transmission line and their respective losses?
Transmission lines
Coaxial transmission line with one source and one load
Impedance bridging is unsuitable for RF connections, because it causes power to be
reflected back to the source from the boundary between the high and the low impedances.
The reflection creates a standing wave if there is reflection at both ends of the
transmission line, which leads to further power waste and may cause frequencydependent loss. In these systems, impedance matching is desirable.
In electrical systems involving transmission lines (such as radio and fiber optics)—where
the length of the line is long compared to the wavelength of the signal (the signal changes
rapidly compared to the time it takes to travel from source to load)— the impedances at
each end of the line must be matched to the transmission line's characteristic impedance (
) to prevent reflections of the signal at the ends of the line. (When the length of the
line is short compared to the wavelength, impedance mismatch is the basis of
transmission-line impedance transformers; see previous section.) In radio-frequency (RF)
systems, a common value for source and load impedances is 50 ohms. A typical RF load
is a quarter-wave ground plane antenna (37 ohms with an ideal ground plane; it can be
matched to 50 ohms by using a modified ground plane or a coaxial matching section, i.e.,
part or all the feeder of higher impedance).
The general form of the voltage reflection coefficient for a wave moving from medium 1
to medium 2 is given by
while the voltage reflection coefficient for a wave moving from medium 2 to medium 1 is
so the reflection coefficient is the same (except for sign), no matter from which direction
the wave approaches the boundary.
There is also a current reflection coefficient; it is the same as the voltage coefficient,
except that it has an opposite sign. If the wave encounters an open at the load end,
positive voltage and negative current pulses are transmitted back toward the source
(negative current means the current is going the opposite direction). Thus, at each
boundary there are four reflection coefficients (voltage and current on one side, and
voltage and current on the other side). All four are the same, except that two are positive
and two are negative. The voltage reflection coefficient and current reflection coefficient
on the same side have opposite signs. Voltage reflection coefficients on opposite sides of
the boundary have opposite signs.
Because they are all the same except for sign it is traditional to interpret the reflection
coefficient as the voltage reflection coefficient (unless otherwise indicated). Either end
(or both ends) of a transmission line can be a source or a load (or both), so there is no
inherent preference for which side of the boundary is medium 1 and which side is
medium 2. With a single transmission line it is customary to define the voltage reflection
coefficient for a wave incident on the boundary from the transmission line side,
regardless of whether a source or load is connected on the other side.
a)imbalanced transmission line
ii.unbalanced transmission line
b) Baluns
c) Metallic transmission line
i.parallel conductor
• Open wire
•twin lead
• Twisted pair
d) co-axial transmission line
Losses
a) Conductor loss
b) Dielectric heating loss
c) Radiatin loss
d) Coupling loss
e) Corona loss
2. Equivalent circuit of transmission line?
a)equivalent circuit for a single section transmission line terminated in a load equal to
Zo.
R-resistance
L-self inductance
c-capacitance
The four terminal model
For the purposes of analysis, an electrical transmission line can be modelled as a two-port
network (also called a quadrapole network), as follows:
In the simplest case, the network is assumed to be linear (i.e. the complex voltage across
either port is proportional to the complex current flowing into it when there are no
reflections), and the two ports are assumed to be interchangeable. If the transmission line
is uniform along its length, then its behaviour is largely described by a single parameter
called the characteristic impedance, symbol Z0. This is the ratio of the complex voltage of
a given wave to the complex current of the same wave at any point on the line. Typical
values of Z0 are 50 or 75 ohms for a coaxial cable, about 100 ohms for a twisted pair of
wires, and about 300 ohms for a common type of untwisted pair used in radio
transmission.
When sending power down a transmission line, it is usually desirable that as much power
as possible will be absorbed by the load and as little as possible will be reflected back to
the source. This can be ensured by making the load impedance equal to Z 0, in which case
the transmission line is said to be matched.
A transmission line is drawn as two black wires. At a distance x into the line, there is
current I(x) traveling through each wire, and there is a voltage difference V(x) between
the wires. If the current and voltage come from a single wave (with no reflection), then
V(x) / I(x) = Z0, where Z0 is the characteristic impedance of the line.
Some of the power that is fed into a transmission line is lost because of its resistance.
This effect is called ohmic or resistive loss. At high frequencies, another effect called
dielectric loss becomes significant, adding to the losses caused by resistance. Dielectric
loss is caused when the insulating material inside the transmission line absorbs energy
from the alternating electric field and converts it to heat. The transmission line is
modeled with a resistance (R) and inductance (L) in series with a capacitance (C) and
conductance (G) in parallel. The resistance and conductance contribute to the loss in a
transmission line.
The total loss of power in a transmission line is often specified in decibels per metre
(dB/m), and usually depends on the frequency of the signal. The manufacturer often
supplies a chart showing the loss in dB/m at a range of frequencies. A loss of 3 dB
corresponds approximately to a halving of the power.
High-frequency transmission lines can be defined as those designed to carry
electromagnetic waves whose wavelengths are shorter than or comparable to the length
of the line. Under these conditions, the approximations useful for calculations at lower
frequencies are no longer accurate.
Telegrapher's equations
The telegrapher's equations (or just telegraph equations) are a pair of linear
differential equations which describe the voltage and current on an electrical transmission
line with distance and time.
Schematic representation of the elementary component of a transmission line.
3. Explain in detail about transmission line model?
The transmission line model represents the transmission line as an infinite series of twoport elementary components, each representing an infinitesimally short segment of the
transmission line:
The distributed resistance
of the conductors is represented by a series resistor
(expressed in ohms per unit length).
The distributed inductance
(due to the magnetic field around the wires, self
inductance, etc.) is represented by a series inductor (henries per unit length).
The capacitance between the two conductors is represented by a shunt capacitor C
(farads per unit length).
The conductance
of the dielectric material separating the two conductors is
represented by a shunt resistor between the signal wire and the return wire (Siemens per
unit length).
The model consists of an infinite series of the elements shown in the figure, and that the
values of the components are specified per unit length so the picture of the component
can be misleading.
, , , and may also be functions of frequency. An alternative
notation is to use
,
,
and
to emphasize that the values are derivatives with
respect to length. These quantities can also be known as the primary line constants to
distinguish from the secondary line constants derived from them, these being the
propagation constant, attenuation constant and phase constant.
The line voltage
as
and the current
can be expressed in the frequency domain
When the elements and are negligibly small the transmission line is considered as a
lossless structure. In this hypothetical case, the model depends only on the and
elements which greatly simplifies the analysis. For a lossless transmission line, the
second order steady-state Telegrapher's equations are:
These are wave equations which have plane waves with equal propagation speed in the
forward and reverse directions as solutions. The physical significance of this is that
electromagnetic waves propagate down transmission lines and in general, there is a
reflected component that interferes with the original signal. These equations are
fundamental to transmission line theory.
If
and
are not neglected, the Telegrapher's equations become:
where
and the characteristic impedance is:
The solutions for
and
are:
The constants
pulse
and
, starting at
transmitted pulse
Transform,
must be determined from boundary conditions. For a voltage
and moving in the positive
at position
, of
-direction, then the
can be obtained by computing the Fourier
, attenuating each frequency component by
,
advancing its phase by
, and taking the inverse Fourier Transform. The real
and imaginary parts of can be computed as
For small losses and high frequencies, to first order in
Noting that an advance in phase by
can be simply computed as
and
is equivalent to a time delay by
one obtains
,
Input impedance of lossless transmission line
Looking towards a load through a length l of lossless transmission line, the impedance
changes as l increases, following the blue circle on this impedance smith chart. (This
impedance is characterized by its reflection coefficient Vreflected / Vincident.) The blue circle,
centered within the chart, is sometimes called an SWR circle (short for constant standing
wave ratio.
The characteristic impedance
of a transmission line is the ratio of the amplitude of a
single voltage wave to its current wave. Since most transmission lines also have a
reflected wave, the characteristic impedance is generally not the impedance that is
measured on the line.
For a lossless transmission line, it can be shown that the impedance measured at a given
position from the load impedance
is
where
is the wave number.
In calculating , the wavelength is generally different inside the transmission line to
what it would be in free-space and the velocity constant of the material the transmission
line is made of needs to be taken into account when doing such a calculation.
Special cases
Half wave length
For the special case where
where n is an integer (meaning that the length of
the line is a multiple of half a wavelength), the expression reduces to the load impedance
so that
for all . This includes the case when
, meaning that the length of the
transmission line is negligibly small compared to the wavelength. The physical
significance of this is that the transmission line can be ignored (i.e. treated as a wire) in
either case.
Quarter wave length
For the case where the length of the line is one quarter wavelength long, or an odd
multiple of a quarter wavelength long, the input impedance becomes
Matched load
Another special case is when the load impedance is equal to the characteristic impedance
of the line (i.e. the line is matched), in which case the impedance reduces to the
characteristic impedance of the line so that
for all and all
.
Short
Standing waves on a transmission line with an open-circuit load (top), and a short-circuit
load (bottom). Colors represent voltages, and black dots represent electrons.
For the case of a shorted load (i.e.
), the input impedance is purely imaginary
and a periodic function of position and wavelength (frequency)
Open
For the case of an open load (i.e.
imaginary and periodic
), the input impedance is once again
Stepped transmission line
A simple example of stepped transmission line consisting of three segments.
A stepped transmission line is used for broad range impedance matching. It can be
considered as multiple transmission line segments connected in series, with the
characteristic impedance of each individual element to be Z0,i. The input impedance can
be obtained from the successive application of the chain relation
where is the wave number of the ith transmission line segment and li is the length of
this segment, and Zi is the front-end impedance that loads the ith segment.
The impedance transformation circle along a transmission line whose characteristic
impedance Z0,i is smaller than that of the input cable Z0. And as a result, the impedance
curve is off-centered towards the -x axis. Conversely, if Z0,i > Z0, the impedance curve
should be off-centered towards the +x axis.
Because the characteristic impedance of each transmission line segment Z 0,i is often
different from that of the input cable Z 0, the impedance transformation circle is off
centered along the x axis of the Smith Chart whose impedance representation is usually
normalized against Z0.
Standing wave
When Zo =ZL all the incident power is absorbed by the load.this is called as matched
line.when Zo≠ZL some of the incident power is absorbed by the load,and some is
returned to the source.this is called an unmatched or mismatched line.with a mismatched
line,there are two electromagnetic waves,traveling in opposite direction,present on the
line at the same time.the two traveling waves set up an interference pattern known as a
standing wave.
4. Give short notes about ground wave propagation and space wave propogation?
Radio waves in the VLF Very low frequency band propagate in a ground, or surface
wave. The wave is confined between the surface of the earth and to the ionosphere. The
ground wave can propagate a considerable distance over the earth's surface and in the low
frequency and medium frequency portion of the radio spectrum. Ground wave radio
propagation is used to provide relatively local radio communications coverage, especially
by radio broadcast stations that require to cover a particular locality.
The radio waves having high frequencies are basically called as space waves. These
waves have the ability to propagate through atmosphere, from transmitter antenna to
receiver antenna. These waves can travel directly or can travel after reflecting from
earth’s surface to the troposphere surface of earth. So, it is also called as Tropospherical
Propagation. In the diagram of medium wave propagation, c shows the space wave
propagation. Basically the technique of space wave propagation is used in bands having
very high frequencies. E.g. V.H.F. band, U.H.F band etc
5. Give short notes about Maximum Usable Frequency?
In radio transmission maximum usable frequency (MUF) is the highest radio frequency
that can be used for transmission between two points via reflection from the ionosphere (
skywave or "skip" propagation) at a specified time, independent of transmitter power.
This index is especially useful in regard to short wave transmissions.
In short wave radio communication, a major mode of long distance propagation is for the
radio waves to reflect off the ionized layers of the atmosphere and return diagonally back
to Earth. In this way radio waves can travel beyond the horizon, around the curve of the
Earth. However the refractive index of the ionosphere decreases with increasing
frequency, so there is an upper limit to the frequency which can be used. Above this
frequency the radio waves are not reflected by the ionosphere but are transmitted through
it into space.
The ionization of the atmosphere varies with time of day and season as well as with solar
conditions, so the upper frequency limit for skywave communication varies on an hourly
basis. MUF is a median frequency, defined as the highest frequency at which skywave
communication is possible 50% of the days in a month, as opposed to the (LUF) which is
the frequency at which communication is possible 90% of the days, and the frequency o
optimum transmission (FOT).
Typically the MUF is a predicted number. Given the maximum observed frequency
(MOF) for a mode on each day of the month at a given hour, the MUF is the highest
frequency for which an ionospheric communications path is predicted on 50% of the days
of the month.
On a given day, communications may or may not succeed at the MUF. Commonly, the
optimal operating frequency for a given path is estimated at 80 to 90% of the MUF. As a
rule of thumb the MUF is approximately 3 times the critical frequency.
It is the highest frequency that can be used for sky wave propagation between specific two
points on earth surface.
Mathematically muf=critical frequency/
6.
Explain in detail about Transmission line impedance matching?
In electronics, impedance matching is the practice of designing the input impedance of an
electrical load (or the output impedance of its corresponding signal source) to maximize the
power transfer or minimize reflections from the load.
In the case of a complex source impedance ZS and load impedance ZL, maximum power
transfer is obtained when
where * indicates the complex conjugate. Minimum reflection is obtained when
Reflection-less matching
Impedance matching to minimize reflections is achieved by making the load impedance equal
to the source impedance. Ideally, the source and load impedances should be purely resistive:
in this special case reflection-less matching is the same as maximum power transfer matching.
A transmission line connecting the source and load together must also be the same impedance:
Zload = Zline = Zsource, where Zline is the characteristic impedance of the transmission line. The
transmission line characteristic impedance should also ideally be purely resistive. Cable
makers try to get as close to this ideal as possible and transmission lines are often assumed to
have a purely real characteristic impedance in calculations, however, it is conventional to still
use the term characteristic impedance rather than characteristic resistance.
Complex conjugate matching
Complex conjugate matching is used when maximum power transfer is required. This is
different from reflection-less matching only when the source or load have a reactive
component.
Zload = Zsource*
(where * indicates the complex conjugate).
If the source has a reactive component, but the load is purely resistive then matching can be
achieved by adding a reactance of the opposite sign to the load. This simple matching network
consisting of a single element will usually only achieve a perfect match at a single frequency.
This is because the added element will either be a capacitor or an inductor, both of which are
frequency dependent and will not, in general, follow the frequency dependence of the source
impedance. For wide bandwidth applications a more complex network needs to be designed.
Power transfer
Whenever a source of power with a fixed output impedance such as an electrical signal source,
a radio transmitter or a mechanical sound (e.g., a loudspeaker) operates into a load, the
maximum possible power is delivered to the load when the impedance of the load (load
impedance or input impedance) is equal to the complex conjugate of the impedance of the
source (that is, its internal impedance or output impedance). For two impedances to be
complex conjugates their resistances must be equal, and their reactances must be equal in
magnitude but of opposite signs. In low-frequency or DC systems (or systems with purely
resistive sources and loads) the reactances are zero, or small enough to be ignored. In this
case, maximum power transfer occurs when the resistance of the load is equal to the resistance
of the source.
Impedance matching is not always necessary. For example, if a source with a low impedance
is connected to a load with a high impedance the power that can pass through the connection
is limited by the higher impedance. This maximum-voltage connection is a common
configuration called impedance bridging or voltage bridging, and is widely used in signal
processing. In such applications, delivering a high voltage (to minimize signal degradation
during transmission or to consume less power by reducing currents) is often more important
than maximum power transfer.
In older audio systems (reliant on transformers and passive filter networks, and based on the
telephone system), the source and load resistances were matched at 600 ohms. One reason for
this was to maximize power transfer, as there were no amplifiers available that could restore
lost signal. Another reason was to ensure correct operation of the hybrid transformers used at
central exchange equipment to separate outgoing from incoming speech, so these could be
amplified or fed to a four-wire circuit. Most modern audio circuits, on the other hand, use
active amplification and filtering and can use voltage-bridging connections for greatest
accuracy. Strictly speaking, impedance matching only applies when both source and load
devices are linear; however, matching may be obtained between nonlinear devices within
certain operating ranges.
Impedance-matching devices
Adjusting the source impedance or the load impedance, in general, is called "impedance
matching". There are three ways to improve an impedance mismatch, all of which are called
"impedance matching":
Devices intended to present an apparent load to the source of Zload = Zsource* (complex
conjugate matching). Given a source with a fixed voltage and fixed source impedance,
the maximum power theorem says this is the only way to extract the maximum power
from the source.
Devices intended to present an apparent load of Zload = Zline (complex impedance
matching), to avoid echoes. Given a transmission line source with a fixed source
impedance, this "reflectionless impedance matching" at the end of the transmission
line is the only way to avoid reflecting echoes back to the transmission line.
Devices intended to present an apparent source resistance as close to zero as possible,
or presenting an apparent source voltage as high as possible. This is the only way to
maximize energy efficiency, and so it is used at the beginning of electrical power
lines. Such an impedance bridging connection also minimizes distortion and
electromagnetic interference; it is also used in modern audio amplifiers and signalprocessing devices.
There are a variety of devices used between a source of energy and a load that perform
"impedance matching". To match electrical impedances, engineers use combinations of
transformers, resistors, inductors, capacitors and transmission lines. These passive (and active)
impedance-matching devices are optimized for different applications and include baluns,
antenna tuners (sometimes called ATUs or roller-coasters, because of their appearance),
acoustic horns, matching networks, and terminators.
Transformers
Transformers are sometimes used to match the impedances of circuits. A transformer converts
alternating current at one voltage to the same waveform at another voltage. The power input to
the transformer and output from the transformer is the same (except for conversion losses).
The side with the lower voltage is at low impedance (because this has the lower number of
turns), and the side with the higher voltage is at a higher impedance (as it has more turns in its
coil).
One example of this method involves a television balun transformer. This transformer
converts a balanced signal from the antenna (via 300-ohm twin-lead) into an unbalanced
signal (75-ohm coaxial cable such as RG-6). To match the impedances of both devices, both
cables must be connected to a matching transformer with a turns ratio of 2 (such as a 2:1
transformer). In this example, the 75-ohm cable is connected to the transformer side with
fewer turns; the 300-ohm line is connected to the transformer side with more turns. The
formula for calculating the transformer turns ratio for this example is:
Resistive network
Resistive impedance matches are easiest to design and can be achieved with a simple L pad
consisting of two resistors. Power loss is an unavoidable consequence of using resistive
networks, and they are only (usually) used to transfer line level signals.
Stepped transmission line
Most lumped-element devices can match a specific range of load impedances. For example, in
order to match an inductive load into a real impedance, a capacitor needs to be used. If the
load impedance becomes capacitive, the matching element must be replaced by an inductor. In
many cases, there is a need to use the same circuit to match a broad range of load impedance
and thus simplify the circuit design. This issue was addressed by the stepped transmission
line, where multiple, serially placed, quarter-wave dielectric slugs are used to vary a
transmission line's characteristic impedance. By controlling the position of each element, a
broad range of load impedances can be matched without having to reconnect the circuit.
Filters
Filters are frequently used to achieve impedance matching in telecommunications and radio
engineering. In general, it is not theoretically possible to achieve perfect impedance matching
at all frequencies with a network of discrete components. Impedance matching networks are
designed with a definite bandwidth, take the form of a filter, and use filter theory in their
design.
Applications requiring only a narrow bandwidth, such as radio tuners and transmitters, might
use a simple tuned filter such as a stub. This would provide a perfect match at one specific
frequency only. Wide bandwidth matching requires filters with multiple sections.
L-section
L networks for narrowband matching a source or load impedance Z to a transmission line with
characteristic impedance Z0. X and B may each be either positive (inductor) or negative
(capacitor). If Z/Z0 is inside the 1+jx circle on the Smith chart (i.e. if Re(Z/Z0)>1), network (a)
can be used; otherwise network (b) can be used.
A simple electrical impedance-matching network requires one capacitor and one inductor.
One reactance is in parallel with the source (or load), and the other is in series with the load
(or source). If a reactance is in parallel with the source, the effective network matches from
high to low impedance. The L-section is inherently a narrow band matching network.
The analysis is as follows. Consider a real source impedance of
and real load impedance
of
. If a reactance
is in parallel with the source impedance, the combined impedance
can be written as:
If the imaginary part of the above impedance is canceled by the series reactance, the real part
is
Solving for
If
the above equation can be approximated as
The inverse connection (impedance step-up) is simply the reverse—for example, reactance in
series with the source. The magnitude of the impedance ratio is limited by reactance losses
such as the Q of the inductor. Multiple L-sections can be wired in cascade to achieve higher
impedance ratios or greater bandwidth. Transmission line matching networks can be modeled
as infinitely many L-sections wired in cascade. Optimal matching circuits can be designed for
a particular system using smith charts.
Power factor correction
These devices are intended to cancel the reactive and nonlinear characteristics of a load at the
end of a power line. This causes the load seen by the power line to be purely resistive. For a
given true power required by a load this minimizes the true current supplied through the
power lines, and minimizes power wasted in the resistance of those power lines. For example,
a maximum power point tracker is used to extract the maximum power from a solar panel and
efficiently transfer it to batteries, the power grid or other loads. The maximum power theorem
applies to its "upstream" connection to the solar panel, so it emulates a load resistance equal to
the solar panel source resistance. However, the maximum power theorem does not apply to its
"downstream" connection. That connection is an impedance bridging connection; it emulates a
high-voltage, low-resistance source to maximize efficiency.
On the power grid the overall load is usually inductive. Consequently, power factor
correction is most commonly achieved with banks of capacitors. It is only necessary for
correction to be achieved at one single frequency, the frequency of the supply. Complex
networks are only required when a band of frequencies must be matched and this is the reason
why simple capacitors are all that is usually required for power factor correction.
UNIT-2
PART-A
1. What is modulation?
Modulation is the process of placing the message signal over some carrier to make it
suitable for transmission over long distances.
2. What is demodulation?
Demodulation is the process of separating the message signal from the modulated
carrier signal.
3. What is multiplexing?
Multiplexing is the process of simultaneously transmitting two or more individual
signals over a single common channel.
Due to multiplexing it is possible to increase the number of communication channels
so that more information can be transmitted.
4. What are the two basic types of multiplexing?
Frequency division multiplexing
Time division multiplexing
In FDM many signals are transmitted simultaneously where each signal occupies a
different frequency slot within a common bandwidth.
In TDM the signals are transmitted at a time, instead they are transmitted in different
time slots.
5. Advantages of FDM?
A large number of signals (Channels) can be transmitted simultaneously
FDM does not need synchronization between its transmitter and receiver for proper
operation
Demodulation of FDM is easy
Due to slow narrow band fading only a single channel gets affected
6. What is Intersymbol Interference (ISI)?
In a communication system when the data is being transmitted in the form of pulses
(bits), the output produced at the receiver due to the other bits or symbols interferes
with the output produced by the desired bit. This is called as intersymbol
interference(ISI). The Intersymbol interference will produce errors in the detected
signal.
7. Advantages of TDM?
Full available channel bandwidth can be utilized for each channel
Intermodulation distortion is absent
TDM circuitry is not very complex
The problem of crosstalk is not severe
8. Disadvantages of TDM?
Synchronization is essential for proper operation
Due to slow narrowband fading, all the TDM channels may get wiped out
9. What is guard bands and mention their importance?
The adjacent spectrums in the spectrum of an FDM signal do not touch each other.
They are separated from each other by the guardbands.
The guard bands are introduced in order to avoid any interference between the
adjacent channels
Wider the guardband, smaller the interference
10. Applications of FDM?
Telephone systems
AM (amplitude modulation) and FM(frequency modulation) radio broadcasting TV
broadcasting and also first generation of cellular phones used FDM.
PART-B
1.
Explain in detail about Amplitude modulation
Amplitude modulation (AM) is a technique used in electronic communication, most commonly
for transmitting information via a radio carrier wave. AM works by varying the strength of the
transmitted signal in relation to the information being sent. For example, changes in signal strength
may be used to specify the sounds to be reproduced by a loudspeaker, or the light intensity of
television pixels. Contrast this with Frequency modulation, in which the frequency is varied, and
phase modulation, in which the phase is varied in accordance to the modulating signal.
In the mid-1870s, a form of amplitude modulation—initially called "undulatory currents"—was the
first method to successfully produce quality audio over telephone lines. Beginning with Reginald
Fessenden’s audio demonstrations in 1906, it was also the original method used for audio radio
transmissions, and remains in use today by many forms of communication—"AM" is often used to
refer to the medium wave broadcast band.
Fig : An audio signal (top) may be carried by an AM or FMradio wave.
Forms of amplitude modulation
In radio communication, a continuous wave radio-frequency signal (a sinusoidal carrier wave) has
its amplitude modulated by an audio waveform before transmission. The audio waveform modifies
the amplitude of the carrier wave and determines the envelope of the waveform. In the frequency
domain, amplitude modulation produces a signal with power concentrated at the carrier frequency
and two adjacent sidebands. Each sideband is equal in bandwidth to that of the modulating signal,
and is a mirror image of the other. Amplitude modulation resulting in two sidebands and a carrier
is called "double-sideband amplitude modulation" (DSB-AM).
Amplitude modulation is inefficient in power usage; at least two-thirds of the power is
concentrated in the carrier signal. The carrier signal contains none of the original information being
transmitted (voice, video, data, etc.). However, it does contain information about the frequency,
phase and amplitude needed to demodulate the received signal most simply and effectively. In
some communications systems, lower total cost can be achieved by eliminating some of the carrier,
thereby lowering electrical power usage even though this requires greater receiver complexity and
cost. If some carrier is retained (reduced-carrier transmission, or DSB-RC) receivers can be
designed to recover the frequency, phase, and amplitude information from this "pilot" carrier and
use it in the demodulation process. If the carrier is eliminated (Double sideband suppressed-carrier
transmission or DSB-SC) the receiver must provide a substitute carrier, with inevitable loss of
fidelity. Completely suppressing both the carrier and one of the sidebands produces singlesideband modulation, widely used in amateur radio and other communications applications. SSB
occupies less than half the spectrum of AM so it also has greatly improved bandwidth efficiency.
In AM broadcasting, where there are many receivers for each transmitter, the full carrier is
provided to allow reception with inexpensive receivers. The broadcaster absorbs the extra power
cost to greatly increase potential audience.
A simple form of AM, often used for digital communications, is on-off keying: a type of
amplitude-shift keying in which binary data is represented by the presence or absence of a carrier.
This is used by radio amateurs to transmit Morse code and is known as Continuous wave (CW)
operation.
ITU designations
In 1982, the International Telecommunication Union (ITU) designated the types of amplitude
modulation:
Example: double-sideband AM
Left part: Modulation Voltage as a function of time. Right part: Spectrum of the amplitude
modulated carrier
Fig : Double-sided spectra of baseband and AM signals.
A carrier wave is modeled as a sine wave:
in which the frequency in Hz is given by:
The constants and
represent the carrier amplitude and initial phase, and are introduced for
generality. For simplicity, their respective values can be set to 1 and 0.
Let m(t) represent an arbitrary waveform that is the message to be transmitted, e.g., a simple audio
tone of form:
where constant M represent the largest magnitude, and the frequency is:
It is assumed that
and that
Amplitude modulation is formed by the product:
represents the carrier amplitude. The values A=1 and M=0.5 produce y(t), depicted by the top
graph (labelled "50% Modulation") in Figure 4.
Using trigonometric identities, y(t) can be written in the form
Therefore, the modulated signal has three components: a carrier wave and two sinusoidal waves
(known as sidebands), whose frequencies are slightly above and below
Spectrum
For more general forms of m(t), trigonometry is not sufficient; however, if the top trace of Figure 2
depicts the frequency of m(t) the bottom trace depicts the modulated carrier. It has two
components: one at a positive frequency (centeredon
) and one at a negative frequency
(centered on
). Each component contains the two sidebands and a narrow segment in
between, representing energy at the carrier frequency. Since the negative frequency is a
mathematical artifact, examining the positive frequency demonstrates that an AM signal's spectrum
consists of its original (two-sided) spectrum, shifted to the carrier frequency. Figure 2 is a result of
computing the Fourier transform of:
pairs:
using the following transform
Figure: The spectrogram of an AM broadcast shows its two sidebands (green), separated by the
carrier signal (red).
Power and spectrum efficiency
In terms of positive frequencies, the transmission bandwidth of AM is twice the signal's original
(baseband) bandwidth; both the positive and negative sidebands are shifted up to the carrier
frequency. Thus, double-sideband AM (DSB-AM) is spectrally inefficient because the same
spectral information is transmitted twice, and fewer radio stations can be accommodated in a given
broadcast band than if only one replica of the original signal's spectrum were transmitted. The
suppression methods described above may be understood in terms of Figure 2. With the carrier
suppressed, there would be no energy at the center of a group; with a sideband suppressed, the
"group" would have the same bandwidth as the positive frequencies of
The transmitterpower efficiency of DSB-AM depends on the type of receiver used. For the most inexpensive type
of AM receiver, the carrier is needed to provide undistorted reception, thus 100% of the power is
useful. With a single sideband suppressed carrier (SSB-SC) capable receiver, only 16.7% of the
transmitted power is useful, since 66.6% of the power is wasted in the carrier and 16.7% in the
unused sideband. DSB-SC systems have had very limited application but would theoretically use
33.3% of the transmitted signal.
Modulation index
The AM modulation index is the measure of the amplitude variation surrounding an unmodulated
carrier. As with other modulation indices, in AM this quantity (also called "modulation depth")
indicates how much the modulation varies around its unmodulated level. For AM, it relates to
variations in carrier amplitude and is defined as:
where
and are the message amplitude and carrier amplitude, respectively, and where the
message amplitude is the maximum change in the carrier amplitude, measured from its
unmodulated value.
So if
, carrier amplitude varies by 50% above (and below) its unmodulated level; for
, it varies by 100%. To avoid distortion, modulation depth must not exceed 100 percent.
Transmitter systems will usually incorporate a limiter circuit (such as a vogad) to ensure this.
However, AM demodulators can be designed to detect the inversion (or 180-degree phase reversal)
that occurs when modulation exceeds 100 percent; they automatically correct for this defect.
Variations of a modulated signal with percentages of modulation are shown below. In each image,
the maximum amplitude is higher than in the previous image (note that the scal (e changes from
one
image
to
the
next).
Fig : Modulation depth
Modulation methods
Anode (plate) modulation. A tetrode's plate and screen grid voltage is modulated via an audio
transformer. The resistor R1 sets the grid bias; both the input and output are tuned circuits with
inductive coupling.
Modulation circuit designs may be classified as low- or high-level (depending on whether they
modulate in a low-power domain—followed by amplification for transmission—or in the highpower domain of the transmitted signal)
Low-level generation
In modern radio systems, modulated signals are generated via digital signal processing (DSP).
With DSP many types of AM are possible with software control (including DSB with carrier, SSB
suppressed-carrier and independent sideband, or ISB). Calculated digital samples are converted to
voltages with a digital to analog converter, typically at a frequency less than the desired RF-output
frequency. The analog signal must then be shifted in frequency and linearly amplified to the
desired frequency and power level (linear amplification must be used to prevent modulation
distortion). This low-level method for AM is used in many Amateur Radio transceivers.
AM may also be generated at a low level, using analog methods described in the next section.
High-level generation
High-power AM transmitters (such as those used for AM broadcasting) are based on highefficiency class-D and class-E power amplifier stages, modulated by varying the supply
voltage.Older designs (for broadcast and amateur radio) also generate AM by controlling the gain
of the transmitter’s final amplifier (generally class-C, for efficiency). The following types are for
vacuum tube transmitters (but similar options are available with transistors):Plate modulation: In
plate modulation, the plate voltage of the RF amplifier is modulated with the audio signal. The
audio power requirement is 50 percent of the RF-carrier power.
Heising (constant-current) modulation: RF amplifier plate voltage is fed through a ―choke‖
(high-value inductor). The AM modulation tube plate is fed through the same inductor, so the
modulator tube diverts current from the RF amplifier. The choke acts as a constant current source
in the audio range. This system has a low power efficiency.
Control grid modulation: The operating bias and gain of the final RF amplifier can be
controlled by varying the voltage of the control grid. This method requires little audio power, but
care must be taken to reduce distortion.
Clamp tube (screen grid) modulation: The screen-grid bias may be controlled through a
―clamp tube‖, which reduces voltage according to the modulation signal. It is difficult to approach
100-percent modulation while maintaining low distortion with this system.
Doherty modulation:One tube provides the power under carrier conditions and another
operates only for positive modulation peaks. Overall efficiency is good, and distortion is low.
Outphasing modulation:Two tubes are operated in parallel, but partially out of phase with
each other. As they are differentially phase modulated their combined amplitude is greater or
smaller. Efficiency is good and distortion low when properly adjusted.
Pulse width modulation (PWM) or Pulse duration modulation (PDM):A highly efficient
high voltage power supply is applied to the tube plate. The output voltage of this supply is varied at
an audio rate to follow the program. This system was pioneered by Hilmer Swanson and has a
number of variations, all of which achieve high efficiency and sound quality.
2.
Explain in detail about Amplitude demodulation or detection?
In order to look at the amplitude demodulation process it is necessary to first look at the format of
an AM signal.An AM signal consists of a carrier which acts as the reference. Any modulation that
is applied then appears as sidebands which stretch out either side of the signal - each sideband is a
mirror image of the other.
Spectrum of an amplitude modulated, AM signal
Within the overall AM signal the carrier possess the majority of the power - a fully modulated, i.e.
100% modulation - AM signal has sidebands which have 25% that of the main carrier.
When demodulating a signal, two basic steps may be considered:
Create baseband signal: The main element of AM demodulation is to create the baseband
signal. This can be achieved in a number of ways - one of the easiest is to use a simple diode and
rectify the signal. This leaves elements of the original RF signal. When other forms of
demodulation are sued, they too leave some elements of an RF signal.
Filter: The filtering removes any unwanted high frequency elements from the demodulation
process. The audio can then be presented to further stages for audio amplification, etc.
The AM demodulation process is outlined in the diagram below. This particular example applies
particularly to a diode detector.
Basics of AM demodulation / detection
Types of AM demodulator
There are a number of ways in which an AM signal can be demodulated. There is a balance that
needs to be made of the performance of the circuit that is required against the complexity, and
hence the cost that can be tolerated.
The major types of AM demodulator are:
Diode detector: This is by far the simplest form of AM demodulator or detector, requiring just a
semiconductor (or other form) of diode along with a capacitor to remove the high frequency
components. It suffers from a number of disadvantages, but its performance is more than adequate
for most applications including broadcast receivers where cost is a significant driver.
Synchronous detector: This form of detector offers a higher level of performance, but at the cost
of considerably the use of considerably more components. This means that it is only used in
receivers where the levels of performance are paramount and can justify the additional component
costs.
Both types of detector are widely used, although the diode detector is far more common in view of
its simplicity and the fact that it is quite adequate for broadcast applications where performance is
not normally an issue.
AM demodulation & detection includes:
One of the advantages of amplitude modulation (AM) is that it is cheap and easy to build a
demodulator circuit for a radio receiver. The simplicity AM radio receivers AM is one of the
reasons why AM has remained in service for broadcasting for so long. One of the key factors of
this is the simplicity of the receiver AM demodulator.
A number of methods can be used to demodulate AM, but the simplest is a diode detector. It
operates by detecting the envelope of the incoming signal. It achieves this by simply rectifying the
signal. Current is allowed to flow through the diode in only one direction, giving either the positive
or negative half of the envelope at the output. If the detector is to be used only for detection it does
not matter which half of the envelope is used, either will work equally well. Only when the
detector is also used to supply the automatic gain control (AGC) circuitry will the polarity of the
diode matter.
The AM detector or demodulator includes a capacitor at the output. Its purpose is to remove any
radio frequency components of the signal at the output. The value is chosen so that it does not
affect the audio base-band signal. There is also a leakage path to enable the capacitor to discharge,
but this may be provided by the circuit into which the demodulator is connected.
A simple diode detector (demodulator) for AM signals
This type of detector or demodulator is called a linear envelope detector because the output is
proportional to the input envelope. Unfortunately the diodes used can introduce appreciable levels
of harmonic distortion unless modulation levels are kept low. As a result these detectors can never
provide a signal suitable for high quality applications.
Additionally these detectors are susceptible to the effects of selective fading experienced on short
wave broadcast transmissions. Here the ionospheric propagation may be such that certain small
bands of the signal are removed. Under normal circumstances signals received via the ionosphere
reach the receiver via a number of different paths. The overall signal is a combination of the
signals received via each path and as a result they will combine with each other, sometimes
constructively to increase the overall signal level and sometimes destructively to reduce it. It is
found that when the path lengths are considerably different this combination process can mean that
small portions of the signal are reduced in strength. An AM signal consists of a carrier with two
sidebands.
If the section of the signal that is removed falls in one of the sidebands, it will change the tone of
the received signal. However if carrier is removed or even reduced in strength, the signal will
appear to be over modulated, and severe distortion will result. This is a comparatively common
occurrence on the short waves, and means that diode detectors are not suitable for high quality
reception. Synchronous demodulation (detection) is far superior.
3.
Explain in detail about Frequency modulation?
A signal may be carried by an AM or FM radio wave.
In telecommunications and signal processing, frequency modulation (FM) is the encoding of
information in a carrier wave by varying the instantaneous frequency of the wave. (Compare with
amplitude modulation, in which the amplitude of the carrier wave varies, while the frequency
remains constant.)
In analog signal applications, the difference between the instantaneous and the base frequency of
the carrier is directly proportional to the instantaneous value of the input-signal amplitude.
Digital data can be encoded and transmitted via carrier wave by shifting the carrier's frequency
among a predefined set of frequencies—a technique known as frequency-shift keying (FSK). FSK
is widely used in modems and fax modems, and can also be used to send Morse code. Radio
teletype also uses FSK.Frequency modulation is used in radio, telemetry, radar, seismic
prospecting, and monitoring newbornsfor seizures via EEG. FM is widely used for broadcasting
music and speech, two-way radio systems, magnetic tape-recording systems and some videotransmission systems. In radio systems, frequency modulation with sufficient bandwidth provides
an advantage in cancelling naturally-occurring noise.
Frequency modulation is known as phase modulation when the carrier phase modulation is the time
integral of the FM signal.
Theory
If the information to be transmitted (i.e., the baseband signal) is
and the sinusoidal carrier
is
, where fc is the carrier's base frequency, and Ac is the carrier's
amplitude, the modulator combines the carrier with the baseband data signal to get the transmitted
signal:
In this equation,
is the instantaneous frequency of the oscillator and
is the frequency
deviation, which represents the maximum shift away from fc in one direction, assuming xm(t) is
limited to the range ±1.
While most of the energy of the signal is contained within fc ± fΔ, it can be shown by Fourier
analysis that a wider range of frequencies is required to precisely represent an FM signal. The
frequency spectrum of an actual FM signal has components extending infinitely, although their
amplitude decreases and higher-order components are often neglected in practical design problems.
Sinusoidal baseband signal
Mathematically, a baseband modulated signal may be approximated by a sinusoidal continuous
wave signal with a frequency fm. The integral of such a signal is:
In this case, equation (1) above simplifies to:
where the amplitude
of the modulating sinusoid is represented by the peak deviation
.
The harmonic distribution of a sine wave carrier modulated by such a sinusoidal signal can be
represented with Bessel functions; this provides the basis for a mathematical understanding of
frequency modulation in the frequency domain.
Modulation index
As in other modulation systems, this quantity indicates by how much the modulated variable varies
around its unmodulated level. It relates to variations in the carrier frequency:
where
is the highest frequency component present in the modulating signal xm(t), and
is
the peak frequency-deviation—i.e. the maximum deviation of the instantaneous frequency from the
carrier frequency. If
, the modulation is called narrowband FM, and its bandwidth is
approximately
.
If
, the modulation is called wideband FM and its bandwidth is approximately
.
While wideband FM uses more bandwidth, it can improve the signal-to-noise ratio significantly;
for example, doubling the value of
, while keeping
constant, results in an eight-fold
improvement in the signal-to-noise ratio. (Compare this with Chirp spread spectrum, which uses
extremely wide frequency deviations to achieve processing gains comparable to traditional, betterknown spread-spectrum modes).
With a tone-modulated FM wave, if the modulation frequency is held constant and the modulation
index is increased, the (non-negligible) bandwidth of the FM signal increases but the spacing
between spectra remains the same; some spectral components decrease in strength as others
increase. If the frequency deviation is held constant and the modulation frequency increased, the
spacing between spectra increases.
Frequency modulation can be classified as narrowband if the change in the carrier frequency is
about the same as the signal frequency, or as wideband if the change in the carrier frequency is
much higher (modulation index >1) than the signal frequency. For example, narrowband FM is
used for two way radio systems such as Family Radio Service, in which the carrier is allowed to
deviate only 2.5 kHz above and below the center frequency with speech signals of no more than
3.5 kHz bandwidth. Wideband FM is used for FM broadcasting, in which music and speech are
transmitted with up to 75 kHz deviation from the center frequency and carry audio with up to a 20kHz bandwidth.
Bessel functions
For the case of a carrier modulated by a single sine wave, the resulting frequency spectrum can be
calculated using Bessel functions of the first kind, as a function of the sideband number and the
modulation index. The carrier and sideband amplitudes are illustrated for different modulation
indices of FM signals. For particular values of the modulation index, the carrier amplitude becomes
zero and all the signal power is in the sidebands.
Since the sidebands are on both sides of the carrier, their count is doubled, and then multiplied by
the modulating frequency to find the bandwidth. For example, 3 kHz deviation modulated by a 2.2
kHz audio tone produces a modulation index of 1.36. Examining the chart shows this modulation
index will produce three sidebands. These three sidebands, when doubled, gives us (6 * 2.2 kHz) or
a 13.2 kHz required bandwidth.
Carson's rule
A rule of thumb, Carson's rule states that nearly all (~98 percent) of the power of a frequencymodulated signal lies within a bandwidth
of:
where
, as defined above, is the peak deviation of the instantaneous frequency
center carrier frequency .
from the
Noise Reduction
A major advantage of FM in a communications circuit, compared for example with AM, is the
possibility of improved (SNR). Compared with an optimum AM scheme, FM typically has poorer
SNR below a certain signal level called the noise threshold, but above a higher level – the full
improvement or full quieting threshold – the SNR is much improved over AM. The improvement
depends on modulation level and deviation. For typical voice communications channels,
improvements are typically 5-15 dB. FM broadcasting using wider deviation can achieve even
greater improvements. Additional techniques, such as pre-emphasis of higher audio frequencies
with corresponding de-emphasis in the receiver, are generally used to improve overall SNR in FM
circuits. Since FM signals have constant amplitude, FM receivers normally have limiters that
remove AM noise, further improving SNR.
Implementation
Modulation
FM signals can be generated using either direct or indirect frequency modulation:
Direct FM modulation can be achieved by directly feeding the message into the input of a VCO.
For indirect FM modulation, the message signal is integrated to generate a phase modulated signal.
This is used to modulate a crystal-controlled oscillator, and the result is passed through a frequency
multiplier to give an FM signal.
Demodulation
Many FM detector circuits exist. A common method for recovering the information signal is
through a Foster-Seeley discriminator. A phase-locked loop can be used as an FM demodulator.
Slope detection demodulates an FM signal by using a tuned circuit which has its resonant
frequency slightly offset from the carrier. As the frequency rises and falls the tuned circuit provides
a changing amplitude of response, converting FM to AM. AM receivers may detect some FM
transmissions by this means, although it does not provide an efficient means of detection for FM
broadcasts.
Applications
Magnetic tape storage
FM is also used at intermediate frequencies by analog VCR systems (including VHS) to record the
luminance (black and white) portions of the video signal. Commonly, the chrominance component
is recorded as a conventional AM signal, using the higher-frequency FM signal as bias. FM is the
only feasible method of recording the luminance ("black and white") component of video to (and
retrieving video from) magnetic tape without distortion; video signals have a large range of
frequency components – from a few hertz to several megahertz, too wide for equalizers to work
with due to electronic noise below −60 dB. FM also keeps the tape at saturation level, acting as a
form of noise reduction; a limiter can mask variations in playback output, and the FM capture
effect removes print-through and pre-echo. A continuous pilot-tone, if added to the signal – as was
done on V2000 and many Hi-band formats – can keep mechanical jitter under control and assist
timebase correction. These FM systems are unusual, in that they have a ratio of carrier to
maximum modulation frequency of less than two; contrast this with FM audio broadcasting, where
the ratio is around 10,000. Consider, for example, a 6-MHz carrier modulated at a 3.5-MHz rate;
by Bessel analysis, the first sidebands are on 9.5 and 2.5 MHz and the second sidebands are on
13 MHz and −1 MHz. The result is a reversed-phase sideband on +1 MHz; on demodulation, this
results in unwanted output at 6−1 = 5 MHz. The system must be designed so that this unwanted
output is reduced to an acceptable level.
4.
Explain in detail about FM Demodulation / Detection?
FM demodulation or detection involves changing the frequency variations in a signal into
amplitude variations at baseband, e.g. audio. There are several techniques and circuits that
can be used each with its own advantages and disadvantages.
Frequency modulation is widely used for radio transmissions for a wide variety of
applications from broadcasting to general point to point communications.
Frequency modulation, FM offers many advantages, particularly in mobile radio applications
where its resistance to fading and interference is a great advantage. It is also widely used for
broadcasting on VHF frequencies where it is able to provide a medium for high quality audio
transmissions.
In view of its widespread use receivers need to be able to demodulate these transmissions.
There is a wide variety of different techniques and circuits that can be used including the
Foster-Seeley, and ratio detectors using discreet components, and where integrated circuits are
used the phase locked loop and quadrature detectors are more widely used.
What is frequency modulation, FM?
As the name suggests frequency modulation, FM uses changes in frequency to carry the sound
or other information that is required to be placed onto the carrier. As shown below it can be
seen that as the modulating or base band signal voltage varies, so the frequency of the signal
changes in line with it. This type of modulation brings several advantages with it:
Interference reduction: When compared to AM, FM offers a marked improvement
in interference. In view of the fact that most received noise is amplitude noise, an FM
receiver can remove any amplitude sensitivity by driving the IF into limiting.
Removal of many effects of signal strength variations: FM is widely used for
mobile applications because the amplitude variations do not cause a change in audio
level. As the audio is carried by frequency variations rather than amplitude ones,
under good signal strength conditions, this does not manifest itself as a change in
audio level.
Transmitter amplifier efficiency: As the modulation is carried by frequency
variations, this means that the transmitter power amplifiers can be made non-linear.
These amplifiers can be made to be far more efficient than linear ones, thereby saving
valuable battery power - a valuable commodity for mobile or portable equipment.
These advantages mean that FM has been widely used for many broadcasting and mobile
applications.
Frequency modulating a signal
Wide band and Narrow band FM
When a signal is frequency modulated, the carrier shifts in frequency in line with the
modulation. This is called the deviation. In the same way that the modulation level can be
varied for an amplitude modulated signal, the same is true for a frequency modulated one,
although there is not a maximum or 100% modulation level as in the case of AM.
The level of modulation is governed by a number of factors. The bandwidth that is available is
one. It is also found that signals with a large deviation are able to support higher quality
transmissions although they naturally occupy a greater bandwidth. As a result of these
conflicting requirements different levels of deviation are used according to the application that
is used.
Those with low levels of deviation are called narrow band frequency modulation (NBFM) and
typically levels of +/- 3 kHz or more are used dependent upon the bandwidth available.
Generally NBFM is used for point to point communications. Much higher levels of deviation
are used for broadcasting. This is called wide band FM (WBFM) and for broadcasting
deviation of +/- 75 kHz is used.
Receiving FM
In order to be able to receive FM a receiver must be sensitive to the frequency variations of
the incoming signals. As already mentioned these may be wide or narrow band. However the
set is made insensitive to the amplitude variations. This is achieved by having a high gain IF
amplifier. Here the signals are amplified to such a degree that the amplifier runs into limiting.
In this way any amplitude variations are removed.
In order to be able to convert the frequency variations into voltage variations, the demodulator
must be frequency dependent. The ideal response is a perfectly linear voltage to frequency
characteristic. Here it can be seen that the centre frequency is in the middle of the response
curve and this is where the un-modulated carrier would be located when the receiver is
correctly tuned into the signal. In other words there would be no offset DC voltage present.
The ideal response is not achievable because all systems have a finite bandwidth and as a
result a response curve known as an "S" curve is obtained. Outside the bandwidth of the
system, the response falls, as would be expected. It can be seen that the frequency variations
of the signal are converted into voltage variations which can be amplified by an audio
amplifier before being passed into headphones, a loudspeaker, or passed into other electronic
circuitry for the appropriate processing.
Characteristic "S" curve of an FM demodulator
To enable the best detection to take place the signal should be centred about the middle of the
curve. If it moves off too far then the characteristic becomes less linear and higher levels of
distortion result. Often the linear region is designed to extend well beyond the bandwidth of a
signal so that this does not occur. In this way the optimum linearity is achieved. Typically the
bandwidth of a circuit for receiving VHF FM broadcasts may be about 1 MHz whereas the
signal is only 200 kHz wide.
FM demodulators
There are a number of circuits that can be used to demodulate FM. Each type has its own
advantages and disadvantages, some being used when receivers used discrete components, and
others now that ICs are widely used.
Below is a list of some of the main types of FM demodulator or FM detector. In view of the
widespread use of FM, even with the competition from digital modes that are widely used
today, FM demodulators are needed in many new designs of electronics equipment.
Slope FM detector: This form of detector uses the slope of a tuned circuit to convert
the frequency variations into amplitude variations. As the frequency of the FM signal
varies, it changes its position on the slope of the tuned circuit, so the amplitude will
vary. This signal can then be converted into a baseband signal by using an AM diode
detector circuit. Read more about the Slope Detector
Ratio detector: This FM demodulator circuit was widely used with discrete
components, providing a good level of performance. It was characterised by the
transformer with three windings that was required. Read more about the Ratio
Detector
Foster-Seeley FM detector: Like the Ratio detector the Foster Seeley detector or
discriminator was used with discrete components, providing excellent performance
for the day in many FM radios. Read more about the Foster-Seeley Detector
PLL, Phase locked loop FM demodulator: FM demodulators using phase locked
loops, PLLs can provide high levels of performance. They do not require a costly
transformer and can easily be incorporated within FM radio ICs. Read more about the
PLL FM Detector
Quadrature FM demodulator: This form of FM demodulator is very convenient
for use within integrated circuits. It provides high levels of linearity, while not
requiring many external components. Read more about the Quadrature FM Detector
Coincidence FM demodulator: This form of demodulator has many similarities to
the quadrature detector. It uses digital technology and replaces a mixer with a logic
NAND gate.
Each of these different types of FM detector or demodulator has its own advantages and
disadvantages. These FM demodulators are described in further pages of this tutorial.
5.
Explain the Basic super heterodyne block diagram and its functionality?
The basic block diagram of a basic superheterodyne radio receiver is shown below. This
details the most basic form of the receiver and serves to illustrate the basic blocks and their
function.
Block diagram of a basic superheterodyne radio receiver
The way in which the receiver works can be seen by following the signal as is passes through
the receiver.
Front end amplifier and tuning block: Signals enter the front end circuitry from the
antenna. This circuit block performs two main functions:
o
Tuning: Broadband tuning is applied to the RF stage. The purpose of this is to reject the
signals on the image frequency and accept those on the wanted frequency. It must also be able to
track the local oscillator so that as the receiver is tuned, so the RF tuning remains on the required
frequency. Typically the selectivity provided at this stage is not high. Its main purpose is to reject
signals on the image frequency which is at a frequency equal to twice that of the IF away from the
wanted frequency. As the tuning within this block provides all the rejection for the image response,
it must be at a sufficiently sharp to reduce the image to an acceptable level. However the RF tuning
may also help in preventing strong off-channel signals from entering the receiver and overloading
elements of the receiver, in particular the mixer or possibly even the RF amplifier.
o
Amplification: In terms of amplification, the level is carefully chosen so that it does not
overload the mixer when strong signals are present, but enables the signals to be amplified
sufficiently to ensure a good signal to noise ratio is achieved. The amplifier must also be a low
noise design. Any noise introduced in this block will be amplified later in the receiver.
Mixer / frequency translator block: The tuned and amplified signal then enters one
port of the mixer. The local oscillator signal enters the other port. The performance of the mixer is
crucial to many elements of the overall receiver performance. It should eb as linear as possible. If
not, then spurious signals will be generated and these may appear as 'phantom' received signals.
Local oscillator: The local oscillator may consist of a variable frequency oscillator that
can be tuned by altering the setting on a variable capacitor. Alternatively it may be a frequency
synthesizer that will enable greater levels of stability and setting accuracy.
Intermediate frequency amplifier, IF block : Once the signals leave the mixer they
enter the IF stages. These stages contain most of the amplification in the receiver as well as the
filtering that enables signals on one frequency to be separated from those on the next. Filters may
consist simply of LC tuned transformers providing inter-stage coupling, or they may be much
higher performance ceramic or even crystal filters, dependent upon what is required.
Detector / demodulator stage: Once the signals have passed through the IF stages of
the superheterodyne receiver, they need to be demodulated. Different demodulators are required
for different types of transmission, and as a result some receivers may have a variety of
demodulators that can be switched in to accommodate the different types of transmission that are
to be encountered. Different demodulators used may include:
o
AM diode detector: This is the most basic form of detector and this circuit block would
simple consist of a diode and possibly a small capacitor to remove any remaining RF. The detector
is cheap and its performance is adequate, requiring a sufficient voltage to overcome the diode
forward drop. It is also not particularly linear, and finally it is subject to the effects of selective
fading that can be apparent, especially on the HF bands.
o
Synchronous AM detector: This form of AM detector block is used in where
improved performance is needed. It mixes the incoming AM signal with another on the same
frequency as the carrier. This second signal can be developed by passing the whole signal through
a squaring amplifier. The advantages of the synchronous AM detector are that it provides a far
more linear demodulation performance and it is far less subject to the problems of selective fading.
o
SSB product detector: The SSB product detector block consists of a mixer and a
local oscillator, often termed a beat frequency oscillator, BFO or carrier insertion oscillator, CIO.
This form of detector is used for Morse code transmissions where the BFO is used to create an
audible tone in line with the on-off keying of the transmitted carrier. Without this the carrier
without modulation is difficult to detect. For SSB, the CIO re-inserts the carrier to make the
modulation comprehensible.
o
Basic FM detector: As an FM signal carries no amplitude variations a demodulator
block that senses frequency variations is required. It should also be insensitive to amplitude
variations as these could add extra noise. Simple FM detectors such as the Foster Seeley or ratio
detectors can be made from discrete components although they do require the use of transformers.
o
PLL FM detector: A phase locked loop can be used to make a very good FM
demodulator. The incoming FM signal can be fed into the reference input, and the VCO drive
voltage used to provide the detected audio output.
o
Quadrature FM detector: This form of FM detector block is widely used within
ICs. IT is simple to implement and provides a good linear output.
Audio amplifier: The output from the demodulator is the recovered audio. This is
passed into the audio stages where they are amplified and presented to the headphones or
loudspeaker.
6.
Give short notes about Time-division multiplexing?
Time-division multiplexing (TDM) is a method of transmitting and receiving independent signals
over a common signal path by means of synchronized switches at each end of the transmission line
so that each signal appears on the line only a fraction of time in an alternating pattern. This form of
signal multiplexing was developed Intelecommunications for telegraphy systems in the late 1800s,
but found its most common application in digital telephony in the second half of the 20th century.
Technology
Time-division multiplexing is used primarily for digital signals, but may be applied in analog
multiplexing in which two or more signals or bit streams are transferred appearing simultaneously
as sub-channels in one communication channel, but are physically taking turns on the channel. The
time domain is divided into several recurrent time slots of fixed length, one for each sub-channel.
A sample byte or data block of sub-channel 1 is transmitted during time slot 1, sub-channel 2
during time slot 2, etc. One TDM frame consists of one time slot per sub-channel plus a
synchronization channel and sometimes error correction channel before the synchronization. After
the last sub-channel, error correction, and synchronization, the cycle starts all over again with a
new frame, starting with the second sample, byte or data block from sub-channel 1, etc.
Application examples
The plesiochronous digital hierarchy (PDH) system, also known as the PCM system, for
digital transmission of several telephone calls over the same four-wire copper cable (T-carrier or Ecarrier) or fiber cable in the circuit switched digital telephone network
The synchronous digital hierarchy digital (SDH)and Synchronous optical networking
(SONET) network transmission standards that have replaced PDH.
The Basic Rate Interface and Primary Rate Interface for the Integrated Services Digital
Network (ISDN).
The RIFF (WAV) audio standard interleaves left and right stereo signals on a per-sample basis
The left-right channel splitting in use for stereoscopic liquid crystal shutter glasses
TDM can be further extended into the (TDMA) scheme, where several stations connected to the
same physical medium, for example sharing the same frequency channel, can communicate.
Application examples include:
TDM versus packet-mode communication
In its primary form, TDM is used for circuit mode communication with a fixed number of channels
and constant bandwidth per channel.
Bandwidth reservation distinguishes time-division multiplexing from statistical multiplexing such
as packet mode communication (also known as statistical time-domain multiplexing, see below)
i.e. the time slots are recurrent in a fixed order and pre-allocated to the channels, rather than
scheduled on a packet-by-packet basis. Statistical time-domain multiplexing resembles, but should
not be considered the same as time-division multiplexing.
History
Time-division multiplexing was first developed for applications in telegraphy to route multiple
transmissions simultaneously over a single transmission line. In the 1870s, Emile Baudotdeveloped
a time-multiplexing system of multiple Hughes machines.
In 1953 a 24-channel TDM was placed in commercial operation by RCA Communications to send
audio information between RCA's facility at Broad Street, New York and their transmitting station
at Rocky Point and the receiving station at Riverhead, Long Island, New York. The
communication was by a microwave system throughout Long Island. The experimental TDM
system was developed by RCA Laboratories between 1950 and 1953.
In 1962, engineers from Bell Labs developed the first D1 Channel Banks, which combined 24
digitised voice calls over a 4-wire copper trunk between Bell central office analogue switches. A
channel bank sliced a 1.544 Mbit/s digital signal into 8,000 separate frames, each composed of 24
contiguous bytes. Each byte represented a single telephone call encoded into a constant bit rate
signal of 64 kbit/s. Channel banks used a byte's fixed position (temporal alignment) in the frame to
determine which call it belonged to.
Multiplexed digital transmission
In circuit-switched networks, such as the (PSTN), it is desirable to transmit multiple subscriber
calls over the same transmission medium to effectively utilize the bandwidth of the medium. TDM
allows transmitting and receiving telephone switches to create channels (tributaries) within a
transmission stream. A standard DSO voice signal has a data bit rate of 64 kbit/s. A TDM circuit
runs at a much higher signal bandwidth, permitting the bandwidth to be divided into time frames
(time slots) for each voice signal which is multiplexed onto the line by the transmitter. If the TDM
frame consists of n voice frames, the line bandwidth is n*64 kbit/s.
Each voice time slot in the TDM frame is called a channel. In European systems, standard TDM
frames contain 30 digital voice channels (E1), and in American systems (T1), they contain 24
channels. Both standards also contain extra bits (or bit time slots) for signalling and
synchronization bits.Multiplexing more than 24 or 30 digital voice channels is called higher order
multiplexing. Higher order multiplexing is accomplished by multiplexing the standard TDM
frames. For example, a European 120 channel TDM frame is formed by multiplexing four standard
30 channel TDM frames. At each higher order multiplex, four TDM frames from the immediate
lower order are combined, creating multiplexes with a bandwidth of n*64 kbit/s, where n = 120,
480, 1920, etc.
Synchronous time-division multiplexing
There are three types of synchronous TDM: T1, SONET/SDH, and ISDN.
Synchronous Digital Hierarchy (SDH)
Plesiochronous digital hierarchy (PDH) was developed as a standard for multiplexing higher order
frames. PDH created larger numbers of channels by multiplexing the standard Europeans 30
channel TDM frames. This solution worked for a while; however PDH suffered from several
inherent drawbacks which ultimately resulted in the development of the Synchronous Digital
Hierarchy (SDH). The requirements which drove the development of SDH were these:Be
synchronous – All clocks in the system must align with a reference clock.
Be service-oriented – SDH must route traffic from End Exchange to End Exchange without
worrying about exchanges in between, where the bandwidth can be reserved at a fixed level for a
fixed period of time.
Allow frames of any size to be removed or inserted into an SDH frame of any size.
Easily manageable with the capability of transferring management data across links.
Provide high levels of recovery from faults.
Provide high data rates by multiplexing any size frame, limited only by technology.
Give reduced bit rate errors.
SDH has become the primary transmission protocol in most PSTN networks.It was developed to
allow streams 1.544 Mbit/s and above to be multiplexed, in order to create larger SDH frames
known as Synchronous Transport Modules (STM). The STM-1 frame consists of smaller streams
that are multiplexed to create a 155.52 Mbit/s frame. SDH can also multiplex packet based frames
e.g., PPP and ATM.
While SDH is considered to be a transmission protocol (Layer 1 in the OSI Reference Model), it
also performs some switching functions, as stated in the third bullet point requirement listed above.
The most common SDH Networking functions are these:
SDH Crossconnect – The SDH Crossconnect is the SDH version of a Time-Space-Time
crosspoint switch. It connects any channel on any of its inputs to any channel on any of its outputs.
The SDH Crossconnect is used in Transit Exchanges, where all inputs and outputs are connected to
other exchanges.
SDH Add-Drop Multiplexer – The SDH Add-Drop Multiplexer (ADM) can add or remove
any multiplexed frame down to 1.544Mb. Below this level, standard TDM can be performed. SDH
ADMs can also perform the task of an SDH Crossconnect and are used in End Exchanges where
the channels from subscribers are connected to the core PSTN network. SDH network functions
are connected using high-speed optic fibre. Optic fibre uses light pulses to transmit data and is
therefore extremely fast.Modern optic fibre transmission makes use of (WDM) where signals
transmitted across the fibre are transmitted at different wavelengths, creating additional channels
for transmission. This increases the speed and capacity of the link, which in turn reduces both unit
and total costs.
Statistical time-division multiplexing
(STDM) is an advanced version of TDM in which both the address of the terminal and the data
itself are transmitted together for better routing. Using STDM allows bandwidth to be split over
one line. Many college and corporate campuses use this type of TDM to distribute bandwidth.
On a 10-Mbit line entering a network, STDM can be used to provide 178 terminals with a
dedicated 56k connection (178 * 56k = 9.96Mb). A more common use however is to only grant the
bandwidth when that much is needed. STDM does not reserve a time slot for each terminal, rather
it assigns a slot when the terminal is requiring data to be sent or received.
Asynchronous time-division multiplexing (ATDM), is an alternative nomenclature in which
STDM designates synchronous time-division multiplexing, the older method that uses fixed time
slots.
UNIT III
PART-A
1. What is PCM?
PCM is a type of pulse modulation like PAM, PWM or PPM but there is an
important difference between them. PAM, PWM or PPM is ―analog‖ pulse modulation
systems whereas PCM is a ―digital‖ pulse modulation system. This means the PCM
output in the form of digital pulses of constant amplitude, width and position.
The information transmitted in the form of code words. A PCM system consists of a
PCM encoder (transmitter) and a PCM decoder (receiver). The essential operations in the
PCM transmitter are sampling, quantizing and encoding.
2. Advantages of Digital Representation of a signal?
Immunity to transmission noise and interference.
Regeneration of the coded signal along the transmission path is possible.
Communication can be kept ―private‖ and ―secured‖ through the use of encryption.
The Possibility of uniform format for different kinds of baseband signals.
It is possible to store the signal and process it further.
3. Explain the block diagram of regenerative repeater?
It has three blocks i) The amplitude equalizer ii) The timing circuit iii) decision
making device.
i)The amplitude equalizer shapes the distorted PCM wave so as to compensate for the
effects of amplitude and phase distortions.
ii) The timing circuit produces a periodic pulse train in that is derived from the input
PCM pulses.
iii) The decision making device uses this pulse train for sampling the equalized PCM
pulses. The sampling is carried out at the instants where the signal to noise ratio is
maximum.
The decision device makes a decision about whether the equalized PCM wave at its input
has a 0 value or 1 value at the instant of sampling.
4. What are modulator and demodulator?
A modulator-demodulator modem is a device that modulates an analog carrier
signal to encode digital information, and also demodulates such a carrier signal to decode
the transmitted information. The goal is to produce a signal that can be transmitted easily
and decoded to reproduce the original digital data. Modems can be used with any means
of transmitting analog signals, from light emitting distortion to radio. The most familiar
example is a voice band modem that turns the digital data of a personal computer into
modulated electrical signals in the voice frequency range of a telephone channel. These
signals can be transmitted over telephone lines and demodulated by another modem at the
receiver side to recover the digital data.
5. Application of BPSK?
a) Phase shift keying is the most efficient of the three modulation methods (i.e.)
ASK, FSK, PSK such and it is used for high bit rates even higher than 1800
bits/sec.
b) Due to low bandwidth requirements the BPSK modems are preferred over the
FSK modems, at higher operating speeds.
6. Advantages of BPSK?
a) BPSK has a high bandwidth which is lower than that of a BFSK signal.
b) BPSK has the best performance of all the systems in presence of noise. It gives
the minimum possibility of error.
c) BPSK has very good noise immunity.
7. Mention the why QPSK is better than PSK?
a) Due to multilevel modulation used in QPSK, it is possible to increase the bit rate to
double the bit rate of QPSK without increasing bandwidth.
b) The noise immunity of QPSK is same as that of PSK system.
c) Available channel bandwidth is utilized in a better way by the QPSK system than PSK
system.
8. Difference between wave form coding and source coding?
The waveform coders are in principle designed to signal independent.
The waveform coders are different from the source coders (e.g.) (linear and predictive).
The source coders depend on parameterization of the analog signal in accordance with an
appropriate model for the generation of the signal.
9. For a binary PCM system, the number of bits per transmitted word is 8 and the
sampling frequencyfs = 8 kHz. Calculate the bit rate and baud rate?
Given N = 8, fs = 8 kHz
Bit rate = N * fs = 8 * 8 kHz = 64 bits/sec
Baud rate = Bit rate = 64 kHz ( as transmission is binary)
10. For the same data in the previous example, calculate the bit rate and baud rate if a
QPSK system is used?
In the QPSK system, two successive bits are clubbed together to form one message.
Hence symbol corresponds to 2 bit duration.
Therefore, Baud rate = ½ * bit rate = 64/2 = 32 k bits/sec
Bit rate does not change.
PART-B
1. Explain in detail about Amplitude-shift keying (ASK)?
It is a form of amplitude modulation that represents digital data as variations in
the amplitude of a carrier wave. Any digital modulation scheme uses a finite number of
distinct signals to represent digital data. ASK uses a finite number of amplitudes, each
assigned a unique pattern of binary digits. Usually, each amplitude encodes an equal
number of bits. Each pattern of bits forms the symbol that is represented by the particular
amplitude. The demodulator, which is designed specifically for the symbol-set used by
the modulator, determines the amplitude of the received signal and maps it back to the
symbol it represents, thus recovering the original data. Frequency and phase of the carrier
are kept constant.
Like AM, ASK is also linear and sensitive to atmospheric noise, distortions,
propagation conditions on different routes in PSTN, etc. Both ASK modulation and
demodulation processes are relatively inexpensive. The ASK technique is also commonly
used to transmit digital data over optical fiber. For LED transmitters, binary 1 is
represented by a short pulse of light and binary 0 by the absence of light. Laser
transmitters normally have a fixed "bias" current that causes the device to emit a low light
level. This low level represents binary 0, while a higher-amplitude light wave represents
binary 1.
The simplest and most common form of ASK operates as a switch, using the
presence of a carrier wave to indicate a binary one and its absence to indicate a binary
zero. This type of modulation is called on-off keying, and is used at radio frequencies to
transmit Morse code (referred to as continuous wave operation),
More sophisticated encoding schemes have been developed which represent data
in groups using additional amplitude levels. For instance, a four-level encoding scheme
can represent two bits with each shift in amplitude; an eight-level scheme can represent
three bits; and so on. These forms of amplitude-shift keying require a high signal-to-noise
ratio for their recovery, as by their nature much of the signal is transmitted at reduced
power.
ASK diagram
ASK system can be divided into three blocks. The first one represents the
transmitter, the second one is a linear model of the effects of the channel, and the third
one shows the structure of the receiver. The following notation is used:
ht(f) is the carrier signal for the transmission
hc(f) is the impulse response of the channel
n(t) is the noise introduced by the channel
hr(f) is the filter at the receiver
L is the number of levels that are used for transmission
Ts is the time between the generation of two symbols
Different symbols are represented with different voltages. If the maximum allowed
value for the voltage is A, then all the possible values are in the range [−A, A] and they
are given by:
the difference between one voltage and the other is:
Considering the picture, the symbols v[n] are generated randomly by the source S,
then the impulse generator creates impulses with an area of v[n]. These impulses are sent
to the filter ht to be sent through the channel. In other words, for each symbol a different
carrier wave is sent with the relative amplitude.
Out of the transmitter, the signal s(t) can be expressed in the form:
In the receiver, after the filtering through hr (t) the signal is:
where we use the notation:
where * indicates the convolution between two signals. After the A/D conversion the
signal z[k] can be expressed in the form:
In this relationship, the second term represents the symbol to be extracted. The
others are unwanted: the first one is the effect of noise, the third one is due to the inter
symbol interference.If the filters are chosen so that g(t) will satisfy the Nyquist ISI
criterion, then there will be no inter symbol interference and the value of the sum will be
zero, so:
the transmission will be affected only by noise.
Probability of error
The probability density function of having an error of a given size can be modeled
by a Gaussian function; the mean value will be the relative sent value, and its variance
will be given by:
where
is the spectral density of the noise within the band and Hr (f) is the
continuous Fourier transform of the impulse response of the filter hr (f).
The probability of making an error is given by:
where, for example,
is the conditional probability of making an error given that a
symbol v0 has been sent and P_{H_0} is the probability of sending a symbol v0.
If the probability of sending any symbol is the same, then:
If we represent all the probability density functions on the same plot against the possible
value of the voltage to be transmitted, we get a picture like this (the particular case of L =
4 is shown):
The probability of making an error after a single symbol has been sent is the area of the
Gaussian function falling under the functions for the other symbols. It is shown in cyan
for just one of them. If we call P+ the area under one side of the Gaussian, the sum of all
the areas will be: 2 L P^+ - 2 P^+. The total probability of making an error can be
expressed in the form:
We have now to calculate the value of P+. In order to do that, we can move the origin of
the reference wherever we want: the area below the function will not change. We are in a
situation like the one shown in the following picture:
it does not matter which Gaussian function we are considering, the area we want to
calculate will be the same. The value we are looking for will be given by the following
integral:
whereerfc() is the complementary error function. Putting all these results together, the
probability to make an error is:
from this formula we can easily understand that the probability to make an error
decreases if the maximum amplitude of the transmitted signal or the amplification of the
system becomes greater; on the other hand, it increases if the number of levels or the
power of noise becomes greater.
This relationship is valid when there is no inter symbol interference, i.e. g(t) is a NyQuist
function.
2. Explain in detail about Bit error rate calculation?
In digital transmission, the number of bit errors is the number of received bits of a
data stream over a communication channel that has been altered due to noise,
interference, distortion or bit synchronization errors.
The bit error rate or bit error ratio (BER) is the number of bit errors divided by the
total number of transferred bits during a studied time interval. BER is a unit less
performance measure, often expressed as a percentage.
The bit error probabilitype is the expectation value of the BER. The BER can be
considered as an approximate estimate of the bit error probability. This estimate is
accurate for a long time interval and a high number of bit errors.
Example
As an example, assume this transmitted bit sequence:
0 1 1 0 0 0 1 0 1 1,
and the following received bit sequence:
0 0 1 0 1 0 1 0 0 1,
The number of bit errors (the underlined bits) is in this case 3. The BER is 3 incorrect bits
divided by 10 transferred bits, resulting in a BER of 0.3 or 30%.
Packet error rate
The packet error rate (PER) is the number of incorrectly received data packets
divided by the total number of received packets. A packet is declared incorrect if at least
one bit is erroneous. The expectation value of the PER is denoted packet error
probabilitypp, which for a data packet length of N bits can be expressed as
,
Assuming that the bit errors are independent of each other. For small bit error
probabilities, this is approximately
Similar measurements can be carried out for the transmission of frame, blocks or
symbols.
Factors affecting the BER,
In a communication system, the receiver side BER may be affected by
transmission channel noise, interference, distortion, bit synchronization problems,
attenuation, wireless multipath fading, etc.
The BER may be improved by choosing a strong signal strength (unless this
causes cross-talk and more bit errors), by choosing a slow and robust modulation scheme
or line coding scheme, and by applying channel coding schemes such as redundant
forward error correction codes.
The transmission BER is the number of detected bits that are incorrect before
error correction, divided by the total number of transferred bits (including redundant error
codes). The information BER, approximately equal to the decoding error probability, is
the number of decoded bits that remain incorrect after the error correction, divided by the
total number of decoded bits (the useful information). Normally the transmission BER is
larger than the information BER. The information BER is affected by the strength of the
forward error correction code.
Analysis of the BER
The BER may be analyzed using stochastic computer simulations. If a simple
transmission channel model and data source model is assumed, the BER may also be
calculated analytically. An example of such a data source model is the Bernoulli source.
Bit-error rate curves for BPSK, QPSK, 8-PSK and 16-PSK, AWGN channel.
BER comparison between BPSK and differentially-encoded BPSK with graycoding operating in white noise.
In a noisy channel, the BER is often expressed as a function of the normalized
carrier-to-noise ratio measure denoted Eb/No, (energy per bit to noise power spectral
density ratio), or Es/No (energy per modulation symbol to noise spectral density).
For example, in the case of QPSK modulation and AWGN channel, the BER as function
of the Eb/N0 is given by:
People usually plot the BER curves to describe the functionality of a digital
communication system. In optical communication, BER (dB) vs. Received Power (dBm)
is usually used; while in wireless communication, BER (dB) vs. SNR(dB) is used.
Bit error rate test
BERT or bit error rate test is a testing method for digital communication circuits
that uses predetermined stress patterns consisting of a sequence of logical ones and zeros
generated by a test pattern generator.
A BERT typically consists of a test pattern generator and a receiver that can be set
to the same pattern. They can be used in pairs, with one at either end of a transmission
link, or singularly at one end with a loop back at the remote end. BERTs are typically
stand-alone specialized instruments, but can be Personal Computer based. In use, the
number of errors, if any, are counted and presented as a ratio such as 1 in 1,000,000, or 1
in 1e06.
Common types of BERT stress patterns
PRBS (pseudo random binary sequence) – A pseudorandom binary sequencer of
N Bits. These pattern sequences are used to measure jitter and eye mask of TX-Data in
electrical and optical data links.
QRSS (Quasi Random Signal Source) – A pseudorandom binary sequencer which
generates every combination of a 20-bit word, repeats every 1,048,575 bits, and
suppresses consecutive zeros to no more than 14. It contains high-density sequences, lowdensity sequences, and sequences that change from low to high and vice versa. This
pattern is also the standard pattern used to measure jitter.
3 in 24 – Pattern contains the longest string of consecutive zeros (15) with the
lowest ones density (12.5%). This pattern simultaneously stresses minimum ones density
and the maximum number of consecutive zeros. The D4 frame format of 3 in 24 may
cause a D4 Yellow Alarm for frame circuits depending on the alignment of one bits to a
frame.
1:7 – Also referred to as ―1 in 8‖. It has only a single one in an 8-bit repeating
sequence. This pattern stresses the minimum ones density of 12.5% and should be used
when testing facilities set for B8ZS coding as the 3 in 24 pattern increases to 29.5% when
converted to B8ZS.
Min/Max – Pattern rapid sequence changes from low density to high density.
Most useful when stressing the repeaters ALBO feature.
All Ones (or Mark) – A pattern composed of ones only. This pattern causes the
repeater to consume the maximum amount of power. If DC to the repeater is regulated
properly, the repeater will have no trouble transmitting the long ones sequence. This
pattern should be used when measuring span power regulation. An unframed all ones
pattern is used to indicate an AIS(also known as a Blue Alarm).
All Zeros – A pattern composed of zeros only. It is effective in finding equipment
misoptioned for AMI, such as fiber/radio multiplex low-speed inputs.
Alternating 0s and 1s - A pattern composed of alternating ones and zeroes.
2 in 8 – Pattern contains a maximum of four consecutive zeros. It will not invoke
a B8ZS sequence because eight consecutive zeros are required to cause a B8ZS
substitution. The pattern is effective in finding equipment misoptioned for B8ZS.
Bridge tap - within a span can be detected by employing a number of test patterns
with a variety of ones and zeros densities. This test generates 21 test patterns and runs for
15 minutes. If a signal error occurs, the span may have one or more bridge taps. This
pattern is only effective for T1 spans that transmit the signal raw. Modulation used in
HDSL spans negates the Bridgetap patterns' ability to uncover bridge taps.
Multipath - This test generates 5 commonly used test patterns to allow DS1 span
testing without having to select each test pattern individually. Patterns are: All Ones, 1:7,
2 in 8, 3 in 24, and QRSS.
T1-DALY and 55 OCTET - Each of these patterns contain fifty-five (55), eight
bit octets of data in a sequence that changes rapidly between low and high density. These
patterns are used primarily to stress the ALBO and equalizer circuitry but they will also
stress timing recovery. 55 OCTET has fifteen (15) consecutive zeroes and can only be
used unframed without violating ones density requirements. For framed signals, the T1DALY pattern should be used. Both patterns will force a B8ZS code in circuits optioned
for B8ZS.
Bit error rate tester
A bit error rate tester (BERT), also known as a bit error ratio testeror bit error rate
test solution (BERTs) is electronic test equipment used to test the quality of signal
transmission of single components or complete systems.
The main building blocks of a BERT are:
Pattern Generator, which transmits a defined test pattern to the DUT or test
system
Error detector connected to the DUT or test system, to count the errors generated
by the DUT or test system
Clock signal generator to synchronize the pattern generator and the error detector
Digital communication analyzer is optional to display the transmitted or received
signal
Electrical-optical converter and optical-electrical converter for testing optical
communication signals.
3. Explain in detail about Frequency-shift keying?
An example of binary FSK
Frequency-shift keying (FSK) is a frequency modulation scheme in which digital
information is transmitted through discrete frequency changes of a carrier wave. The
simplest FSK is binary FSK (BFSK). BFSK uses a pair of discrete frequencies to transmit
binary (0s and 1s) information. With this scheme, the "1" is called the mark frequency
and the "0" is called the space frequency. The time domain of an FSK modulated carrier
is illustrated in the figures to the right.
Definition - What does Frequency-Shift Keying (FSK) mean?
Frequency-shift keying (FSK) allows digital information to be transmitted by
changes or shifts in the frequency of a carrier signal, most commonly an analog carrier
sine wave. There are two binary states in a signal, zero (0) and one (1), each of which is
represented by an analog wave form. This binary data is converted by a modem into an
FSK signal, which can be transmitted via telephone lines, fiber optics or wireless media.
FSK is commonly used for caller ID and remote metering applications.
FSK is also known as frequency modulation (FM).
For example, a low-speed Hayes-compatible modem uses an unbit FM technique.
When no digital information is transmitted, the frequency is 1,700 Hz. When a one is
transmitted, the frequency shifts to 2,200 Hz. When a zero is transmitted, the frequency
shifts to 1,200 Hz. The number of these frequency shifts per second is measured as the
baud or modulation rate. Thus, a 2,400 baud modem can process zeros and ones from a
computer at the rate of 2,400 bits per second using FSK. This is the simplest digital
communication, where baud and bit rate are the same and measured in bits per second.
In more advanced modems and data transmission techniques, a symbol may have
more than two states, not just zeros and ones. It may also represent more than one bit of
information. However, a single bit always represents one of two states – either a zero (0)
or a one (1). In this case, baud (or symbol rate expressed in symbols/second or pulses
/second) and bit rate are different and must not be confused with one another.
Other forms of FSK
Minimum-shift keying
Minimum frequency-shift keying or minimum-shift keying (MSK) is a particular
spectrally efficient form of coherent FSK. In MSK, the difference between the higher and
lower frequency is identical to half the bit rate. Consequently, the waveforms that
represent a 0 and a 1 bit differ by exactly half a carrier period. The maximum frequency
deviation is δ = 0.25 fm, where fm is the maximum modulating frequency. As a result, the
modulation index m is 0.25. This is the smallest FSK modulation index that can be
chosen such that the waveforms for 0 and 1 are orthogonal. A variant of MSK called
GMSK is used in the GSM mobile phone standard.
Audio FSK
Audio frequency-shift keying (AFSK) is a modulation technique by which digital
data is represented by changes in the frequency (pitch) of an audio tone, yielding an
encoded signal suitable for transmission via radio or telephone. Normally, the transmitted
audio alternates between two tones: one, the "mark", represents a binary one; the other,
the "space", represents a binary zero.
AFSK differs from regular frequency-shift keying in performing the modulation
at baseband frequencies. In radio applications, the AFSK-modulated signal normally is
being used to modulate an RF carrier (using a conventional technique, such as AM or
FM) for transmission.
AFSK is not always used for high-speed data communications, since it is far less
efficient in both power and bandwidth than most other modulation modes. In addition to
its simplicity, however, AFSK has the advantage that encoded signals will pass through
AC-coupled links, including most equipment originally designed to carry music or
speech.
AFSK is used in the U.S. based Emergency Alert System to notify stations of the
type of emergency, locations affected, and the time of issue without actually hearing the
text of the alert.
Applications
1200 baud AFSK signal
Menu
0:00
Listen to an example of a 1200 baud AFSK-modulated signal.
In 1910, Reginald Fessenden invented a two-tone method of transmitting Morse
code. Dots and dashes were different tones of equal length. The intent was to minimize
transmission time.
Some early CW transmitters employed an arc converter that could not be
conveniently keyed. Instead of turning the arc on and off, the key slightly changed the
transmitter frequency in a technique known as the compensation-wave method. The
compensation-wave was not used at the receiver. The method consumed a lot of
bandwidth and caused interference, so it was discouraged by 1921.
Most early telephone-line modems used audio frequency-shift keying (AFSK) to
send and receive data at rates up to about 1200 bits per second. The common Bell 103
and Bell 202 modems used this technique. Even today, North American caller ID uses
1200 baud AFSK in the form of the Bell 202 standard. AFSK is still widely used in
amateur radio, as it allows data transmission through unmodified voiceband equipment.
Radio control gear uses FSK, but calls it FM and PPM instead.
AFSK is also used in the United States Emergency Alert System to transmit
warning information. It is used at higher bitrates for Weather copy used on Weather radio
by NOAA in the U.S.
The CHU shortwave radio station in Ottawa, Canada broadcasts an exclusive
digital time signal encoded using AFSK modulation.
FSK is commonly usedin Caller ID and remote metering applications: see FSK
standards for use in Caller ID and remote metering for more details.
4. Give short notes about PSK?
In phase shift keying (PSK), the phase of a carrier is changed according to the
modulating waveform which is a digital signal. In BPSK, the transmitted signal is a
sinusoid of fixed amplitude. It has one fixed phase when the data is at one level and when
the data is at the other level, phase is different by 180 degree. A Binary Phase Shift
Keying(BPSK) signal can be defined as
where b(t) = +1 or -1, fc is the carrier frequency, and T is the bit duration. The signal
has a power
sinusoidal carrier.
, so that
, where A represents the peak value of
Thus the above equation can be written as
=
=
=
, where E=PT is the energy contained in the bit
duration.
Figure.Shows the BPSK signal for bit sequence 1001,
Where
and
The received signal has the form
=
, where is the phase
shift introduced by the channel. The signal b(t) is recovered in the demodulator. If
synchronous demodulation is used, the waveform
is required at the
demodulator. Carrier recovery scheme in the demodulator is shown in Fig 2.
Figure: BPSK modulator and demodulator
The recovered carrier is multiplied with the received signal to generate
Assuming that integral number of carrier cycles is present in bit duration
voltage
and the bit synchronizer in Fig 2 knows the end of a bit interval and beginning of
the next, the output voltage
at the output of the integrate and dump circuit is:
If the channel is noisy, some of the demodulated bits will be in error.
5. What is modulator and demodulator (Modem) and give its applications?
A modem (modulator-demodulator) is a device that modulates an analog carrier
signalto encode digital information, and also demodulates such a carrier signal to decode
the transmitted information. The goal is to produce a signal that can be transmitted easily
and decoded to reproduce the original digital data. Modems can be used with any means
of transmitting analog signals, from light emitting diodes to radio. The most familiar
example is a voice band modem that turns the digital data of a personal computer into
modulated electrical signals in the voice frequency range of a telephone channel. These
signals can be transmitted over telephone lines and demodulated by another modem at the
receiver side to recover the digital data.
Modems are generally classified by the amount of data they can send in a given
unit of time, usually expressed in bits per second (bit/s, or bps), or bytes per second (B/s).
Modems can alternatively be classified by their symbol rate, measured in baud. The baud
unit denotes symbols per second, or the number of times per second the modem sends a
new signal. For example, the ITU V.21 standard used audio frequency shift keying with
two possible frequencies corresponding to two distinct symbols (or one bit per symbol),
to carry 300 bits per second using 300 baud. By contrast, the original ITU V.22 standard,
which could transmit and receive four distinct symbols (two bits per symbol), handled
1,200 bit/s by sending 600 symbols per second (600 baud) using phase shift keying.
Acoustic couplers
The Novation CAT acoustically coupled modem
For many years, the Bell System (AT&T) maintained a monopoly on the use of its
phone lines, and what devices could be connected to its lines. However, the seminal
Hush-a-Phone v. FCC case of 1956 concluded that it was within the FCC's jurisdiction to
regulate the operation of the System. Subsequently, the FCC examiner found that as long
as the device was not electrically attached it would not threaten to degenerate the system.
This led to a number of devices that mechanically connected to the phone, through a
standard handset. Since most handsets were supplied from Western Electric, it was
relatively easy to build such an acoustic coupler, and this style of connection was used for
many devices like answering machines.
Acoustically coupled Bell 103A-compatible 300 bit/s modems became common
during the 1970s, with well-known models including the Novation CAT and the
Anderson-Jacobson, the later spun off from an in-house project at Stanford Research
Institute. An even lower-cost option was the Penny whistle modem, designed to be built
using parts found at electronics scrap and surplus stores.
In December 1972, Vadicintroduced the VA3400, which was notable because it
provided full duplex operation at 1,200 bit/s over the phone network. Like the 103A, it
used different frequency bands for transmit and receive. In November 1976, AT&T
introduced the 212A modem to compete with Vadic. It was similar in design to Vadic's
model, but used the lower frequency set for transmission. One could also use the 212A
with a 103A modem at 300 bit/s. According to Vadic, the change in frequency
assignments made the 212 intentionally incompatible with acoustic coupling, thereby
locking out many potential modem manufacturers. In 1977, Vadic responded with the
VA3467 triple modem, an answer-only modem sold to computer center operators that
supported Vadic's 1,200-bit/s mode, AT&T's 212A mode, and 103A operation.
Carterfone and direct connection
The Hush-a-Phone decision applied only to mechanical collections, but the
Cartefonedecision of 1968 led to the FCC introducing a rule setting stringent AT&Tdesigned tests for electronically coupling a device to the phone lines. AT&T's tests were
complex, making electronically coupled modems expensive. so acoustically coupled
modems remained common into the early 1980s.
However, the rapidly falling prices of electronics in the late 1970s led to an
increasing number of direct-connect models around 1980. In spite of being directly
connected, these modems were generally operated like their earlier acoustic versions dialling and other phone-control operations were completed by hand, using an attached
handset. A small number of modems added the ability to automatically answer incoming
calls, or automatically place an outgoing call to a single number, but even these limited
features were relatively rare or limited to special models in a lineup. When more flexible
solutions were needed, 3rd party "dialers" were used to automate calling, normally using
a separate serial port.
The Smartmodem and the rise of BBSs
The original model 300 baud Smartmodem
The next major advance in modems was the Hayes Smart modem, introduced in
1981. The Smartmodem was an otherwise standard 103A 300-bit/s modem, but it was
attached to a small microcontroller that let the computer send it commands. The
command set included instructions for picking up and hanging up the phone, dialing
numbers, and answering calls. This eliminated the need for any manual operation, a
handset, or a dialer. Terminal programs that maintained lists of phone numbers and sent
the dialing commands became common. The basic Hayes command set remains the basis
for computer control of most modern modems.
The Smartmodem and its clones also aided the spread of bulletin board systems
(BBSs) because it was the first low-cost modem that could answer calls. Modems had
previously been typically either the call-only, acoustically coupled models used on the
client side, or the much more expensive, answer-only models used on the server side.
These were fine for large computer installations, but useless for the hobbies who wanted
to run a BBS but then periodically use the same telephone line to call other systems. The
first hobby BBS system, CBBS, started as an experiment in ways to better use the Smart
modem.
Almost all modern modems can inter-operate with fax machines. Digital faxes,
introduced in the 1980s, are simply an image format sent over a high-speed (commonly
14.4 kbit/s) modem. Software running on the host computer can convert any image into
fax format, which can then be sent using the modem. Such software was at one time an
add-on, but has since become largely universal.
V.34/28.8k and 33.6k
An ISA modem manufactured to conform to the V.34 protocol.
Any interest in these systems was destroyed during the lengthy introduction of the
28,800 bit/s V.34 standard. While waiting, several companies decided to release
hardware and introduced modems they referred to as V.FAST. In order to guarantee
compatibility with V.34 modems once the standard was ratified (1994), the
manufacturers were forced to use more flexible parts, generally a DSP and
microcontroller, as opposed to purpose-designed ASIC modem chips.
V.61/V.70 Analog/Digital Simultaneous Voice and Data
The V.61 Standard introduced Analog Simultaneous Voice and Data (ASVD).
This technology allowed users of v.61 modems to engage in point-to-point voice
conversations with each other while their respective modems communicated.
In 1995, the first DSVD (Digital Simultaneous Voice and Data) modems became
available to consumers, and the standard was ratified as v.70 by the International
Telecommunication Union (ITU) in 1996.
Two DSVD modems can establish a completely digital link between each other
over standard phone lines. Sometimes referred to as "the poor man's ISDN", and
employing a similar technology, v.70 compatible modems allow for a maximum speed of
33.6 kbit/s between peers. By using a majority of the bandwidth for data and reserving
part for voice transmission, DSVD modems allow users to pick up a telephone handset
interfaced with the modem, and initiate a call to the other peer.
One practical use for this technology was realized by early two-player video
gamers, who could hold voice communication with each other over the phone while
playing.
Using digital lines and PCM (V.90/92)
Softmodem
A Win modem or soft modem is a stripped-down modem that replaces tasks
traditionally handled in hardware with software. In this case the modem is a simple
interface designed to act as a digital-to-analog and an analog-to-digital converter.
Softmodems are cheaper than traditional modems because they have fewer hardware
components. However, the software generating and interpreting the modem tones to be
sent to the softmodem uses many system resources. For online gaming, this can be a real
concern. Another problem is the lack of cross-platform compatibility, meaning that nonWindows operating systems (such as Linux) often do not have an equivalent driver to
operate the modem.
Radio modems
Direct broadcast satellite, Wi-Fi, and mobile phones all use modems to
communicate, as do most other wireless services today. Modern telecommunications and
data networks also make extensive use of radio modems where long distance data links
are required. Such systems are an important part of the PSTN, and are also in common
use for high-speed computer network links to outlying areas where fiber is not
economical.
Even where a cable is installed, it is often possible to get better performance or
make other parts of the system simpler by using radio frequencies and modulation
techniques through a cable. Coaxial cable has a very large bandwidth; however signal
attenuation becomes a major problem at high data rates if a baseband digital signal is
used. By using a modem, a much larger amount of digital data can be transmitted through
a single wire. Digital cable television and cable Internet services use radio frequency
modems to provide the increasing bandwidth needs of modern households. Using a
modem also allows for frequency-division multiple accessto be used, making full-duplex
digital communication with many users possible using a single wire.
Wireless modems come in a variety of types, bandwidths, and speeds. Wireless
modems are often referred to as transparent or smart. They transmit information that is
modulated onto a carrier frequency to allow many simultaneous wireless communication
links to work simultaneously on different frequencies.
Transparent modems operate in a manner similar to their phone line modem
cousins. Typically, they were half duplex, meaning that they could not send and receive
data at the same time. Typically transparent modems are polled in a round robin manner
to collect small amounts of data from scattered locations that do not have easy access to
wired infrastructure. Transparent modems are most commonly used by utility companies
for data collection.
Smart modems come with media access controllers inside, which prevent random
data from colliding and resend data that is not correctly received. Smart modems
typically require more bandwidth than transparent modems, and typically achieve higher
data rates. The IEEE 802.11 standard defines a short range modulation scheme that is
used on a large scale throughout the world.
Wi-Fi and Wi-Max
Wireless data modems are used in the Wi-Fi and Wi-Max standards, operating at
microwave frequencies.
Mobile broadband modems
USB wireless modem
Modems which use a mobile telephone system (GPRS, UMTS, HSPA, EVDO,
Wi-Max, etc.), are known as mobile broadband modems (sometimes also called wireless
modems). Wireless modems can be embedded inside a laptopor appliance, or be external
to it. External wireless modems are connecting cards, USB modems for mobile
broadband and cellular routers. A connect card is a PC card or Express card which slides
into a PCMCIA/PC card/Express Card slot on a computer. USB wireless modems use a
USB port on the laptop instead of a PC card or Express Card slot. A USB modem used
for mobile broadband Internet is also sometimes referred to as a dongle. A cellular router
may have an external datacard (Air Card) that slides into it. Most cellular routers do
allow such data cards or USB modems. Cellular routers may not be modems by
definition, but they contain modems or allow modems to be slid into them. The difference
between a cellular router and a wireless modem is that a cellular router normally allows
multiple people to connect to it (since it can route data or support multipoint to multipoint
connections), while a modem is designed for one connection.
Most of GSM wireless modems come with an integrated SIM card holder and
some models are also provided with a microSD memory slot and/or jack for additional
external antenna such as Huawei E1762 and Sierra Wireless Compass 885. The CDMA
(EVDO) versions do not use R-UIM cards, but use Electronic Serial Number (ESN)
instead.
The cost of using a wireless modem varies from country to country. Some carriers
implement flat rate plans for unlimited data transfers. Some have caps (or maximum
limits) on the amount of data that can be transferred per month. Other countries have
plans that charge a fixed rate per data transferred—per megabyte or even kilobyte of data
downloaded; this tends to add up quickly in today's content-filled world, which is why
many peopleare pushing for flat data rates.
The faster data rates of the newest wireless modem technologies (UMTS, HSPA,
EVDO, Wi-Max) are also considered to be broadband wireless modems and compete
with other broadband modems below.
Until the end of April 2011, worldwide shipments of USB modems surpassed
embedded 3G and 4G modules by 3:1 because USB modems can be easily discarded, but
embedded modems could start to gain popularity as tablet sales grow and as the
incremental cost of the modems shrinks, so by 2016 the ratio may change to 1:1.Like
mobile phones, mobile broadband modems can be SIM locked to a particular network
provider. Unlocking a modem is achieved the same way as unlocking a phone, by using
an 'unlock code'
Broadband
DSL Modem
(asymmetric digital subscriber line) ADSL modems, a more recent development,
are not limited to the telephone's voiceband audio frequencies. Some ADSL modems use
coded orthogonal frequency division modulation (DMT, for Discrete MultiTone; also
called COFDM, for digital TV in much of the world).
DSL modems utilize a property that standard twisted-pair telephone cable can be
used for short distances to carry much higher frequency signals than what the cable is
actually rated to handle. This is also why DSL modems have a distance limitation.
Standard voice and slower 56 kilobit modem communications are possible over many
kilometers of cable, but the higher frequencies used by DSL are attenuated and DSL's
maximum performance gradually declines as the cable length increases.
Cable Modemsuse a range of frequencies originally intended to carry RF
television channels, and can coexist on the same single cable alongside standard RF
channel signals. Multiple cable modems attached to a single cable can use the same
frequency band, using a low-level media access protocol to allow them to work together
within the same channel. Typically, uplink and downlink signals are kept separate using
frequency division multiple access. For a single-cable distribution system, the return
signals from customers require special bidirectional amplifiers or reverse path amplifiers
that can send specific customer frequency bands upstream to the cable plant amongst the
other downstream frequency bands.
New types of broadband modems are beginning to appear, such as double way
satellite and power line modems.
Broadband modems should still be classified as modems, since they use complex
waveforms to carry digital data. They are more advanced devices than traditional dial-up
modems as they are either capable of modulating/demodulating hundreds of channels
simultaneously and/or are capable of using much wider channels than dial-up modems.
Many broadband modems include the functions of a router, such as Ethernet and
Wi-Fi, and other features such as DHCP, NAT and firewalls.
When broadband technology was introduced, networking and routers were
unfamiliar to consumers. However, many people knew what a modem was because
Internet access was still commonly done through dial-up. Due to this familiarity,
companies started selling broadband modems using the familiar term "modem", rather
than vaguer ones such as "adapter," "transceiver," or "bridge."
Home networking
Although the name modem is seldom used in this case, modems are also used for
high-speed home networking applications, especially those using existing home wiring.
One example is the G.hn standard, developed by ITU-T, which provides a high-speed (up
to 1 Gbit/s) LAN using existing home wiring (power lines, phone lines and coaxial
cables). G.hn devices use orthogonal frequency division multiplexing (OFDM) to
modulate a digital signal for transmission over the wire.
The phrase "null modem‖ was used to describe attaching a specially wired cable
between the serial ports of two personal computers. Basically, the transmit output of one
computer was wired to the receive input of the other; this was true for both computers.
The same software used with modems (such as ProComm or Mincom) could be used with
the null modem connection.
Voice modem
Voice modems are regular modems that are capable of recording or playing audio
over the telephone line. They are used for telephony applications. This type of modem
can be used as an FXO card for Private branch exchange systems (compare V.92).
Popularity
A CEA study in 2006 found that dial-up Internet access is declining in the U.S. In
2000, dial-up Internet connections accounted for 74% of all U.S. residential Internet
connections. The US demographic pattern for dial-up modem users per capita has been
more or less mirrored in Canada and Australia for the past 20 years.
Dial-up modem use in the US had dropped to 60% by 2003, and in 2006 stood at
36%. Voiceband modems were once the most popular means of Internet access in the
U.S., but with the advent of new ways of accessing the Internet, the traditional 56K
modem is losing popularity. The dial up modem is still widely used by customers in rural
areas, where DSL, Cable or Fiber Optic Service is not available, or they are unwilling to
pay what these companies charge.AOL in its 2012 annual report showed it still collects
around $700 million in fees from dial-up users; about 3 million people.
6. Give a Comparison of CW (Continuous Wave Systems)?
Parameter
ASK
BPSK
QPSK
QAM
M-ary
PSK
BPSK
Information is
transmitted by
change in
Amplitude
Phase
Phase
Amplitude
& Phase
Phase
Frequency
Number of bits
per symbol
N=1
N=1
N=2
N
N
N=1
Number of
possible
symbols M=2N
Two
Two
Four
M=2N
M=2N
Two
Detection
Method
Coherent
Coherent
Coherent
Coherent
Coherent
NonCoherent
For M=16
2
Sin π/M
Minimum
Euclidean
distance
Minimum
bandwidth
Symbol
duration Ts
b
2
2
2
2fb
2fb
fb
2fb/N
2fb/N
4fb
Tb
Tb
2 Tb
N Tb
N Tb
Tb
IV UNIT 2 MARKS
1. Define data communication codes.
Data communication codes are prescribed bit sequences used for encoding
Characters and symbols.
2. Define error detection.
Error detection is simply the process of monitoring the received data and
determining when a transmission has occurred.
3. Define Echoplex.
Echoplex is a relatively simple type of error detection scheme that is used
almost exclusively in data communications systems where human operators are
used to enter the data manually from a keyboard.
4. Describe serial interface.
Serial interface is used to ensure an orderly flow of data between the line
Control unit and the modem.
5. Define parallel interface.
Parallel interfaces transfer data between two devices eight or more bits a a
time. That is one entire data word is transmitted at a time .Parallel transmission is
sometimes referred to as serial by word transmission.
6. What are the advantages of parallel transmission?
The advantage of parallel transmission is data are transmitted much faster
than with serial transmission because there is a transmission path for each bit of
the word.
In parallel interface there is no need to convert data from parallel to serial
or vice versa.
7. What is the purpose of data modem?
The primary purpose of data modem is to interface computers, computer
networks, and other digital terminal equipment to analog communication lines
and radio terminals.
8. Classify data modems.
Data modems are generally classified in to synchronous and asynchronous
data modems.
9. Define OSI.
The term open system interconnection is the name for a set of standards
For communications among computers. The primary purpose of OSI standards is
To serve as a structural guideline for exchanging information between computers,
Terminals and networks.
10. Describe LAN.
A local area network is usually a privately owned and links the devices in
a single office, building or campus of up to a few kilometres in size.
IV UNIT 16 MARK
1. Describe LAN?
A local area network (LAN) is a computer network that interconnects computers in a
limited area such as a home, school, computer laboratory, or office building using network
media.[1] The defining characteristics of LANs, in contrast to wide area networks (WANs),
include their usually higher data-transfer rates, smaller geographic area, and lack of a need
for leased telecommunication lines.
ARCNET, Token Ring and other technology standards have been used in the past, but
Ethernet over twisted pair cabling, and Wi-Fi are the two most common technologies
currently used to build LANs.
A conceptual diagram of a local area network using 10BASE5 Ethernet
The increasing demand and use of computers in universities and research labs in the late
1960s generated the need to provide high-speed interconnections between computer
systems. A 1970 report from the Lawrence Radiation Laboratory detailing the growth of
their "Octopus" network gave a good indication of the situation.
Cambridge Ring was developed at Cambridge University in 1974 but was never developed
into a successful commercial product.
Ethernet was developed at Xerox PARC in 1973–1975, and filed as U.S. Patent 4,063,220.
In 1976, after the system was deployed at PARC, Metcalfe and Boggs published a seminal
paper, "Ethernet: Distributed Packet-Switching For Local Computer Networks."
ARCNET was developed by Data point Corporation in 1976 and announced in 1977. It
had the first commercial installation in December 1977 at Chase Manhattan Bank in New
York.
Standards evolution
The development and proliferation of personal computers using the CP/M operating
system in the late 1970s, and later DOS-based systems starting in 1981, meant that many
sites grew to dozens or even hundreds of computers. The initial driving force for
networking was generally to share storage and printers, which were both expensive at the
time. There was much enthusiasm for the concept and for several years, from about 1983
onward, computer industry pundits would regularly declare the coming year to be ―the
year of the LAN‖.
In practice, the concept was marred by proliferation of incompatible physical layer and
network protocol implementations, and a plethora of methods of sharing resources.
Typically, each vendor would have its own type of network card, cabling, protocol, and
network operating system. A solution appeared with the advent of Novell NetWare which
provided even-handed support for dozens of competing card/cable types, and a much more
sophisticated operating system than most of its competitors. Netware dominated the
personal computer LAN business from early after its introduction in 1983 until the mid1990s when Microsoft introduced Windows NT Advanced Server and Windows for
Workgroups.
Of the competitors to NetWare, only Banyan Vines had comparable technical strengths,
but Banyan never gained a secure base. Microsoft and 3Com worked together to create a
simple network operating system which formed the base of 3Com's 3+Share, Microsoft's
LAN Manager and IBM's LAN Server - but none of these was particularly successful.
During the same period, Unix computer workstations from vendors such as Sun
Microsystems, Hewlett-Packard, Silicon Graphics, Intergraph, NeXT and Apollo were
using TCP/IP based networking. Although this market segment is now much reduced, the
technologies developed in this area continue to be influential on the Internet and in both
Linux and Apple Mac OS X networking—and the TCP/IP protocol has now almost
completely replaced IPX, AppleTalk, NBF, and other protocols used by the early PC
LANs.
Cabling
Early LAN cabling had been based on various grades of coaxial cable. Shielded twisted
pair was used in IBM's Token Ring LAN implementation. In 1984, Star LAN showed the
potential of simple unshielded twisted pair by using Cat3 cable—the same simple cable
used for telephone systems. This led to the development of 10Base-T (and its successors)
and structured cabling which is still the basis of most commercial LANs today. In
addition, fiber-optic cabling is increasingly used in commercial applications.
As cabling is not always possible, Wi-Fi is now very common in residential premises, and
elsewhere where support for laptops and smart phones is important.
Technical aspects
Network topology describes the layout of interconnections between devices and network
segments. At the Data Link Layer and Physical Layer, a wide variety of LAN topologies
have been used, including ring, bus, mesh and star, but the most common LAN topology
in use today is switched Ethernet. At the higher layers, the Internet Protocol (TCP/IP) has
become the standard, replacing NetBEUI, IPX/SPX, AppleTalk and others.
Simple LANs generally consist of one or more switches. A switch can be connected to a
router, cable modem, or ADSL modem for Internet access. Complex LANs are
characterized by their use of redundant links with switches using the spanning tree
protocol to prevent loops, their ability to manage differing traffic types via quality of
service (Qos), and to segregate traffic with VLANs. A LAN can include a wide variety of
network devices such as switches, firewalls, routers, load balancers, and sensors.
LANs can maintain connections with other LANs via leased lines, leased services, or the
Internet using virtual private network technologies. Depending on how the connections are
established and secured in a LAN, and the distance involved, a LAN may also be
classified as a metropolitan area network (MAN) or a wide area network (WAN).
2. Give short notes about BAUDOT code and illustrate with an example?
Data are communicated between digital computers as sequences of bits. To provide
meaning to a sequence of bits, the bits are grouped to form a data character and an
encoding scheme, or translation table, is provided to allow a computer system to translate
each group of bits into a character. The ideal encoding scheme will provide a unique code
for every possible character to be communicated and stored in the computer, but this
requires that each group have a sufficient number of bits for each data character.
A code used early in the data communications industry is the Baudot code. Baudot uses
five bits per character, thus allowing up to 32 distinct characters. As a technique used to
extend this limitation, the code uses up-shift and down-shift modes as is used on a
typewriter. In the Baudot code, each five bits transmitted must be interpreted according to
whether they are up-shifted (figures) or down-shifted (letters). For example, the bit pattern
11111 represents up-shift and the bit pattern 11011 represents down-shift characters. All
characters transmitted after the sequence 11111 but before the shifted sequence 11011 are
treated as up-shift characters. All characters transmitted after the sequence 11011 are
treated as down-shift characters until the pattern 11111 is recognized. The complete
BAUDOT code (modified for this problem) is shown in the table at the end of this
problem.
Input
The input consists of two parts. The first part is the Baudot character set: line one contains
the 32 down-shift characters and line two contains the 32 up-shift characters. (Note:
spaces are inserted for the shift characters.) The remainder of the input file consists of one
or more messages encoded using the Baudot code. Each message will be on a line in the
input file. Each line will consist of 1's and 0's, with no characters between the bits. There
can be up to 80 bits per message. The input file will be terminated by end-of-file. The
initial state of each message should be assumed to be in the down-shift state.
Output
The output should consist of one line of text for each message. The output should contain
the character representation, as translated using BAUDOT, of each of the messages.
Sample Input
<T*O HNM=LRGIPCVEZDBSYFXAWJ UQK
>5@9 %,.+)4&80:;3"$?#6!/-2' 71(
100100110011000010011111101110000111110111101
001100001101111001001111100001001100010001100110111100000111
Sample Output
DIAL:911
NOV 5, 8AM
Bit Pattern
Down-Shift Character
Up-Shift Character
00000
<
>
00001
T
5
00010
*
@
00011
O
9
00100
Space
Space
00101
H
%
00110
N
,
00111
M
.
01000
=
+
01001
L
)
01010
R
4
01011
G
&
01100
I
8
01101
P
0
01110
C
:
01111
V
;
10000
E
3
10001
Z
"
10010
D
$
10011
B
?
10100
S
#
10101
Y
6
10110
F
!
10111
X
/
11000
A
-
11001
W
2
11010
J
'
11011
Shift Down
Shift Down
11100
U
7
11101
Q
1
11110
K
(
11111
Shift Up
Shift Up
Table: The BAUDOT code
3. Explain in detail about error detection and correction?
In information theory and coding theory with applications in computer science and
telecommunication, error detection and correction or error control are techniques that
enable reliable delivery of digital data over unreliable communication channels. Many
communication channels are subject to channel noise, and thus errors may be
introduced during transmission from the source to a receiver. Error detection
techniques allow detecting such errors, while error correction enables reconstruction
of the original data.
Error correction may generally be realized in two different ways:
Automatic repeat request (ARQ) (sometimes also referred to as backward error
correction): This is an error control technique whereby an error detection scheme is
combined with requests for retransmission of erroneous data. Every block of data
received is checked using the error detection code used, and if the check fails,
retransmission of the data is requested – this may be done repeatedly, until the data
can be verified.
Forward error correction (FEC): The sender encodes the data using an error-correcting
code (ECC) prior to transmission. The additional information (redundancy) added by
the code is used by the receiver to recover the original data. In general, the
reconstructed data is what is deemed the "most likely" original data.
ARQ and FEC may be combined, such that minor errors are corrected without
retransmission, and major errors are corrected via a request for retransmission: this is
called hybrid automatic repeat-request (HARQ).
Error detection schemes
Error detection is most commonly realized using a suitable hash function (or
checksum algorithm). A hash function adds a fixed-length tag to a message, which
enables receivers to verify the delivered message by recomputing the tag and
comparing it with the one provided.
There exists a vast variety of different hash function designs. However, some are of
particularly widespread use because of either their simplicity or their suitability for
detecting certain kinds of errors (e.g., the cyclic redundancy check's performance in
detecting burst errors).
Random-error-correcting codes based on minimum distance coding can provide a
suitable alternative to hash functions when a strict guarantee on the minimum number
of errors to be detected is desired. Repetition codes, described below, are special cases
of error-correcting codes: although rather inefficient, they find applications for both
error correction and detection due to their simplicity.
Repetition codes
A repetition code is a coding scheme that repeats the bits across a channel to achieve
error-free communication. Given a stream of data to be transmitted, the data is divided
into blocks of bits. Each block is transmitted some predetermined number of times.
For example, to send the bit pattern "1011", the four-bit block can be repeated three
times, thus producing "1011 1011 1011". However, if this twelve-bit pattern was
received as "1010 1011 1011" – where the first block is unlike the other two – it can
be determined that an error has occurred.
Repetition codes are very inefficient, and can be susceptible to problems if the error
occurs in exactly the same place for each group (e.g., "1010 1010 1010" in the
previous example would be detected as correct). The advantage of repetition codes is
that they are extremely simple, and are in fact used in some transmissions of numbers
stations.
Parity bits
A parity bit is a bit that is added to a group of source bits to ensure that the number of
set bits (i.e., bits with value 1) in the outcome is even or odd. It is a very simple
scheme that can be used to detect single or any other odd number (i.e., three, five,
etc.) of errors in the output. An even number of flipped bits will make the parity bit
appear correct even though the data is erroneous.
Extensions and variations on the parity bit mechanism are horizontal redundancy
checks, vertical redundancy checks, and "double," "dual," or "diagonal" parity (used
in RAID-DP).
Checksums
A checksum of a message is a modular arithmetic sum of message code words of a
fixed word length (e.g., byte values). The sum may be negated by means of aones'-
complement operation prior to transmission to detect errors resulting in all-zero
messages.
Checksum schemes include parity bits, check digits, and longitudinal redundancy
checks. Some checksum schemes, such as the Damm algorithm, the Luhn algorithm,
and the Verhoeff algorithm, are specifically designed to detect errors commonly
introduced by humans in writing down or remembering identification numbers.
Cyclic redundancy checks (CRCs)
A cyclic redundancy check (CRC) is a single-burst-error-detecting cyclic code and
non-secure hash function designed to detect accidental changes to digital data in
computer networks. It is not suitable for detecting maliciously introduced errors. It is
characterized by specification of a so-called generator polynomial, which is used as
the divisor in a polynomial long division over a finite field, taking the input data as
the dividend, and where the remainder becomes the result.
Cyclic codes have favorable properties in that they are well suited for detecting burst
errors. CRCs are particularly easy to implement in hardware, and are therefore
commonly used in digital networks and storage devices such as hard disk drives.
Even parity is a special case of a cyclic redundancy check, where the single-bit CRC
is generated by the divisor x + 1.
Cryptographic hash functions
The output of a cryptographic hash function, also known as a message digest, can
provide strong assurances about data integrity, whether changes of the data are
accidental (e.g., due to transmission errors) or maliciously introduced. Any
modification to the data will likely be detected through a mismatching hash value.
Furthermore, given some hash value, it is infeasible to find some input data (other
than the one given) that will yield the same hash value. If an attacker can change not
only the message but also the hash value, then a keyed hash or message authentication
code (MAC) can be used for additional security. Without knowing the key, it is
infeasible for the attacker to calculate the correct keyed hash value for a modified
message.
Error-correcting codes
Any error-correcting code can be used for error detection. A code with minimum
Hamming distance, d, can detect up to d − 1 errors in a code word. Using minimumdistance-based error-correcting codes for error detection can be suitable if a strict limit
on the minimum number of errors to be detected is desired.
Codes with minimum Hamming distance d = 2 are degenerate cases of errorcorrecting codes, and can be used to detect single errors. The parity bit is an example
of a single-error-detecting code.
Error correction
Automatic repeat request
Automatic Repeat request (ARQ) is an error control method for data transmission that
makes use of error-detection codes, acknowledgment and/or negative
acknowledgment messages, and timeouts to achieve reliable data transmission. An
acknowledgment is a message sent by the receiver to indicate that it has correctly
received a data frame.
Usually, when the transmitter does not receive the acknowledgment before the
timeout occurs (i.e., within a reasonable amount of time after sending the data frame),
it retransmits the frame until it is either correctly received or the error persists beyond
a predetermined number of retransmissions.
Three types of ARQ protocols are Stop-and-wait ARQ, Go-Back-N ARQ, and
Selective Repeat ARQ.
ARQ is appropriate if the communication channel has varying or unknown capacity,
such as is the case on the Internet. However, ARQ requires the availability of a back
channel, results in possibly increased latency due to retransmissions, and requires the
maintenance of buffers and timers for retransmissions, which in the case of network
congestion can put a strain on the server and overall network capacity.
Error-correcting code
An error-correcting code (ECC) or forward error correction (FEC) code is a system of
adding redundant data, or parity data, to a message, such that it can be recovered by a
receiver even when a number of errors (up to the capability of the code being used)
were introduced, either during the process of transmission, or on storage. Since the
receiver does not have to ask the sender for retransmission of the data, a back-channel
is not required in forward error correction, and it is therefore suitable for simplex
communication such as broadcasting. Error-correcting codes are frequently used in
lower-layer communication, as well as for reliable storage in media such as CDs,
DVDs, hard disks, and RAM.
Error-correcting codes are usually distinguished between convolution codes and block
codes:
Convolution codes are processed on a bit-by-bit basis. They are particularly suitable
for implementation in hardware, and the Viterbi decoder allows optimal decoding.
Block codes are processed on a block-by-block basis. Early examples of block codes
are repetition codes, Hamming codes and multidimensional parity-check codes. They
were followed by a number of efficient codes, Reed–Solomon codes being the most
notable due to their current widespread use. Turbo codes and low-density parity-check
codes (LDPC) are relatively new constructions that can provide almost optimal
efficiency.
Shannon's theorem is an important theorem in forward error correction, and describes
the maximum information rate at which reliable communication is possible over a
channel that has a certain error probability or signal-to-noise ratio (SNR). This strict
upper limit is expressed in terms of the channel capacity. More specifically, the
theorem says that there exist codes such that with increasing encoding length the
probability of error on a discrete memory less channel can be made arbitrarily small,
provided that the code rate is smaller than the channel capacity. The code rate is
defined as the fraction k/n of k source symbols and n encoded symbols.
The actual maximum code rate allowed depends on the error-correcting code used,
and may be lower. This is because Shannon's proof was only of existential nature, and
did not show how to construct codes which are both optimal and have efficient
encoding and decoding algorithms.
Hybrid schemes
Hybrid ARQ is a combination of ARQ and forward error correction. There are two
basic approaches:
Messages are always transmitted with FEC parity data (and error-detection
redundancy). A receiver decodes a message using the parity information, and requests
retransmission using ARQ only if the parity data was not sufficient for successful
decoding (identified through a failed integrity check).
Messages are transmitted without parity data (only with error-detection information).
If a receiver detects an error, it requests FEC information from the transmitter using
ARQ, and uses it to reconstruct the original message.
The latter approach is particularly attractive on an erasure channel when using a rate
less erasure code.
Applications
Applications that require low latency (such as telephone conversations) cannot use
Automatic Repeat reQuest (ARQ); they must use Forward Error Correction (FEC). By
the time an ARQ system discovers an error and re-transmits it, the re-sent data will
arrive too late to be any good.
Applications where the transmitter immediately forgets the information as soon as it is
sent (such as most television cameras) cannot use ARQ; they must use FEC because
when an error occurs, the original data is no longer available. (This is also why FEC is
used in data storage systems such as RAID and distributed data store).
Applications that use ARQ must have a return channel. Applications that have no
return channel cannot use ARQ.
Applications that require extremely low error rates (such as digital money transfers)
must use ARQ.
Internet
In a typical TCP/IP stack, error control is performed at multiple levels:
Each Ethernet frame carries a CRC-32checksum. Frames received with incorrect
checksums are discarded by the receiver hardware.
The IPv4 header contains a checksum protecting the contents of the header. Packets
with mismatching checksums are dropped within the network or at the receiver.
The checksum was omitted from the IPv6 header in order to minimize processing
costs in network routing and because current link layer technology is assumed to
provide sufficient error detection (see also RFC 3819).
UDP has an optional checksum covering the payload and addressing information from
the UDP and IP headers. Packets with incorrect checksums are discarded by the
operating system network stack. The checksum is optional under IPv4, only, because
the Data-Link layer checksum may already provide the desired level of error
protection.
TCP provides a checksum for protecting the payload and addressing information from
the TCP and IP headers. Packets with incorrect checksums are discarded within the
network stack, and eventually get retransmitted using ARQ, either explicitly (such as
through triple-ack) or implicitly due to a timeout.
Deep-space telecommunications
Development of error-correction codes was tightly coupled with the history of deepspace missions due to the extreme dilution of signal power over interplanetary
distances, and the limited power availability aboard space probes. Whereas early
missions sent their data uncoded, starting from 1968 digital error correction was
implemented in the form of (sub-optimally decoded) convolutional codes and Reed–
Muller codes. The Reed–Muller code was well suited to the noise the spacecraft was
subject to (approximately matching a bell curve), and was implemented at the Mariner
spacecraft for missions between 1969 and 1977.
The Voyager 1 and Voyager 2 missions, which started in 1977, were designed to
deliver color imaging amongst scientific information of Jupiter and Saturn. This
resulted in increased coding requirements, and thus the spacecraft were supported by
(optimally Viterbi-decoded) convolutional codes that could be concatenated with an
outer Golay (24,12,8) code. The Voyager 2 probe additionally supported an
implementation of a Reed–Solomon code: the concatenated Reed–Solomon–Viterbi
(RSV) code allowed for very powerful error correction, and enabled the spacecraft's
extended journey to Uranus and Neptune.
The CCSDS currently recommends usage of error correction codes with performance
similar to the Voyager 2 RSV code as a minimum. Concatenated codes are
increasingly falling out of favor with space missions, and are replaced by more
powerful codes such as Turbo codes or LDPC codes.
The different kinds of deep space and orbital missions that are conducted suggest that
trying to find a "one size fits all" error correction system will be an ongoing problem
for some time to come. For missions close to earth the nature of the channel noise is
different from that of a spacecraft on an interplanetary mission experiences.
Additionally, as a spacecraft increases its distance from earth, the problem of
correcting for noise gets larger.
Satellite broadcasting (DVB)
The demand for satellite transponder bandwidth continues to grow, fueled by the
desire to deliver television (including new channels and High Definition TV) and IP
data. Transponder availability and bandwidth constraints have limited this growth,
because transponder capacity is determined by the selected modulation scheme and
Forward error correction (FEC) rate.
Overview
QPSK coupled with traditional Reed Solomon and Viterbi codes have been used for
nearly 20 years for the delivery of digital satellite TV.
Higher order modulation schemes such as 8PSK, 16QAM and 32QAM have enabled
the satellite industry to increase transponder efficiency by several orders of
magnitude.
This increase in the information rate in a transponder comes at the expense of an
increase in the carrier power to meet the threshold requirement for existing antennas.
Tests conducted using the latest chipsets demonstrate that the performance achieved
by using Turbo Codes may be even lower than the 0.8 dB figure assumed in early
designs.
Data storage
Error detection and correction codes are often used to improve the reliability of data
storage media. A "parity track" was present on the first magnetic tape data storage in
1951. The "Optimal Rectangular Code" used in group code recording tapes not only
detects but also corrects single-bit errors.
Some file formats, particularly archive formats, include a checksum (most often
CRC32) to detect corruption and truncation and can employ redundancy and/or parity
files to recover portions of corrupted data.
Reed Solomon codes are used in compact discs to correct errors caused by scratches.
Modern hard drives use CRC codes to detect and Reed–Solomon codes to correct
minor errors in sector reads, and to recover data from sectors that have "gone bad" and
store that data in the spare sectors.
RAID systems use a variety of error correction techniques, to correct errors when a
hard drive completely fails.
Systems such as ZFS and some RAID support data scrubbing and recovering, which
allows bad blocks to be detected and (hopefully) recovered before they are used. The
recovered data may be re-written to exactly the same physical location, to spare
blocks elsewhere on the same piece of hardware, or to replacement hardware.
Error-correcting memory
DRAM memory may provide increased protection against soft errors by relying on
error correcting codes. Such error-correcting memory, known as ECC or EDACprotected memory is particularly desirable for high fault-tolerant applications, such as
servers, as well as deep-space applications due to increased radiation.
Error-correcting memory controllers traditionally use Hamming codes, although some
use triple modular redundancy.
Interleaving allows distributing the effect of a single cosmic ray potentially upsetting
multiple physically neighboring bits across multiple words by associating neighboring
bits to different words. As long as a single event upset (SEU) does not exceed the
error threshold (e.g., a single error) in any particular word between accesses, it can be
corrected (e.g., by a single-bit error correcting code), and the illusion of an error-free
memory system may be maintained.
A few systems support memory scrubbing.
4. Give a brief note on ISDN?
Integrated Services Digital Network (ISDN) is a set of communication standards for
simultaneous digital transmission of voice, video, data, and other network services
over the traditional circuits of the public switched telephone network. It was first
defined in 1988 in the CCITT red book. Prior to ISDN, the telephone system was
viewed as a way to transport voice, with some special services available for data. The
key feature of ISDN is that it integrates speech and data on the same lines, adding
features that were not available in the classic telephone system. There are several
kinds of access interfaces to ISDN defined as Basic Rate Interface (BRI), Primary
Rate Interface (PRI), Narrowband ISDN (N-ISDN), and Broadband ISDN (B-ISDN).
ISDN is a circuit-switched telephone network system, which also provides access to
packet switched networks, designed to allow digital transmission of voice and data
over ordinary telephone copper wires, resulting in potentially better voice quality than
an analog phone can provide. It offers circuit-switched connections (for either voice
or data), and packet-switched connections (for data), in increments of 64 kilobit/s. A
major market application for ISDN in some countries is Internet access, where ISDN
typically provides a maximum of 128 kbit/s in both upstream and downstream
directions. Channel bonding can achieve a greater data rate; typically the ISDN Bchannels of three or four BRIs (six to eight 64 Kbit/s channels) are bonded.
ISDN should not be mistaken for its use with a specific protocol, such as Q.931
whereby ISDN is employed as the network, data-link and physical layers in the
context of the OSI model. In a broad sense ISDN can be considered a suite of digital
services existing on layers 1, 2, and 3 of the OSI model. ISDN is designed to provide
access to voice and data services simultaneously.
However, common use reduced ISDN to be limited to Q.931 and related protocols,
which are a set of protocols for establishing and breaking circuit switched
connections, and for advanced calling features for the user. They were introduced in
1986.
In a videoconference, ISDN provides simultaneous voice, video, and text transmission
between individual desktop videoconferencing systems and group (room)
videoconferencing systems.
ISDN elements
An integrated service refers to ISDN's ability to deliver at minimum two simultaneous
connections, in any combination of data, voice, video, and fax, over a single line.
Multiple devices can be attached to the line, and used as needed. That means an ISDN
line can take care of most people's complete communications needs (apart from
broadband Internet access and entertainment television) at a much higher transmission
rate, without forcing the purchase of multiple analog phone lines. It also refers to
integrated switching and transmission in that telephone switching and carrier wave
transmission are integrated rather than separate as in earlier technology.
Basic Rate Interface
Main article: Basic Rate Interface
The entry level interface to ISDN is the Basic(s) Rate Interface (BRI), a 128 kbit/s
service delivered over a pair of standard telephone copper wires. The 128 kbit/s
payload rate is broken down into two 64 kbit/s bearer channels ('B' channels) and one
16 kbit/s signaling channel ('D' channel or data channel). This is sometimes referred to
as 2B+D.
The interface specifies the following network interfaces:
The U interface is a two-wire interface between the exchange and a network
terminating unit, which is usually the demarcation point in non-North American
networks.
The T interface is a serial interface between a computing device and a terminal
adapter, which is the digital equivalent of a modem.
The S interface is a four-wire bus that ISDN consumer devices plug into; the S & T
reference points are commonly implemented as a single interface labeled 'S/T' on an
Network termination 1 (NT1).
The R interface defines the point between a non-ISDN device and a terminal adapter
(TA) which provides translation to and from such a device.
BRI-ISDN is very popular in Europe but is much less common in North America. It is
also common in Japan — where it is known as INS64.
Primary Rate Interface
The other ISDN access available is the Primary Rate Interface (PRI), which is carried
over an E1 (2048 kbit/s) in most parts of the world. An E1 is 30 'B' channels of 64
kbit/s, one 'D' channel of 64 kbit/s and a timing and alarm channel of 64 kbit/s.
In North America PRI service is delivered on one or more T1 carriers (often referred
to as 23B+D) of 1544 kbit/s (24 channels). A PRI has 23 'B' channels and 1 'D'
channel for signalling (Japan uses a circuit called a J1, which is similar to a T1). Interchangeably but incorrectly, a PRI is referred to as T1 because it uses the T1 carrier
format. A true T1 (commonly called "Analog T1" to avoid confusion) uses 24
channels of 64 kbit/s of in-band signaling. Each channel uses 56 kb for data and voice
and 8 kb for signaling and messaging. PRI uses out of band signaling which provides
the 23 B channels with clear 64 kb for voice and data and one 64 kb 'D' channel for
signaling and messaging. In North America, Non-Facility Associated Signalling
allows two or more PRIs to be controlled by a single D channel, and is sometimes
called "23B+D + n*24B". D-channel backup allows for a second D channel in case
the primary fails. NFAS is commonly used on a T3.
PRI-ISDN is popular throughout the world, especially for connecting PBXs to PSTN.
While the North American PSTN can use PRI or Analog T1 format from PBX to
PBX, the POTS or BRI can be delivered to a business or residence. North American
PSTN can connect from PBX to PBX via Analog T1, T3, PRI, OC3, etc...
Even though many network professionals use the term "ISDN" to refer to the lowerbandwidth BRI circuit, in North America BRI is relatively uncommon whilst PRI
circuits serving PBXs are commonplace.
Bearer channels
The bearer channel (B) is a standard 64 kbit/s voice channel of 8 bits sampled at
8 kHz with G.711 encoding. B-Channels can also be used to carry data, since they are
nothing more than digital channels.
Each one of these channels is known as a DS0.
Most B channels can carry a 64 kbit/s signal, but some were limited to 56K because
they traveled over RBS lines. This was commonplace in the 20th century, but has
since become less so.
Signaling channel
The signaling channel (D) uses Q.931 for signaling with the other side of the link.
X.25
X.25 can be carried over the B or D channels of a BRI line, and over the B channels
of a PRI line. X.25 over the D channel is used at many point-of-sale (credit card)
terminals because it eliminates the modem setup, and because it connects to the
central system over a B channel, thereby eliminating the need for modems and
making much better use of the central system's telephone lines.
X.25 was also part of an ISDN protocol called "Always On/Dynamic ISDN", or
AO/DI. This allowed a user to have a constant multi-link PPP connection to the
internet over X.25 on the D channel, and brought up one or two B channels as needed.
Frame Relay
In theory, Frame Relay can operate over the D channel of BRIs and PRIs, but it is
seldom, if ever, used.
Consumer and industry perspectives
There are two points of view into the ISDN world. The most common is that of the
end-user, who wants a digital connection into the telephone network from home,
whose performance would be better than a 20th-century analog 56K modem
connection. Discussion on the merits of various ISDN modems, carriers' offerings and
tariffs (features, pricing) are from this perspective. Since the principal consumer
application is for Internet access, ISDN was mostly superseded by DSL in the early
21st century. Inexpensive ADSL service offers speeds up to 5 Mbit/s, while more
expensive versions are improving in speed all the time. By 2010, ADSL speeds of
several millions of bits per second had become commonplace, thus making ISDN
obsolete before it fairly started.
There is a second viewpoint: that of the telephone industry, where ISDN is a core
technology. A telephone network can be thought of as a collection of wires strung
between switching systems. The common electrical specification for the signals on
these wires is T1 or E1. Between telephone company switches, the signaling is
performed via SS7. Normally, a PBX is connected via a T1 with robbed bit signaling
to indicate on-hook or off-hook conditions and MF and DTMF tones to encode the
destination number. ISDN is much better because messages can be sent much more
quickly than by trying to encode numbers as long (100 ms per digit) tone sequences.
This results in faster call setup times. Also, a greater number of features are available
and fraud is reduced.
ISDN is also used as a smart-network technology intended to add new services to the
public switched telephone network (PSTN) by giving users direct access to end-to-end
circuit-switched digital services and as a backup or failsafe circuit solution for critical
use data circuits.
ISDN and broadcast industry
ISDN is used heavily by the broadcast industry as a reliable way of switching low
latency, high quality, long distance audio circuits. In conjunction with an appropriate
codec using MPEG or various manufacturers proprietary algorithms, an ISDN BRI
can be used to send stereo bi-directional audio coded at 128 Kbit/s with 20 Hz –
20 kHz audio bandwidth, although commonly the G.722 algorithm is used with a
single 64 kbit/s B channel to send much lower latency mono audio at the expense of
audio quality. Where very high quality audio is required multiple ISDN BRIs can be
used in parallel to provide a higher bandwidth circuit switched connection. BBC
Radio 3 commonly makes use of three ISDN BRIs to carry 320 Kbit/s audio stream
for live outside broadcasts. ISDN BRI services are used to link remote studios, sports
grounds and outside broadcasts into the main broadcast studio. ISDN via satellite is
used by field reporters around the world. It's also common to use ISDN for the return
audio links to remote satellite broadcast vehicles.
In many countries, such as the UK and Australia, ISDN has displaced the older
technology of equalized analogue landlines, with these circuits being phased out by
telecommunications providers. IP based streaming codec’s are starting to gain a
foothold in the broadcast sector, using broadband internet to connect remote studios.
However reliability and latency is crucially important for broadcasters and the quality
of service offered by ISDN has not yet been matched by packet switched alternatives.
Countries
United States and Canada
ISDN-BRI never gained popularity as a general use telephone access technology in
Canada and the US, and remains a niche product. The service was seen as a solution
in search of a problem,[8] and the extensive array of options and features were difficult
for customers to understand and use. ISDN has long been known by derogatory
acronyms highlighting these issues, such as It Still Does Nothing, Innovations
Subscribers Don't Need, and I Still Don't know.
Once the concept of broadband Internet access came to be associated with data rates
incoming to the customer at 256 Kbit/s or more,[a] and alternatives like ADSL grew in
popularity, the consumer market for BRI did not develop. Its only remaining
advantage is that while ADSL has a functional distance limitation and can use ADSL
loop extenders, BRI has a greater limit and can use repeaters. As such, BRI may be
acceptable for customers who are too remote for ADSL. Widespread use of BRI is
further stymied by some small North American CLECs such as CenturyTel having
given up on it and not providing Internet access using it. [14] However, AT&T in most
states (especially the former SBC/SWB territory) will still install an ISDN BRI line
anywhere a normal analog line can be placed and the monthly charge is roughly $55.
ISDN-BRI is currently primarily used in industries with specialized and very specific
needs. High-end videoconferencing hardware made by companies such as Sony,
Polycom, Tandberg, and Life-size via the Life-size Networker[15] can bond up to 8 Bchannels together (using a BRI circuit for every 2 channels) to provide digital, circuitswitched video connections to almost anywhere in the world. This is very expensive,
and is being replaced by IP-based conferencing, but where concern cost is less of an
issue than predictable quality and where a QoS-enabled IP does not exist, BRI is the
preferred choice.
Most modern non-VoIP PBXs use ISDN-PRI circuits. These are connected via T1
lines with the central office switch, replacing older analog two-way and direct inward
dialing (DID) trunks. PRI is capable of delivering Calling Line Identification (CLID)
in both directions so that the telephone number of an extension, rather than a
company's main number, can be sent. It is still commonly used in recording studios,
when a voice-over actor is in one studio, but the director and producer are in a studio
at another location.[4] The ISDN protocol delivers channelized, not-over-the-Internet
service, powerful call setup and routing features, faster setup and tear down, superior
audio fidelity as compared to POTS (plain old telephone service), lower delay and, at
higher densities, lower cost.
In 2013, Verizon announced it would no longer take orders for ISDN service in the
Northeastern United States.
India
Bharat Sanchar Nigam Limited, Reliance Communications and Bharti Airtel are the
largest communication service providers, and offer both ISDN BRI and PRI services
across the country. Reliance Communications and Bharti Airtel uses the DLC
technology for providing these services. With the introduction of broadband
technology, the load on bandwidth is being absorbed by ADSL. ISDN continues to be
an important backup network for point-to-point leased line customers such as banks,
Eseva Centers,[16]Life Insurance Corporation of India, and SBI ATMs.
Japan
On April 19, 1988, Japanese telecommunications company NTT began offering
nationwide ISDN services trademarked INS Net 64, and INS Net 1500, a fruition of
NTT's independent research and trial from the 1970s of what it referred to the INS
(Information Network System).
Previously, on April 1985, Japanese digital telephone exchange hardware made by
Fujitsu was used to experimentally deploy the world's first I interface ISDN. The I
interface, unlike the older and incompatible Y interface, is what modern ISDN
services use today.
Since 2000, NTT's ISDN offering have been known as FLET's ISDN, incorporating
the "FLET's" brand that NTT uses for all of its ISP offerings.
In Japan, the number of ISDN subscribers dwindled as alternative technologies such
as ADSL, cable Internet access, and fiber to the home gained greater popularity. On
November 2, 2010, NTT announced plans to migrate their backend from PSTN to the
IP network from around 2020 to around 2025. For this migration, ISDN services will
be retired, and fiber optic services are recommended as an alternative.
United Kingdom
In the United Kingdom, British Telecom (BT) provides ISDN2e (BRI) as well as
ISDN30 (PRI). Until April 2006, they also offered services named Home Highway
and Business Highway, which were BRI ISDN-based services that offered integrated
analogue connectivity as well as ISDN. Later versions of the Highway products also
included built-in Universal serial bus (USB) sockets for direct computer access. Home
Highway was bought by many home users, usually for Internet connection, although
not as fast as ADSL, because it was available before ADSL and in places where
ADSL does not reach.
France
France Telecom offers ISDN services under their product name Numeris (2 B+D), of
which a professional Duo and home Itoo version is available. ISDN is generally
known as RNIS in France and has widespread availability. The introduction of ADSL
is reducing ISDN used for data transfer and Internet access, although it is still
common in more rural and outlying areas, and for applications such as business voice
and point-of-sale terminals.
Germany
German stamp
In Germany, ISDN is very popular with an installed base of 25 million channels (29%
of all subscriber lines in Germany as of 2003 and 20% of all ISDN channels
worldwide). Due to the success of ISDN, the number of installed analog lines is
decreasing. Deutsche Telekom (DTAG) offers both BRI and PRI. Competing phone
companies often offer ISDN only and no analog lines. However, these operators
generally offer free hardware that also allows the use of POTS equipment, such as
NTBAs with integrated terminal adapters. Because of the widespread availability of
ADSL services, ISDN is today primarily used for voice and fax traffic, but is still very
popular thanks to the pricing policy of German telecommunication providers.
Today ISDN (BRI) and ADSL/VDSL are often bundled on the same line, mainly
because the combination of ADSL with an analog line has no cost advantage over a
combined ISDN-ADSL line. Some German operators started to implement Next
Generation Networking, generally realized via DSL and unbundled local loop.
However, a few operators offer the same services via the cable television
infrastructure or, in selected areas, via FTTH. Because of the popularity of ISDN,
virtually all these telecommunication providers bundle their products with residential
gateways that include both integrated analog telephony adapters and ISDN-NGN
adapters.
Greece
OTE, the incumbent telecommunications operator, offers ISDN BRI (BRA) services
in Greece. Following the launch of ADSL in 2003, the importance of ISDN for data
transfer began to decrease and is today limited to niche business applications with
point-to-point requirements.
5. Describe in detail about MODEM?
A modem is a device or program that enables a computer to transmit data over, for
example, telephone or cable lines. Computer information is stored digitally, whereas
information transmitted over telephone lines is transmitted in the form of analog
waves. A modem converts between these two forms.
Fortunately, there is one standard interface for connecting external modems to
computers called RS-232. Consequently, any external modem can be attached to any
computer that has an RS-232 port, which almost all personal computers have. There
are also modems that come as an expansion board that you can insert into a vacant
expansion slot. These are sometimes called onboard or internal modems.
While the modem interfaces are standardized, a number of different protocols for
formatting data to be transmitted over telephone lines exist. Some, like CCITTV.34,
are official standards, while others have been developed by private companies. Most
modems have built-in support for the more common protocols -- at slow data
transmission speeds at least, most modems can communicate with each other. At high
transmission speeds, however, the protocols are less standardized.
Aside from the transmission protocols that they support, the following characteristics
distinguish one modem from another:
bps: How fast the modem can transmit and receive data. At slow rates, modems are
measured in terms of baud rates. The slowest rate is 300 baud (about 25 cps). At
higher speeds, modems are measured in terms of bits per second (bps). The fastest
modems run at 57,600 bps, although they can achieve even higher data transfer rates
by compressing the data. Obviously, the faster the transmission rate, the faster you
can send and receive data. Note, however, that you cannot receive data any faster than
it is being sent. If, for example, the device sending data to your computer is sending it
at 2,400 bps, you must receive it at 2,400 bps. It does not always pay, therefore, to
have a very fast modem. In addition, some telephone lines are unable to transmit data
reliably at very high rates.
Voice/data: Many modems support a switch to change between voice and data
modes. In data mode, the modem acts like a regular modem. In voice mode, the
modem acts like a regular telephone. Modems that support a voice/data switch have a
built-in loudspeaker and microphone for voice communication.
Auto-answer: An auto-answer modem enables your computer to receive calls in your
absence. This is only necessary if you are offering some type of computer service that
people can call in to use.
data compression: Some modems perform data compression, which enables them to
send data at faster rates. However, the modem at the receiving end must be able to
decompress the data using the same compression technique.
Flash memory: Some modems come with flash memory rather than conventional
ROM, which means that the communications protocolscan be easily updated if
necessary.
Fax capability: Most modern modems are fax modems, which means that they can
send and receive faxes.
Acoustic coupler modem
A modem (modulator-demodulator) is a device that modulates an analog carrier
signal to encode digital information, and also demodulates such a carrier signal to
decode the transmitted information. The goal is to produce a signal that can be
transmitted easily and decoded to reproduce the original digital data. Modems can be
used with any means of transmitting analog signals, from light emitting diodes to
radio. The most familiar example is a voice band modem that turns the digital data of
a personal computer into modulated electrical signals in the voice frequency range of
a telephone channel. These signals can be transmitted over telephone lines and
demodulated by another modem at the receiver side to recover the digital data.
Modems are generally classified by the amount of data they can send in a given unit
of time, usually expressed in bits per second (bit/s, or bps), or bytes per second (B/s).
Modems can alternatively be classified by their symbol rate, measured in baud. The
baud unit denotes symbols per second, or the number of times per second the modem
sends a new signal. For example, the ITU V.21 standard used audio frequency shift
keying with two possible frequencies corresponding to two distinct symbols (or one
bit per symbol), to carry 300 bits per second using 300 baud. By contrast, the original
ITU V.22 standard, which could transmit and receive four distinct symbols (two bits
per symbol), handled 1,200 bit/s by sending 600 symbols per second (600 baud) using
phase shift keying.
Tele Guide terminal
News wire services in the 1920s used multiplex devices that satisfied the definition of
a modem. However the modem function was incidental to the multiplexing function,
so they are not commonly included in the history of modems.
Modems grew out of the need to connect teleprinters over ordinary phone lines
instead of the more expensive leased lines which had previously been used for current
loop–based tele printers and automated telegraphs. In 1942, IBM adapted this
technology to their unit record equipment and were able to transmit punched cards at
25 bits/second.[citation needed]
Mass-produced modems in the United States began as part of the SAGE air-defense
system in 1958 (the year the word modem was first used[1]), connecting terminals at
various airbases, radar sites, and command-and-control centers to the SAGE director
centers scattered around the U.S. and Canada. SAGE modems were described by
AT&T's Bell Labs as conforming to their newly published Bell 101 dataset standard.
While they ran on dedicated telephone lines, the devices at each end were no different
from commercial acoustically coupled Bell 101, 110 baud modems.
In summer 1960[citation needed], the name Data-Phone was introduced to replace the
earlier term digital subset. The 202 Data-Phone was a half-duplex asynchronous
service that was marketed extensively in late 1960.In 1962the 201A and 201B DataPhones were introduced. They were synchronous modems using two-bit-per-baud
phase-shift keying (PSK). The 201A operated half-duplex at 2,000 bit/s over normal
phone lines, while the 201B provided full duplex 2,400 bit/s service on four-wire
leased lines, the send and receive channels each running on their own set of two wires.
The famous Bell 103A dataset standard was also introduced by AT&T in 1962. It
provided full-duplex service at 300 bit/s over normal phone lines. Frequency-shift
keying was used, with the call originator transmitting at 1,070 or 1,270 Hz and the
answering modem transmitting at 2,025 or 2,225 Hz. The readily available 103A2
gave an important boost to the use of remote low-speed terminals such as the Teletype
Model 33 ASR and KSR, and the IBM 2741. AT&T reduced modem costs by
introducing the originate-only 113D and the answer-only 113B/C modems.
Acoustic couplers
The Novation CAT acoustically coupled modem
For many years, the Bell System (AT&T) maintained a monopoly on the use of its
phone lines, and what devices could be connected to its lines. However, the seminal
Hush-a-Phone v. FCC case of 1956 concluded that it was within the FCC's
jurisdiction to regulate the operation of the System. Subsequently, the FCC examiner
found that as long as the device was not electrically attached it would not threaten to
degenerate the system. This led to a number of devices that mechanically connected to
the phone, through a standard handset. Since most handsets were supplied from
Western Electric, it was relatively easy to build such an acoustic coupler, and this
style of connection was used for many devices like answering machines.
Acoustically coupled Bell 103A-compatible 300 bit/s modems became common
during the 1970s, with well-known models including the Novation CAT and the
Anderson-Jacobson, the later spun off from an in-house project at Stanford Research
Institute (now SRI International). An even lower-cost option was the Pennywhistle
modem, designed to be built using parts found at electronics scrap and surplus stores.
In December 1972, Vadic introduced the VA3400, which was notable because it
provided full duplex operation at 1,200 bit/s over the phone network. Like the 103A,
it used different frequency bands for transmit and receive. In November 1976, AT&T
introduced the 212A modem to compete with Vadic. It was similar in design to
Vadic's model, but used the lower frequency set for transmission. One could also use
the 212A with a 103A modem at 300 bit/s. According to Vadic, the change in
frequency assignments made the 212 intentionally incompatible with acoustic
coupling, thereby locking out many potential modem manufacturers. In 1977, Vadic
responded with the VA3467 triple modem, an answer-only modem sold to computer
center operators that supported Vadic's 1,200-bit/s mode, AT&T's 212A mode, and
103A operation.
Carterfone and direct connection
The Hush-a-Phone decision applied only to mechanical collections, but the Carterfone
decision of 1968 led to the FCC introducing a rule setting stringent AT&T-designed
tests for electronically coupling a device to the phone lines. AT&T's tests were
complex, making electronically coupled modems expensive, [citation needed] so
acoustically coupled modems remained common into the early 1980s.
However, the rapidly falling prices of electronics in the late 1970s led to an increasing
number of direct-connect models around 1980. In spite of being directly connected,
these modems were generally operated like their earlier acoustic versions - dialling
and other phone-control operations were completed by hand, using an attached
handset. A small number of modems added the ability to automatically answer
incoming calls, or automatically place an outgoing call to a single number, but even
these limited features were relatively rare or limited to special models in a lineup.
When more flexible solutions were needed, 3rd party "diallers" were used to automate
calling, normally using a separate serial port.
The Smart modem and the rise of BBSs
The original model 300 baud Smartmodem
The next major advance in modems was the Hayes Smartmodem, introduced in 1981.
The Smartmodem was an otherwise standard 103A 300-bit/s modem, but it was
attached to a small microcontroller that let the computer send it commands. The
command set included instructions for picking up and hanging up the phone, dialing
numbers, and answering calls. This eliminated the need for any manual operation, a
handset, or a dialler. Terminal programs that maintained lists of phone numbers and
sent the dialing commands became common. The basic Hayes command set remains
the basis for computer control of most modern modems.
The Smartmodem and its clones also aided the spread of bulletin board systems
(BBSs) because it was the first low-cost modem that could answer calls. Modems had
previously been typically either the call-only, acoustically coupled models used on the
client side, or the much more expensive, answer-only models used on the server side.
These were fine for large computer installations, but useless for the hobbiest who
wanted to run a BBS but then periodically use the same telephone line to call other
systems. The first hobby BBS system, CBBS, started as an experiment in ways to
better use the Smartmodem.
Almost all modern modems can inter-operate with fax machines. Digital faxes,
introduced in the 1980s, are simply an image format sent over a high-speed
(commonly 14.4 kbit/s) modem. Software running on the host computer can convert
any image into fax format, which can then be sent using the modem. Such software
was at one time an add-on, but has since become largely universal.
1200 and 2400 bps
The 300 bit/s modems used audio frequency-shift keying to send data. In this system
the stream of 1s and 0s in computer data is translated into sounds which can be easily
sent on the phone lines. In the Bell 103 system, the originating modem sends 0s by
playing a 1,070 Hz tone, and 1s at 1,270 Hz, with the answering modem transmitting
its 0s on 2,025 Hz and 1s on 2,225 Hz. These frequencies were chosen carefully, they
are in the range that suffer minimum distortion on the phone system and not
harmonics of each other.
In the 1,200 bit/s and faster systems, phase-shift keying was used. In this system the
two tones for any one side of the connection are sent at similar frequencies as in the
300 bit/s systems, but slightly out of phase. Voice band modems generally remained
at 300 and 1,200 bit/s (V.21 and V.22) into the mid-1980s. A V.22bis 2,400-bit/s
system similar in concept to the 1,200-bit/s Bell 212 signaling was introduced in the
U.S., and a slightly different one in Europe. The limited available frequency range
meant the symbol rate of 1,200 bit/s modems was still only 600 baud (symbols per
second). The bit rate increases were achieved by defining 4 or 8 distinct symbols,
which allowed the encoding of 2 or 3 bits per symbol instead of only 1. The use of
smaller shifts had the drawback of making each symbols more vulnerable to
interference, but improvements in phone line quality at the same time helped
compensate for this. By the late 1980s, most modems could support all of these
standards and 2,400-bit/s operation was becoming common.
Proprietary standards
Many other standards were also introduced for special purposes, commonly using a
high-speed channel for receiving, and a lower-speed channel for sending. One typical
example was used in the French Minitel system, in which the user's terminals spent
the majority of their time receiving information. The modem in the Minitel terminal
thus operated at 1,200 bit/s for reception, and 75 bit/s for sending commands back to
the servers.
Three U.S. companies became famous for high-speed versions of the same concept.
Telebit introduced its Trailblazer modem in 1984, which used a large number of
36 bit/s channels to send data one-way at rates up to 18,432 bit/s. A single additional
channel in the reverse direction allowed the two modems to communicate how much
data was waiting at either end of the link, and the modems could change direction on
the fly. The Trailblazer modems also supported a feature that allowed them to spoof
the UUCPg protocol, commonly used on Unix systems to send e-mail, and thereby
speed UUCP up by a tremendous amount. Trailblazers thus became extremely
common on Unix systems, and maintained their dominance in this market well into
the 1990s.
USRobotics (USR) introduced a similar system, known as HST, although this
supplied only 9,600 bit/s (in early versions at least) and provided for a larger
backchannel. Rather than offer spoofing, USR instead created a large market among
Fidonet users by offering its modems to BBS sysops at a much lower price, resulting
in sales to end users who wanted faster file transfers. Hayes was forced to compete,
and introduced its own 9,600-bit/s standard, Express 96 (also known as Ping-Pong),
which was generally similar to Telebit's PEP. Hayes, however, offered neither
protocol spoofing nor sysop discounts, and its high-speed modems remained rare.
Echo cancellation, 9600 and 14,400
US Robotics Sportster 14,400 Fax modem (1994)
Echo cancellation was the next major advance in modem design.
Local telephone lines use the same wires to send and receive, which results in a small
amount of the outgoing signal being reflected back. This is useful for people talking
on the phone, as it provides a signal to the speaker that their voice is making it
through the system. However, this reflected signal causes problems for the modem,
which is unable to distinguish between a signal from the remote modem and the echo
of its own signal. This was why earlier modems split the signal frequencies into
"answer" and "originate"; the modem could then ignore any signals in the frequency
range it was using for transmission. Even with improvements to the phone system
allowing higher speeds, this splitting of available phone signal bandwidth still
imposed a half-speed limit on modems.
Echo cancellation eliminated this problem. Measuring the echo delays and magnitudes
allowed the modem to tell if the received signal was from itself or the remote modem,
and create an equal and opposite signal to cancel its own. Modems were then able to
send over the whole frequency spectrum in both directions at the same time, leading
to the development of 4,800 and 9,600 bit/s modems.
Increases in speed have used increasingly complicated communications theory.
Twelve hundred and 2,400 bit/s modems used the phase shift key (PSK) concept. This
could transmit two or three bits per symbol. The next major advance encoded four bits
into a combination of amplitude and phase, known as Quadrature Amplitude
Modulation (QAM).
The new V.27ter and V.32 standards were able to transmit 4 bits per symbol, at a rate
of 1,200 or 2,400 baud, giving an effective bit rate of 4,800 or 9,600 bit/s. The carrier
frequency was 1,650 Hz. For many years, most engineers considered this rate to be
the limit of data communications over telephone networks.
Error correction and compression
Operations at these speeds pushed the limits of the phone lines, resulting in high error
rates. This led to the introduction of error-correction systems built into the modems,
made most famous with Microcom's MNP systems. A string of MNP standards came
out in the 1980s, each increasing the effective data rate by minimizing overhead, from
about 75% theoretical maximum in MNP 1, to 95% in MNP 4. The new method
called MNP 5 added data compression to the system, thereby increasing overall
throughput above the modem's rating. Generally the user could expect an MNP5
modem to transfer at about 130% the normal data rate of the modem. Details of MNP
were later released and became popular on a series of 2,400-bit/s modems, and
ultimately led to the development of V.42 and V.42bis ITU standards. V.42 and
V.42bis were non-compatible with MNP but were similar in concept because they
featured error correction and compression.
Another common feature of these high-speed modems was the concept of fallback, or
speed hunting, allowing them to communicate with less-capable modems. During the
call initiation, the modem would transmit a series of signals and wait for the remote
modem to respond. They would start at high speeds and get progressively slower until
there was a response. Thus, two USR modems would be able to connect at 9,600 bit/s,
but, when a user with a 2,400-bit/s modem called in, the USR would fall back to the
common 2,400-bit/s speed. This would also happen if a V.32 modem and a HST
modem were connected. Because they used a different standard at 9,600 bit/s, they
would fall back to their highest commonly supported standard at 2,400 bit/s. The same
applies to V.32bis and 14,400 bit/s HST modem, which would still be able to
communicate with each other at 2,400 bit/s.
Breaking the 9.6k barrier
In 1980, Gottfried Ungerboeck from IBM Zurich Research Laboratory applied
channel coding techniques to search for new ways to increase the speed of modems.
His results were astonishing but only conveyed to a few colleagues.[2] In 1982, he
agreed to publish what is now a landmark paper in the theory of information
coding.[citation needed] By applying parity check coding to the bits in each symbol, and
mapping the encoded bits into a two-dimensional diamond pattern, Ungerboeck
showed that it was possible to increase the speed by a factor of two with the same
error rate. The new technique was called mapping by set partitions, now known as
trellis modulation.
Error correcting codes, which encode code words (sets of bits) in such a way that they
are far from each other, so that in case of error they are still closest to the original
word (and not confused with another) can be thought of as analogous to sphere
packing or packing pennies on a surface: the further two bit sequences are from one
another, the easier it is to correct minor errors.
V.32bis was so successful that the older high-speed standards had little to recommend
them. USR fought back with a 16,800 bit/s version of HST, while AT&T introduced a
one-off 19,200 bit/s method they referred to as V.32ter, but neither non-standard
modem sold well.
V.34/28.8k and 33.6k
An ISA modem manufactured to conform to the V.34 protocol.
Any interest in these systems was destroyed during the lengthy introduction of the
28,800 bit/s V.34 standard. While waiting, several companies decided to release
hardware and introduced modems they referred to as V.FAST. In order to guarantee
compatibility with V.34 modems once the standard was ratified (1994), the
manufacturers were forced to use more flexible parts, generally a DSP and
microcontroller, as opposed to purpose-designed ASIC modem chips.
Today, the ITU standard V.34 represents the culmination of the joint efforts. It
employs the most powerful coding techniques including channel encoding and shape
encoding. From the mere 4 bits per symbol (9.6 kbit/s), the new standards used the
functional equivalent of 6 to 10 bits per symbol, plus increasing baud rates from 2,400
to 3,429, to create 14.4, 28.8, and 33.6 kbit/s modems. This rate is near the theoretical
Shannon limit. When calculated, the Shannon capacity of a narrowband line is
, with
the (linear) signal-to-noise ratio.
Narrowband phone lines have a bandwidth of 3000 Hz so using
(SNR = 24 dB), the capacity is approximately 24 kbit/s.
Without the discovery and eventual application of trellis modulation, maximum
telephone rates using voice-bandwidth channels would have been limited to
3,429 baud × 4 bit/symbol = approximately 14 kbit/s using traditional QAM.
V.61/V.70 Analog/Digital Simultaneous Voice and Data
The V.61 Standard introduced Analog Simultaneous Voice and Data (ASVD). This
technology allowed users of v.61 modems to engage in point-to-point voice
conversations with each other while their respective modems communicated.
In 1995, the first DSVD (Digital Simultaneous Voice and Data) modems became
available to consumers, and the standard was ratified as v.70 by the International
Telecommunication Union (ITU) in 1996.
Two DSVD modems can establish a completely digital link between each other over
standard phone lines. Sometimes referred to as "the poor man's ISDN", and
employing a similar technology, v.70 compatible modems allow for a maximum
speed of 33.6 kbit/s between peers. By using a majority of the bandwidth for data and
reserving part for voice transmission, DSVD modems allow users to pick up a
telephone handset interfaced with the modem, and initiate a call to the other peer.
One practical use for this technology was realized by early two-player video gamers,
who could hold voice communication with each other over the phone while playing.
Using digital lines and PCM (V.90/92)
Modem bank at an ISP
In the late 1990s Rockwell/Lucent and USRoboticsintroduced new competing
technologies based upon the digital transmission used in modern telephony networks.
The standard digital transmission in modern networks is 64 kbit/s but some networks
use a part of the bandwidth for remote office signaling (e.g. to hang up the phone),
limiting the effective rate to 56 kbit/s DS0. This new technology was adopted into
ITU standards V.90 and is common in modern computers. The 56 kbit/s rate is only
possible from the central office to the user site (downlink). In the United States,
government regulation limits the maximum power output, resulting in a maximum
data rate of 53.3 kbit/s. The uplink (from the user to the central office) still uses V.34
technology at 33.6 kbit/s.
Later in V.92, the digital PCM technique was applied to increase the upload speed to a
maximum of 48 kbit/s, but at the expense of download rates. A 48 kbit/s upstream rate
would reduce the downstream as low as 40 kbit/s due to echo on the telephone line.
To avoid this problem, V.92 modems offer the option to turn off the digital upstream
and instead use a 33.6 kbit/s analog connection, in order to maintain a high digital
downstream of 50 kbit/s or higher.[4] V.92 also adds two other features. The first is the
ability for users who have call waiting to put their dial-up Internet connection on hold
for extended periods[vague] of time while they answer a call. The second feature is the
ability to quickly connect to one's ISP. This is achieved by remembering the analog
and digital characteristics of the telephone line, and using this saved information when
reconnecting.
Using compression to exceed 56k
Today's V.42, V.42bis and V.44 standards allow the modem to transmit data faster
than its basic rate would imply. For instance, a 53.3 kbit/s connection with V.44 can
transmit up to 53.3*6 == 320 kbit/s using pure text. However, the compression ratio
tends to vary due to noise on the line, or due to the transfer of already-compressed
files (ZIP files, JPEG images, MP3 audio, MPEG video). [5] At some points the
modem will be sending compressed files at approximately 50 kbit/s, uncompressed
files at 160 kbit/s, and pure text at 320 kbit/s, or any value in between.
In such situations a small amount of memory in the modem, a buffer, is used to hold
the data while it is being compressed and sent across the phone line, but in order to
prevent overflow of the buffer, it sometimes becomes necessary to tell the computer
to pause the datastream. This is accomplished through hardware flow control using
extra lines on the modem–computer connection. The computer is then set to supply
the modem at some higher rate, such as 320 kbit/s, and the modem will tell the
computer when to start or stop sending data.
Compression by the ISP
This section does not cite any references or sources. Please help improve this
section by adding citations to reliable sources. Unsourced material may be challenged
and removed. (March 2013)
As telephone-based 56k modems began losing popularity, some Internet service
providers such as Netzero/Juno, Netscape, and others started using pre-compression to
increase the throughput and maintain their customer base. The server-side
compression operates much more efficiently than the on-the-fly compression done by
modems due to the fact these compression techniques are application-specific (JPEG,
text, EXE, etc.). The website text, images, and Flash executables are compacted to
approximately 4%, 12%, and 30%, respectively. The drawback of this approach is a
loss in quality, which causes image content to become pixelated and smeared. ISPs
employing this approach often advertise it as "accelerated dial-up."
These accelerated downloads are now integrated into the Opera and Amazon Silk web
browsers, using their own server-side text and image compression.
Softmodem
A PCI Winmodem/softmodem (on the left) next to a traditional ISA modem (on
the right).
A Winmodem or softmodem is a stripped-down modem that replaces tasks
traditionally handled in hardware with software. In this case the modem is a simple
interface designed to act as a digital-to-analog and an analog-to-digital converter.
Softmodems are cheaper than traditional modems because they have fewer hardware
components. However, the software generating and interpreting the modem tones to
be sent to the softmodem uses many system resources. For online gaming, this can be
a real concern. Another problem is the lack of cross-platform compatibility, meaning
that non-Windows operating systems (such as Linux) often do not have an equivalent
driver to operate the modem.
List of dialup speeds
These values are maximum values, and actual values may be slower under certain
conditions (for example, noisy phone lines). [7] For a complete list see the companion
article list of device bandwidths. A baud is one symbol per second; each symbol may
encode one or more data bits.
FSK
Bitrate
[kbit/s]
0.1
Year
Released
1958
FSK
0.3
1962
FSK
1.2
QPSK
1.2
1980[8][9]
QAM
2.4
1984 [8]
PSK
2.4
PSK
4.8
[10]
QAM
9.6
1984 [8]
Connection
Modulation
110 baudBell 101 modem
300 baud (Bell 103 or
V.21)
1200 modem (1200 baud)
(Bell 202)
1200 Modem (600 baud)
(Bell 212A or V.22)
(V.22bis)
(V.26bis)
(V.27ter)
(V.32)
14.4k Modem (2400
baud) (V.32bis)
28.8k Modem (3200
baud) (V.34)
33.6k Modem (3429
baud) (V.34)
56k Modem (8000/3429
baud) (V.90)
56k Modem (8000/8000
baud) (V.92)
Bonding modem (two 56k
modems) (V.92)[12]
Hardware
compression
(variable) (V.90/V.42bis)
Hardware
compression
(variable) (V.92/V.44)
Server-side
web
compression
(variable)
(Netscape ISP)
trellis
14.4
1991 [8]
trellis
28.8
1994 [8]
trellis
33.6
1996 [11]
digital
56.0/33.6
1998 [8]
digital
56.0/48.0
2000 [8]
112.0/96.0
56.0–220.0
56.0–320.0
100.0–
1,000.0
Radio modems
Direct broadcast satellite, WiFi, and mobile phones all use modems to communicate,
as do most other wireless services today. Modern telecommunications and data
networks also make extensive use of radio modems where long distance data links are
required. Such systems are an important part of the PSTN, and are also in common
use for high-speed computer network links to outlying areas where fibre is not
economical.
Even where a cable is installed, it is often possible to get better performance or make
other parts of the system simpler by using radio frequencies and modulation
techniques through a cable. Coaxial cable has a very large bandwidth, however signal
attenuation becomes a major problem at high data rates if a baseband digital signal is
used. By using a modem, a much larger amount of digital data can be transmitted
through a single wire. Digital cable television and cable Internet services use radio
frequency modems to provide the increasing bandwidth needs of modern households.
Using a modem also allows for frequency-division multiple access to be used, making
full-duplex digital communication with many users possible using a single wire.
Wireless modems come in a variety of types, bandwidths, and speeds. Wireless
modems are often referred to as transparent or smart. They transmit information that
is modulated onto a carrier frequency to allow many simultaneous wireless
communication links to work simultaneously on different frequencies.
Transparent modems operate in a manner similar to their phone line modem cousins.
Typically, they were half duplex, meaning that they could not send and receive data at
the same time. Typically transparent modems are polled in a round robin manner to
collect small amounts of data from scattered locations that do not have easy access to
wired infrastructure. Transparent modems are most commonly used by utility
companies for data collection.
Smart modems come with media access controllers inside, which prevents random
data from colliding and resends data that is not correctly received. Smart modems
typically require more bandwidth than transparent modems, and typically achieve
higher data rates. The IEEE 802.11 standard defines a short range modulation scheme
that is used on a large scale throughout the world.
WiFi and WiMax
Wireless data modems are used in the WiFi and WiMax standards, operating at
microwave frequencies.
Mobile broadband modems
HuaweiCDMA2000Evolution-Data Optimized USB wireless modem
Modems which use a mobile telephone system (GPRS, UMTS, HSPA, EVDO, WiMax, etc.), are known as mobile broadband modems (sometimes also called wireless
modems). Wireless modems can be embedded inside a laptop or appliance, or be
external to it. External wireless modems are connect cards, USB modems for mobile
broadband and cellular routers. A connect card is a PC card or Express Card which
slides into a PCMCIA/PC card/Express Card slot on a computer. USB wireless
modems use a USB port on the laptop instead of a PC card or Express Card slot. A
USB modem used for mobile broadband Internet is also sometimes referred to as a
dongle.[13] A cellular router may have an external data card (Air Card) that slides into
it. Most cellular routers do allow such data cards or USB modems. Cellular routers
may not be modems by definition, but they contain modems or allow modems to be
slid into them. The difference between a cellular router and a wireless modem is that a
cellular router normally allows multiple people to connect to it (since it can route data
or support multipoint to multipoint connections), while a modem is designed for one
connection.
Most of GSM wireless modems come with an integrated SIM cardholder (i.e., Huawei
E220, Sierra 881, etc.) and some models are also provided with a microSD memory
slot and/or jack for additional external antenna such as Huawei E1762 and Sierra
Wireless Compass 885.[14][15] The CDMA (EVDO) versions do not use R-UIM cards,
but use Electronic Serial Number (ESN) instead.
The cost of using a wireless modem varies from country to country. Some carriers
implement flat rate plans for unlimited data transfers. Some have caps (or maximum
limits) on the amount of data that can be transferred per month. Other countries have
plans that charge a fixed rate per data transferred—per megabyte or even kilobyte of
data downloaded; this tends to add up quickly in today's content-filled world, which is
why many people[who?] are pushing for flat data rates.
The faster data rates of the newest wireless modem technologies (UMTS, HSPA,
EVDO, WiMax) are also considered to be broadband wireless modems and compete
with other broadband modems below.
Until the end of April 2011, worldwide shipments of USB modems surpassed
embedded 3G and 4G modules by 3:1 because USB modems can be easily discarded,
but embedded modems could start to gain popularity as tablet sales grow and as the
incremental cost of the modems shrinks, so by 2016 the ratio may change to 1:1. [16]
Like mobile phones, mobile broadband modems can be SIM locked to a particular
network provider. Unlocking a modem is achieved the same way as unlocking a
phone, by using an 'unlock code'[17]
Broadband
DSL modem
ADSL (asymmetric digital subscriber line) modems, a more recent development, are
not limited to the telephone's voice band audio frequencies. Some ADSL modems use
coded orthogonal frequency division modulation (DMT, for Discrete Multi Tone; also
called COFDM, for digital TV in much of the world).
DSL modems utilize a property that standard twisted-pair telephone cable can be used
for short distances to carry much higher frequency signals than what the cable is
actually rated to handle. This is also why DSL modems have a distance limitation.
Standard voice and slower 56 kilobit modem communications are possible over many
kilometers of cable, but the higher frequencies used by DSL are attenuated and DSL's
maximum performance gradually declines as the cable length increases.
Cable modems use a range of frequencies originally intended to carry RF television
channels, and can coexist on the same single cable alongside standard RF channel
signals. Multiple cable modems attached to a single cable can use the same frequency
band, using a low-level media access protocol to allow them to work together within
the same channel. Typically, uplink and downlink signals are kept separate using
frequency division multiple access.
For a single-cable distribution system, the return signals from customers require
special bidirectional amplifiers or reverse path amplifiers that can send specific
customer frequency bands upstream to the cable plant amongst the other downstream
frequency bands.
New types of broadband modems are beginning to appear, such as double way
satellite and power line modems.
Broadband modems should still be classified as modems, since they use complex
waveforms to carry digital data. They are more advanced devices than traditional dialup modems as they are either capable of modulating/demodulating hundreds of
channels simultaneously and/or are capable of using much wider channels than dialup modems.
Many broadband modems include the functions of a router, such as Ethernet and
WiFi, and other features such as DHCP, NAT and firewalls.
When broadband technology was introduced, networking and routers were unfamiliar
to consumers. However, many people knew what a modem was because Internet
access was still commonly done through dial-up. Due to this familiarity, companies
started selling broadband modems using the familiar term "modem", rather than
vaguer ones such as "adapter," "transceiver," or "bridge."
Bridged mode
DSL modems without internal routing use what is known as bridge mode to connect
to an upstream router device. DSL modems with built-in routing can also sometimes
be set to bridged mode to disable the internal built-in router in order to use an
upstream router instead, such as for Multiple-WAN load balancing, Multilink PPP, or
to replace the built-in router with an external device with more capabilities.
In bridged mode, although Ethernet cabling is used to connect the modem to the
upstream router, Internet Protocol is not used for communication between the devices.
Instead the Ethernet cable is treated as a high speed serial Asynchronous Transfer
Mode data connection according to RFC 1483. Multiple bridged DSL modems can all
have the same configuration IP address without conflict or error, since the address is
not used.
The upstream router is expected to use Point-to-point protocol over Ethernet (PPPoE)
or Multilink PPP in order to establish a connection on the DSL phone line.
The static IP address assigned to the bridged DSL modem is only used when the
modem is plugged into a client computer for directly configuring the modem,
typically through a web interface.
Home networking
Although the name modem is seldom used in this case, modems are also used for
high-speed home networking applications, especially those using existing home
wiring. One example is the G.hn standard, developed by ITU-T, which provides a
high-speed (up to 1 Gbit/s) Local area network using existing home wiring (power
lines, phone lines and coaxial cables). G.hn devices use orthogonal frequency-division
multiplexing (OFDM) to modulate a digital signal for transmission over the wire.
The phrase "null modem" was used to describe attaching a specially wired cable
between the serial ports of two personal computers. Basically, the transmit output of
one computer was wired to the receive input of the other; this was true for both
computers. The same software used with modems (such as ProComm or Mincom)
could be used with the null modem connection.
Deep-space communications
Many modern modems have their origin in space telecommunications from the 1960s.
Differences between deep space telecom modems and landline modems include:
Digital modulation formats that have high Doppler immunity are typically used.
Waveform complexity tends to be low—typically binary phase shift keying.
Error correction varies mission to mission, but is typically much stronger than most
landline modems.
Voice modem
Voice modems are regular modems that are capable of recording or playing audio
over the telephone line. They are used for telephony applications. See Voice modem
command set for more details on voice modems. This type of modem can be used as
an FXO card for Private branch exchange systems (compare V.92).
6. Explain the ISO/OSI reference model?
The Open Systems Interconnection (OSI) model (ISO/IEC 7498-1) is a conceptual
model that characterizes and standardizes the internal functions of a communication
system by partitioning it into abstraction layers. The model is a product of the Open
Systems Interconnection project at the International Organization for Standardization
(ISO).
The model group’s similar communication functions into one of seven logical layers.
A layer serves the layer above it and is served by the layer below it. For example, a
layer that provides error-free communications across a network provides the path
needed by applications above it, while it calls the next lower layer to send and receive
packets that make up the contents of that path. Two instances at one layer are
connected by a horizontal connection on that layer.
OSI Model
Communication in the OSI-Model (example with layers 3 to 5)
History
Work on a layered model of network architecture was started and the International
Organization for Standardization (ISO) began to develop its OSI framework
architecture. OSI had two major components: an abstract model of networking, called
the Basic Reference Model or seven-layer model, and a set of specific protocols.
The concept of a seven-layer model was provided by the work of Charles Bachman,
Honeywell Information Services. Various aspects of OSI design evolved from
experiences with the ARPANET, the fledgling Internet, NPLNET, EIN, CYCLADES
network and the work in IFIP WG6.1. The new design was documented in ISO 7498
and its various addenda. In this model, a networking system was divided into layers.
Within each layer, one or more entities implement its functionality. Each entity
interacted directly only with the layer immediately beneath it, and provided facilities
for use by the layer above it.
Protocols enabled an entity in one host to interact with a corresponding entity at the
same layer in another host. Service definitions abstractly described the functionality
provided to an (N)-layer by an (N-1) layer, where N was one of the seven layers of
protocols operating in the local host.
The OSI standards documents are available from the ITU-T as the X.200-series of
recommendations.[1] Some of the protocol specifications were also available as part of
the ITU-T X series. The equivalent ISO and ISO/IEC standards for the OSI model
were available from ISO, but only some of them without fees. [2]
Description of OSI layers
According to recommendation X.200, there are seven layers, labeled 1 to 7, with layer
1 at the bottom. Each layer is generically known as an N layer. An "N+1 entity" (at
layer N+1) requests services from an "N entity" (at layer N).
Data
unit
Layer
Function
7. Application
Network process to application
6. Presentation
Data representation, encryption and
decryption, convert machine
dependent data to machine
independent data
5. Session
Interhost communication, managing
sessions between applications
4. Transport
Reliable delivery of packets
between points on a network.
Packet/
Datagra
m
3. Network
Addressing, routing and (not
necessarily reliable) delivery of
datagram’s between points on a
network.
Bit
2. Data link
A reliable direct point-to-point data
connection.
Bit
1. Physical
A (not necessarily) reliable direct
point-to-point data connection.
Data
Host
layers
Segmen
ts
Media
layers
At each level, two entities (N-entity peers) interact by means of the N protocol by
transmitting protocol data units (PDU).
A service data unit (SDU) is a specific unit of data that has been passed down from an
OSI layer to a lower layer, and which the lower layer has not yet encapsulated into a
protocol data unit (PDU). An SDU is a set of data that is sent by a user of the services
of a given layer, and is transmitted semantically unchanged to a peer service user.
The PDU at a layer N is the SDU of layer N-1. In effect the SDU is the 'payload' of a
given PDU. That is, the process of changing an SDU to a PDU, consists of an
encapsulation process, performed by the lower layer. All the data contained in the
SDU becomes encapsulated within the PDU. The layer N-1 adds headers or footers, or
both, to the SDU, transforming it into a PDU of layer N-1. The added headers or
footers are part of the process used to make it possible to get data from a source to a
destination.
Some orthogonal aspects, such as management and security, involve every layer.
Security services are not related to a specific layer: they can be related by a number of
layers, as defined by ITU-T X.800 Recommendation.[3]
These services are aimed to improve the CIA triad (confidentiality, integrity, and
availability) of transmitted data. In practice, the availability of communication service
is determined by the interaction between network design and network management
protocols. Appropriate choices for both of these are needed to protect against denial of
service.[citation needed]
Layer 1: physical layer
The physical layer has the following major functions:
It defines the electrical and physical specifications of the data connection. It defines
the relationship between a device and a physical transmission medium (e.g. a copper
or fiber optical cable). This includes the layout of pins, voltages, line impedance,
cable specifications, signal timing, hubs, repeaters, network adapters, host bus
adapters (HBA used in storage area networks) and more.
It defines the protocol to establish and terminate a connection between two directly
connected nodes over a communications medium.
It may define the protocol for flow control.
It defines a protocol for the provision of a (not necessarily reliable) connection
between two directly connected nodes, and the Modulation or conversion between the
representation of digital data in user equipment and the corresponding signals
transmitted over the physical communications channel. This channel can involve
physical cabling (such as copper and optical fiber) or a wireless radio link.
The physical layer of Parallel SCSI operates in this layer, as do the physical layers of
Ethernet and other local-area networks, such as token ring, FDDI, ITU-TG.hn, and
IEEE 802.11, as well as personal area networks such as Bluetooth and IEEE 802.15.4.
Layer 2: data link layer
The data link layer provides a reliable link between two directly connected nodes, by
detecting and possibly correcting errors that may occur in the physical layer.
An example, in the TCP/IP protocol stack, of a data link layer is the Point-to-Point
Protocol (PPP).
The ITU-TG.hn standard, which provides high-speed local area networking over
existing wires (power lines, phone lines and coaxial cables), includes a complete data
link layer which provides both error correction and flow control by means of a
selective repeat Sliding Window Protocol.
Layer 3: network layer
The network layer provides the functional and procedural means of transferring
variable length data sequences (called data grams) from one node to another
connected to the same network. A network is a medium to which many nodes can be
connected, on which every node has an address and which permits nodes connected to
it to transfer messages to other nodes connected to it by merely providing the content
of a message and the address of the destination node and letting the network find the
way to deliver ("route") the message to the destination node. In addition to message
routing, the network may (or may not) implement message delivery by splitting the
message into several fragments, delivering each fragment by a separate route and
reassembling the fragments, report delivery errors, etc.
Datagram delivery at the network layer is not guaranteed to be reliable.
Examples of network layers in the usual Internet protocol stack are the Ethernet
(MAC) layer, the IP layer, and the UDP layer. The IP layer is an example of a
network layer often implemented on top of another network layer (MAC) rather than a
data link layer, as is UDP, which is a network layer implemented over IP, another
network layer. In the original design of the Internet, the network layer on top of which
the Internet was built was often called the "network layer" to distinguish it from the
Internet / IP "inter-network" network layer, which was built on top of it. Both the
network and the internetwork layers fit into the network layer in the OSI model.
A number of layer-management protocols, a function defined in the Management
Annex, ISO 7498/4, belong to the network layer. These include routing protocols,
multicast group management, network-layer information and error, and network-layer
address assignment. It is the function of the payload that makes these belong to the
network layer, not the protocol that carries them.
Layer 4: transport layer
The transport layer provides the reliable sending of data packets between nodes (with
addresses) located on a network, providing reliable data transfer services to the upper
layers.
An example of a transport layer protocol in the standard Internet protocol stack is
TCP, usually built on top of the IP protocol.
The transport layer controls the reliability of a given link through flow control,
segmentation/desegmentation, and error control. Some protocols are state- and
connection-oriented. This means that the transport layer can keep track of the
segments and retransmit those that fail. The transport layer also provides the
acknowledgement of the successful data transmission and sends the next data if no
errors occurred.
OSI defines five classes of connection-mode transport protocols ranging from class 0
(which is also known as TP0 and provides the least features) to class 4 (TP4, designed
for less reliable networks, similar to the Internet). Class 0 contains no error recovery,
and was designed for use on network layers that provide error-free connections. Class
4 is closest to TCP, although TCP contains functions, such as the graceful close,
which OSI assigns to the session layer. Also, all OSI TP connection-mode protocol
classes provide expedited data and preservation of record boundaries. Detailed
characteristics of TP0-4 classes are shown in the following table:[4]
Feature Name
Connection oriented network
Connectionless network
Concatenation and separation
Segmentation and reassembly
Error Recovery
Reinitiate connection (if an
excessive number of PDUs are
unacknowledged)
Multiplexing and demultiplexing
over a single virtual circuit
Explicit flow control
Retransmission on timeout
Reliable Transport Service
TP0
Yes
No
No
Yes
No
TP1
Yes
No
Yes
Yes
Yes
TP2
Yes
No
Yes
Yes
Yes
TP3
Yes
No
Yes
Yes
Yes
TP4
Yes
Yes
Yes
Yes
Yes
No
Yes
No
Yes
No
No
No
Yes
Yes
Yes
No
No
No
No
No
Yes
Yes
No
No
Yes
No
Yes
Yes
Yes
Yes
An easy way to visualize the transport layer is to compare it with a Post Office, which
deals with the dispatch and classification of mail and parcels sent. Do remember,
however, that a post office manages the outer envelope of mail. Higher layers may
have the equivalent of double envelopes, such as cryptographic presentation services
that can be read by the addressee only. Roughly speaking, tunneling protocols operate
at the transport layer, such as carrying non-IP protocols such as IBM's SNA or
Novell's IPX over an IP network, or end-to-end encryption with IPSec. While Generic
Routing Encapsulation (GRE) might seem to be a network-layer protocol, if the
encapsulation of the payload takes place only at endpoint, GRE becomes closer to a
transport protocol that uses IP headers but contains complete frames or packets to
deliver to an endpoint. L2TP carries PPP frames inside transport packet.
Although not developed under the OSI Reference Model and not strictly conforming
to the OSI definition of the transport layer, the Transmission Control Protocol (TCP)
and the User Datagram Protocol (UDP) of the Internet Protocol Suite are commonly
categorized as layer-4 protocols within OSI.
Layer 5: session layer
The session layer controls the dialogues (connections) between computers. It
establishes, manages and terminates the connections between the local and remote
application. It provides for full-duplex, half-duplex, or simplex operation, and
establishes check pointing, adjournment, termination, and restart procedures. The OSI
model made this layer responsible for graceful close of sessions, which is a property
of the Transmission Control Protocol, and also for session check pointing and
recovery, which is not usually used in the Internet Protocol Suite. The session layer is
commonly implemented explicitly in application environments that use remote
procedure calls.
Layer 6: presentation layer
The presentation layer establishes context between application-layer entities, in which
the higher-layer entities may use different syntax and semantics if the presentation
service provides a mapping between them. If a mapping is available, presentation
service data units are encapsulated into session protocol data units, and passed down
the stack.
This layer provides independence from data representation (e.g., encryption) by
translating between application and network formats. The presentation layer
transforms data into the form that the application accepts. This layer formats and
encrypts data to be sent across a network. It is sometimes called the syntax layer. [5]
The original presentation structure used the Basic Encoding Rules of Abstract Syntax
Notation One (ASN.1), with capabilities such as converting an EBCDIC-coded text
file to an ASCII-coded file, or serialization of objects and other data structures from
and to XML.
Layer 7: application layer
The application layer is the OSI layer closest to the end user, which means that both
the OSI application layer and the user interact directly with the software application.
This layer interacts with software applications that implement a communicating
component. Such application programs fall outside the scope of the OSI model.
Application-layer functions typically include identifying communication partners,
determining resource availability, and synchronizing communication. When
identifying communication partners, the application layer determines the identity and
availability of communication partners for an application with data to transmit. When
determining resource availability, the application layer must decide whether sufficient
network or the requested communication exists. In synchronizing communication, all
communication between applications requires cooperation that is managed by the
application layer. Some examples of application-layer implementations also include:
o
o
o
o
o
o
o
On OSI stack:
FTAM File Transfer and Access Management Protocol
X.400 Mail
Common Management Information Protocol (CMIP)
On TCP/IP stack:
Hypertext Transfer Protocol (HTTP),
File Transfer Protocol (FTP),
Simple Mail Transfer Protocol (SMTP)
Simple Network Management Protocol (SNMP).
Non-Linear Layering
It is not always the case that layer (N+1) is implemented on top of layer N of the OSI
stack.
For example, PPPoE implements PPP, a link layer protocol, on top of Ethernet, a
network layer protocol. Similarly, Modem used on top of a data link provided by a
v.42bis capable modem is providing a data link layer protocol on top of another data
link layer. Modem used within a telnet or ssh connection operating over the Internet,
provides a data link layer protocol on top of an application layer. The IP and ICMP
protocols implemented over Ethernet implement a network layer protocol over
another network layer, as does UDP implemented over IP.
Cross-layer functions
There are some functions or services that are not tied to a given layer, but they can
affect more than one layer. Examples include the following:
Security service (telecommunication) as defined by ITU-T X.800 Recommendation.
management functions, i.e. functions that permit to configure, instantiate, monitor,
terminate the communications of two or more entities: there is a specific application
layer protocol, common management information protocol (CMIP) and its
corresponding service, common management information service (CMIS), they need
to interact with every layer in order to deal with their instances.
Multiprotocol Label Switching (MPLS) operates at an OSI-model layer that is
generally considered to lie between traditional definitions of layer 2 (data link layer)
and layer 3 (network layer), and thus is often referred to as a "layer-2.5" protocol. It
was designed to provide a unified data-carrying service for both circuit-based clients
and packet-switching clients which provide a datagram-based service model. It can be
used to carry many different kinds of traffic, including IP packets, as well as native
ATM, SONET, and Ethernet frames.
ARP is used to translate IPv4 addresses (OSI layer 3) into Ethernet MAC addresses
(OSI layer 2).
Interfaces
Neither the OSI Reference Model nor OSI protocols specify any programming
interfaces, other than as deliberately abstract service specifications. Protocol
specifications precisely define the interfaces between different computers, but the
software interfaces inside computers, known as network sockets are implementationspecific.
For example Microsoft Windows' Winsock, and Unix's Berkeley sockets and System
V transport Layer Interface, are interfaces between applications (layer 5 and above)
and the transport (layer 4). NDIS and ODI are interfaces between the media (layer 2)
and the network protocol (layer 3).
V UNIT 2 MARKS
1. Define satellite.
Satellite is a celestial body that orbits around a planet. In aerospace terms,
A satellite is a space vehicle launched by humans and orbits earth or another
Celestial body.
2. State Kepler‘s first law.
Kepler‘s first law states that a satellite will orbit a primary body following
an elliptical path.
3. State Kepler‘s second law.
Kepler‘s second law states that for equal time intervals of time a satellite
will sweep out equal areas in the orbital plane, focused at the barycentre.
4. State Kepler‘s third law.
The third law states that the square of the periodic time of orbit is
proportional to the cube of the mean distance between the primary and the
satellite.
5. Define orbital satellite.
Orbital satellites are also called as nonsynchronous
Satellite. Nonsynchronous satellites rotate around earth in an elliptical or
Circular pattern. In a circular orbit, the speed or rotation is constant however in
Elliptical orbits the speed depends on the height the satellite is above the earth.
6. Define Azimuth angle.
Azimuth is the horizontal angular distance from a reference direction,
either the southern or northern most point of the horizon.
7. Define retrograde orbit.
If the satellite is orbiting in the opposite direction as the earth‘s rotation or
in the same direction with an angular velocity less than that of earth, the orbit is
Called a retrograde orbit.
8. Define Geo synchronous satellite.
Geo synchronous or geo stationary satellites are those that orbit in a
circular pattern with an angular velocity equal to that of Erath. Geosynchronous
satellites have an orbital time of approximately 24 hours, the same as earth; thus
geosynchronous satellites appear to be stationary as they remain in a fixed
position in respect to a given point on earth.
9. Define apogee and perigee.
The point in an orbit which is located farthest from the earth is called
apogee.
The point in an orbit which is located closest to earth is called perigee.
10. Define angle of inclination.
The angle of inclination is the angle between the earth‘s equatorial plane
and the orbital plane of a satellite measured counter clockwise at the point in the
orbit where it crosses the equatorial plane travelling from south to north.
V UNIT 16 MARKS
1.
Give a brief discussion about Cellular network?
Top of a cellular radio tower
A cellular network or mobile network is a radio network distributed over land areas called
cells, each served by at least one fixed-location transceiver, known as a cell site or base
station. In a cellular network, each cell uses a different set of frequencies from neighboring
cells, to avoid interference and provide guaranteed bandwidth within each cell.
When joined together these cells provide radio coverage over a wide geographic area. This
enables a large number of portable transceivers (e.g., mobile phones, pagers, etc.) to
communicate with each other and with fixed transceivers and telephones anywhere in the
network, via base stations, even if some of the transceivers are moving through more than
one cell during transmission.
Cellular networks offer a number of advantages over alternative solutions:
flexible enough to use the features and functions of almost all public and private
networks
increased capacity
reduced power use
larger coverage area
reduced interference from other signals
An example of a simple non-telephone cellular system is an old taxi drivers' radio system, in
which a taxi company has several transmitters based around a city that can communicate
directly with each other.
Concept
Example of frequency reuses factor or pattern 1/4
In a cellular radio system, a land area to be supplied with radio service is divided into regular
shaped cells, which can be hexagonal, square, circular or some other regular shapes, although
hexagonal cells are conventional. Each of these cells is assigned multiple frequencies (f1 – f6)
which have corresponding radio base stations. The group of frequencies can be reused in
other cells, provided that the same frequencies are not reused in adjacent neighboring cells as
that would cause co-channel interference.
The increased capacity in a cellular network, compared with a network with a single
transmitter, comes from the fact that the same radio frequency can be reused in a different
area for a completely different transmission. If there is a single plain transmitter, only one
transmission can be used on any given frequency. Unfortunately, there is inevitably some
level of interference from the signal from the other cells which use the same frequency. This
means that, in a standard FDMA system, there must be at least a one cell gap between cells
which reuse the same frequency.
In the simple case of the taxi company, each radio had a manually operated channel selector
knob to tune to different frequencies. As the drivers moved around, they would change from
channel to channel. The drivers knew which frequency covered approximately what area.
When they did not receive a signal from the transmitter, they would try other channels until
they found one that worked. The taxi drivers would only speak one at a time, when invited by
the base station operator (this is, in a sense, time division multiple access (TDMA)).
Cell signal encoding
To distinguish signals from several different transmitters, frequency division multiple access
(FDMA) and code division multiple access (CDMA) were developed.
With FDMA, the transmitting and receiving frequencies used in each cell are different from
the frequencies used in each neighboring cell. In a simple taxi system, the taxi driver
manually tuned to a frequency of a chosen cell to obtain a strong signal and to avoid
interference from signals from other cells.
The principle of CDMA is more complex, but achieves the same result; the distributed
transceivers can select one cell and listen to it.
Other available methods of multiplexing such as polarization division multiple access
(PDMA) and time division multiple access (TDMA) cannot be used to separate signals from
one cell to the next since the effects of both vary with position and this would make signal
separation practically impossible. Time division multiple access, however, is used in
combination with either FDMA or CDMA in a number of systems to give multiple channels
within the coverage area of a single cell.
Frequency reuse
The key characteristic of a cellular network is the ability to re-use frequencies to increase
both coverage and capacity. As described above, adjacent cells must use different
frequencies, however there is no problem with two cells sufficiently far apart operating on the
same frequency. The elements that determine frequency reuse are the reuse distance and the
reuse factor.
The reuse distance, D is calculated as
where R is the cell radius and N is the number of cells per cluster. Cells may vary in radius in
the ranges (1 km to 30 km). The boundaries of the cells can also overlap between adjacent
cells and large cells can be divided into smaller cells.
The frequency reuse factor is the rate at which the same frequency can be used in the
network. It is 1/K (or K according to some books) where K is the number of cells which
cannot use the same frequencies for transmission. Common values for the frequency reuse
factor are 1/3, 1/4, 1/7, 1/9 and 1/12 (or 3, 4, 7, 9 and 12 depending on notation).
In case of N sector antennas on the same base station site, each with different direction, the
base station site can serve N different sectors. N is typically 3. A reuse pattern of N/K
denotes a further division in frequency among N sector antennas per site. Some current and
historical reuse patterns are 3/7 (North American AMPS), 6/4 (Motorola NAMPS), and 3/4
(GSM).
If the total available bandwidth is B, each cell can only use a number of frequency channels
corresponding to a bandwidth of B/K, and each sector can use a bandwidth of B/NK.
Code division multiple access-based systems use a wider frequency band to achieve the same
rate of transmission as FDMA, but this is compensated for by the ability to use a frequency
reuse factor of 1, for example using a reuse pattern of 1/1. In other words, adjacent base
station sites use the same frequencies, and the different base stations and users are separated
by codes rather than frequencies. While N is shown as 1 in this example that does not mean
the CDMA cell has only one sector, but rather that the entire cell bandwidth is also available
to each sector individually.
Depending on the size of the city, a taxi system may not have any frequency-reuse in its own
city, but certainly in other nearby cities, the same frequency can be used. In a large city, on
the other hand, frequency-reuse could certainly be in use.
Recently also orthogonal frequency-division multiple access based systems such as LTE are
being deployed with a frequency reuse of 1. Since such systems do not spread the signal
across the frequency band, inter-cell radio resource management is important to coordinate
resource allocation between different cell sites and to limit the inter-cell interference. There
are various means of Inter-cell Interference Coordination (ICIC) already defined in the
standard. Coordinated scheduling, multi-site MIMO or multi-site beam forming are other
examples for inter-cell radio resource management that might be standardized in the future.
Directional antennas
Cellular telephone frequency reuse pattern.
Although the original 2-way-radio cell towers were at the centers of the cells and were omnidirectional, a cellular map can be redrawn with the cellular telephone towers located at the
corners of the hexagons where three cells converge. Each tower has three sets of directional
antennas aimed in three different directions with 120 degrees for each cell (totaling 360
degrees) and receiving/transmitting into three different cells at different frequencies. This
provides a minimum of three channels (from three towers) for each cell. The numbers in the
illustration are channel numbers, which repeat every 3 cells. Large cells can be subdivided
into smaller cells for high volume areas.
Broadcast messages and paging
Practically every cellular system has some kind of broadcast mechanism. This can be used
directly for distributing information to multiple mobiles, commonly, for example in mobile
telephony systems, the most important use of broadcast information are to set up channels for
one to one communication between the mobile transceiver and the base station. This is called
paging. The three different paging procedures generally adopted are sequential, parallel and
selective paging.
The details of the process of paging vary somewhat from network to network, but normally
we know a limited number of cells where the phone is located (this group of cells is called a
Location Area in the GSM or UMTS system, or Routing Area if a data packet session is
involved; in LTE, cells are grouped into Tracking Areas). Paging takes place by sending the
broadcast message to all of those cells. Paging messages can be used for information transfer.
This happens in pagers, in CDMA systems for sending SMS messages, and in the UMTS
system where it allows for low downlink latency in packet-based connections.
Movement from cell to cell and handover
In a primitive taxi system, when the taxi moved away from a first tower and closer to a
second tower, the taxi driver manually switched from one frequency to another as needed. If
a communication was interrupted due to a loss of a signal, the taxi driver asked the base
station operator to repeat the message on a different frequency.
In a cellular system, as the distributed mobile transceivers move from cell to cell during an
ongoing continuous communication, switching from one cell frequency to a different cell
frequency is done electronically without interruption and without a base station operator or
manual switching. This is called the handover or handoff. Typically, a new channel is
automatically selected for the mobile unit on the new base station which will serve it. The
mobile unit then automatically switches from the current channel to the new channel and
communication continues.
The exact details of the mobile system‘s move from one base station to the other vary
considerably from system to system (see the example below for how a mobile phone network
manages handover).
Mobile phone network
GSM network architecture
The most common example of a cellular network is a mobile phone (cell phone) network. A
mobile phone is a portable telephone which receives or makes calls through a cell site (base
station), or transmitting tower. Radio waves are used to transfer signals to and from the cell
phone.
Modern mobile phone networks use cells because radio frequencies are a limited, shared
resource. Cell-sites and handsets change frequency under computer control and use low
power transmitters so that a limited number of radio frequencies can be simultaneously used
by many callers with less interference.
A cellular network is used by the mobile phone operator to achieve both coverage and
capacity for their subscribers. Large geographic areas are split into smaller cells to avoid lineof-sight signal loss and to support a large number of active phones in that area. All of the cell
sites are connected to telephone exchanges (or switches), which in turn connect to the public
telephone network.
In cities, each cell site may have a range of up to approximately ½ mile, while in rural areas;
the range could be as much as 5 miles. It is possible that in clear open areas, a user may
receive signals from a cell site 25 miles away.
Since almost all mobile phones use cellular technology, including GSM, CDMA, and AMPS
(analog), the term "cell phone" is in some regions, notably the US, used interchangeably with
"mobile phone". However, satellite phones are mobile phones that do not communicate
directly with a ground-based cellular tower, but may do so indirectly by way of a satellite.
There are a number of different digital cellular technologies, including: Global System for
Mobile Communications (GSM), General Packet Radio Service (GPRS), cdma One,
CDMA2000, Evolution-Data Optimized (EV-DO), Enhanced Data Rates for GSM Evolution
(EDGE), Universal Mobile Telecommunications System (UMTS), Digital Enhanced Cordless
Telecommunications (DECT), Digital AMPS (IS-136/TDMA), and Integrated Digital
Enhanced Network (iDEN).
Structure of the mobile phone cellular network
A simple view of the cellular mobile-radio network consists of the following:
A network of radio base stations forming the base station subsystem.
The core circuit switched network for handling voice calls and text
A packet switched network for handling mobile data
The public switched telephone network to connect subscribers to the wider telephony
network
This network is the foundation of the GSM system network. There are many functions that
are performed by this network in order to make sure customers get the desired service
including mobility management, registration, call set up, and handover.
Any phone connects to the network via an RBS (Radio Base Station) at a corner of the
corresponding cell which in turn connects to the Mobile switching center (MSC). The MSC
provides a connection to the public switched telephone network (PSTN). The link from a
phone to the RBS is called an uplink while the other way is termed downlink.
Radio channels effectively use the transmission medium through the use of the following
multiplexing & access schemes: frequency division multiple access (FDMA), time division
multiple access (TDMA), code division multiple access (CDMA), and space division
multiple access (SDMA).
Cellular handover in mobile phone networks
As the phone user moves from one cell area to another cell while a call is in progress, the
mobile station will search for a new channel to attach to in order not to drop the call. Once a
new channel is found, the network will command the mobile unit to switch to the new
channel and at the same time switch the call onto the new channel.
With CDMA, multiple CDMA handsets share a specific radio channel. The signals are
separated by using a pseudo noise code (PN code) specific to each phone. As the user moves
from one cell to another, the handset sets up radio links with multiple cell sites (or sectors of
the same site) simultaneously. This is known as "soft handoff" because, unlike with
traditional cellular technology, there is no one defined point where the phone switches to the
new cell.
In IS-95 inter-frequency handovers and older analog systems such as NMT it will typically be
impossible to test the target channel directly while communicating. In this case other
techniques have to be used such as pilot beacons in IS-95. This means that there is almost
always a brief break in the communication while searching for the new channel followed by
the risk of an unexpected return to the old channel.
If there is no ongoing communication or the communication can be interrupted, it is possible
for the mobile unit to spontaneously move from one cell to another and then notify the base
station with the strongest signal.
Cellular frequency choice in mobile phone networks
The effect of frequency on cell coverage means that different frequencies serve better for
different uses. Low frequencies, such as 450 MHz NMT, serve very well for countryside
coverage. GSM 900 (900 MHz) is a suitable solution for light urban coverage. GSM 1800
(1.8 GHz) starts to be limited by structural walls. UMTS, at 2.1 GHz is quite similar in
coverage to GSM 1800.
Higher frequencies are a disadvantage when it comes to coverage, but it is a decided
advantage when it comes to capacity. Pico cells, covering e.g. one floor of a building, become
possible, and the same frequency can be used for cells which are practically neighbors.
Cell service area may also vary due to interference from transmitting systems, both within
and around that cell. This is true especially in CDMA based systems. The receiver requires a
certain signal-to-noise ratio, and the transmitter should not send with too high transmission
power in view to not cause interference with other transmitters. As the receiver moves away
from the transmitter, the power received decreases, so the power control algorithm of the
transmitter increases the power it transmits to restore the level of received power. As the
interference (noise) rises above the received power from the transmitter, and the power of the
transmitter cannot be increased any more, the signal becomes corrupted and eventually
unusable. In CDMA-based systems, the effect of interference from other mobile transmitters
in the same cell on coverage area is very marked and has a special name, cell breathing.
One can see examples of cell coverage by studying some of the coverage maps provided by
real operators on their web sites or by looking at independently crowd sourced maps such as
Open Signal. In certain cases they may mark the site of the transmitter, in others it can be
calculated by working out the point of strongest coverage.
Coverage comparison of different frequencies
The following table shows the dependency of the coverage area of one cell on the frequency
of a CDMA2000 network:
Frequency (MHz) Cell radius (km) Cell area (km2) Relative Cell Count
2.
450
48.9
7521
1
950
26.9
2269
3.3
1800
14.0
618
12.2
2100
12.0
449
16.2
Give a detailed discussion about Fibre Optic Communication?
Fibre-optic communication
An optical fiber junction box. The yellow cables are single mode fibers; the orange and blue
cables are multi-mode fibers: 62.5/125 µm OM1 and 50/125 µm OM3 fibers respectively.
Fiber-optic communication is a method of transmitting information from one place to another
by sending pulses of light through an optical fiber. The light forms an electromagnetic carrier
wave that is modulated to carry information. First developed in the 1970s, fiber-optic
communication systems have revolutionized the telecommunications industry and have
played a major role in the advent of the Information Age. Because of its advantages over
electrical transmission, optical fibers have largely replaced copper wire communications in
core networks in the developed world.
The process of communicating using fiber-optics involves the following basic steps: Creating
the optical signal involving the use of a transmitter, relaying the signal along the fiber,
ensuring that the signal does not become too distorted or weak, receiving the optical signal,
and converting it into an electrical signal.
Applications
Optical fiber is used by many telecommunications companies to transmit telephone signals,
Internet communication, and cable television signals. Due to much lower attenuation and
interference, optical fiber has large advantages over existing copper wire in long-distance and
high-demand applications. However, infrastructure development within cities was relatively
difficult and time-consuming, and fiber-optic systems were complex and expensive to install
and operate. Due to these difficulties, fiber-optic communication systems have primarily been
installed in long-distance applications, where they can be used to their full transmission
capacity, offsetting the increased cost. Since 2000, the prices for fiber-optic communications
have dropped considerably. The price for rolling out fiber to the home has currently become
more cost-effective than that of rolling out a copper based network. Prices have dropped to
$850 per subscriber[citation needed] in the US and lower in countries like The Netherlands, where
digging costs are low.
Since 1990, when optical-amplification systems became commercially available, the
telecommunications industry has laid a vast network of intercity and transoceanic fiber
communication lines. By 2002, an intercontinental network of 250,000 km of submarine
communications cable with a capacity of 2.56 Tb/s was completed, and although specific
network capacities are privileged information, telecommunications investment reports
indicate that network capacity has increased dramatically since 2004.
Technology
Modern fiber-optic communication systems generally include an optical transmitter to
convert an electrical signal into an optical signal to send into the optical fiber, a cable
containing bundles of multiple optical fibers that is routed through underground conduits and
buildings, multiple kinds of amplifiers, and an optical receiver to recover the signal as an
electrical signal. The information transmitted is typically digital information generated by
computers, telephone systems, and cable television companies.
Transmitters
A GBIC module (shown here with its cover removed), is an optical and electrical transceiver.
The electrical connector is at top right, and the optical connectors are at bottom left
The most commonly used optical transmitters are semiconductor devices such as lightemitting diodes (LEDs) and laser diodes. The difference between LEDs and laser diodes is
that LEDs produce incoherent light, while laser diodes produce coherent light. For use in
optical communications, semiconductor optical transmitters must be designed to be compact,
efficient, and reliable, while operating in an optimal wavelength range, and directly
modulated at high frequencies.
In its simplest form, an LED is a forward-biased p-n junction, emitting light through
spontaneous emission, a phenomenon referred to as electroluminescence. The emitted light is
incoherent with a relatively wide spectral width of 30-60 nm. LED light transmission is also
inefficient, with only about 1%[citation needed] of input power, or about 100 microwatts,
eventually converted into launched power which has been coupled into the optical fiber.
However, due to their relatively simple design, LEDs are very useful for low-cost
applications.
Communications LEDs are most commonly made from Indium gallium arsenide phosphide
(InGaAsP) or gallium arsenide (GaAs). Because InGaAsP LEDs operate at a longer
wavelength than GaAs LEDs (1.3 micrometers vs. 0.81-0.87 micrometers), their output
spectrum, while equivalent in energy is wider in wavelength terms by a factor of about 1.7.
The large spectrum width of LEDs is subject to higher fiber dispersion, considerably limiting
their bit rate-distance product (a common measure of usefulness). LEDs are suitable
primarily for local-area-network applications with bit rates of 10-100 Mbit/s and transmission
distances of a few kilometers. LEDs have also been developed that use several quantum wells
to emit light at different wavelengths over a broad spectrum, and are currently in use for
local-area WDM networks.
Today, LEDs have been largely superseded by VCSEL (Vertical Cavity Surface Emitting
Laser) devices, which offer improved speed, power and spectral properties, at a similar cost.
Common VCSEL devices couple well to multi mode fiber.
A semiconductor laser emits light through stimulated emission rather than spontaneous
emission, which results in high output power (~100 mW) as well as other benefits related to
the nature of coherent light. The output of a laser is relatively directional, allowing high
coupling efficiency (~50 %) into single-mode fiber. The narrow spectral width also allows for
high bit rates since it reduces the effect of chromatic dispersion. Furthermore, semiconductor
lasers can be modulated directly at high frequencies because of short recombination time.
Commonly used classes of semiconductor laser transmitters used in fiber optics include
VCSEL (Vertical Cavity Surface Emitting Laser), Fabry–Pérot and DFB (Distributed Feed
Back).
Laser diodes are often directly modulated, that is the light output is controlled by a current
applied directly to the device. For very high data rates or very long distance links, a laser
source may be operated continuous wave, and the light modulated by an external device such
as an electro-absorption modulator or Mach–Zehnder interferometer. External modulation
increases the achievable link distance by eliminating laser chirp, which broadens the
linewidth of directly modulated lasers, increasing the chromatic dispersion in the fiber.
A transceiver is a device combining a transmitter and a receiver in a single housing (see
picture on right).
Receivers
The main component of an optical receiver is a photo detector, which converts light into
electricity using the photoelectric effect. The primary photo detectors for telecommunications
are made from Indium gallium arsenide The photo detector is typically a semiconductorbased photodiode. Several types of photodiodes include p-n photodiodes, p-i-n photodiodes,
and avalanche photodiodes. Metal-semiconductor-metal (MSM) photo detectors are also used
due to their suitability for circuit integration in regenerators and wavelength-division
multiplexers.
Optical-electrical converters are typically coupled with a trans impedance amplifier and a
limiting amplifier to produce a digital signal in the electrical domain from the incoming
optical signal, which may be attenuated and distorted while passing through the channel.
Further signal processing such as clock recovery from data (CDR) performed by a phaselocked loop may also be applied before the data is passed on.
Fiber cable types
A cable reel trailer with conduit that can carry optical fiber
Single-mode optical fiber in an underground service pit
An optical fiber consists of a core, cladding, and a buffer (a protective outer coating), in
which the cladding guides the light along the core by using the method of total internal
reflection. The core and the cladding (which has a lower-refractive-index) are usually made
of high-quality silica glass, although they can both be made of plastic as well. Connecting
two optical fibers is done by fusion splicing or mechanical splicing and requires special skills
and interconnection technology due to the microscopic precision required to align the fiber
cores.[4]
Two main types of optical fiber used in optic communications include multi-mode optical
fibers and single-mode optical fibers. A multi-mode optical fiber has a larger core (≥ 50
micrometers), allowing less precise, cheaper transmitters and receivers to connect to it as well
as cheaper connectors. However, a multi-mode fiber introduces multimode distortion, which
often limits the bandwidth and length of the link. Furthermore, because of its higher dopant
content, multi-mode fibres are usually expensive and exhibit higher attenuation. The core of a
single-mode fiber is smaller (<10 micrometers) and requires more expensive components and
interconnection methods, but allows much longer, higher-performance links.
In order to package fiber into a commercially viable product, it typically is protectively
coated by using ultraviolet (UV), light-cured acrylate polymers, then terminated with optical
fiber connectors, and finally assembled into a cable. After that, it can be laid in the ground
and then run through the walls of a building and deployed aerially in a manner similar to
copper cables. These fibers require less maintenance than common twisted pair wires, once
they are deployed.[5]
Specialized cables are used for long distance subsea data transmission, e.g. transatlantic
communications cable. New (2011–2013) cables operated by commercial enterprises
(Emerald Atlantis, Hibernia Atlantic) typically have four strands of fiber and cross the
Atlantic (NYC-London) in 60-70ms. Cost of each such cable was about $300M in 2011.
source: The Chronicle Herald.
Another common practice is to bundle many fiber optic strands within long-distance power
transmission cable. This exploits power transmission rights of way effectively, ensures a
power company can own and control the fiber required to monitor its own devices and lines,
is effectively immune to tampering, and simplifies the deployment of smart grid technology.
Amplifiers
The transmission distance of a fiber-optic communication system has traditionally been
limited by fiber attenuation and by fiber distortion. By using opto-electronic repeaters, these
problems have been eliminated. These repeaters convert the signal into an electrical signal,
and then use a transmitter to send the signal again at a higher intensity than it was before.
Because of the high complexity with modern wavelength-division multiplexed signals
(including the fact that they had to be installed about once every 20 km), the cost of these
repeaters is very high.
An alternative approach is to use an optical amplifier, which amplifies the optical signal
directly without having to convert the signal into the electrical domain. It is made by doping a
length of fiber with the rare-earth mineral erbium, and pumping it with light from a laser with
a shorter wavelength than the communications signal (typically 980 nm). Amplifiers have
largely replaced repeaters in new installations.
Wavelength-division multiplexing
Wavelength-division multiplexing (WDM) is the practice of multiplying the available
capacity of optical fibers through use of parallel channels, each channel on a dedicated
wavelength of light. This requires a wavelength division multiplexer in the transmitting
equipment and a demultiplexer (essentially a spectrometer) in the receiving equipment.
Arrayed waveguide gratings are commonly used for multiplexing and demultiplexing in
WDM. Using WDM technology now commercially available, the bandwidth of a fiber can be
divided into as many as 160 channels to support a combined bit rate in the range of 1.6 Tbit/s.
Parameters
Bandwidth–distance product
Because the effect of dispersion increases with the length of the fiber, a fiber transmission
system is often characterized by its bandwidth–distance product, usually expressed in units of
MHz·km. This value is a product of bandwidth and distance because there is a trade off
between the bandwidth of the signal and the distance it can be carried. For example, a
common multi-mode fiber with bandwidth–distance product of 500 MHz·km could carry a
500 MHz signal for 1 km or a 1000 MHz signal for 0.5 km.
Engineers are always looking at current limitations in order to improve fiber-optic
communication, and several of these restrictions are currently being researched.
Record speeds
Each fiber can carry many independent channels, each using a different wavelength of light
(wavelength-division multiplexing). The net data rate (data rate without overhead bytes) per
fiber is the per-channel data rate reduced by the FEC overhead, multiplied by the number of
channels (usually up to eighty in commercial dense WDM systems as of 2008).
Year Organization Effective speed WDM
channels Per channel
2009 Alcatel-Lucent 15 Tbit/s
155
100 Gbit/s
2010 NTT
69.1 Tbit/s
432
171 Gbit/s
2011 KIT
26 Tbit/s
1
26 Tbit/s
2011 NEC
101 Tbit/s
370
273 Gbit/s
2012 NEC, Corning 1.05 Petabit/s 12 core fiber
speed Distance
90 km
240 km
50 km
165 km
52.4 km
While the physical limitations of electrical cable prevent speeds in excess of 10 Gigabits per
second, the physical limitations of fiber optics have not yet been reached.
In 2013, New Scientist reported that a team at the University of Southampton had achieved a
throughput of 73.7 Tbit per second, with the signal traveling at 99.7% the speed of light
through a hollow-core photonic crystal fiber.
Dispersion
For modern glass optical fiber, the maximum transmission distance is limited not by direct
material absorption but by several types of dispersion, or spreading of optical pulses as they
travel along the fiber. Dispersion in optical fibers is caused by a variety of factors. Intermodal
dispersion, caused by the different axial speeds of different transverse modes, limits the
performance of multi-mode fiber. Because single-mode fiber supports only one transverse
mode, intermodal dispersion is eliminated.
In single-mode fiber performance is primarily limited by chromatic dispersion (also called
group velocity dispersion), which occurs because the index of the glass varies slightly
depending on the wavelength of the light, and light from real optical transmitters necessarily
has nonzero spectral width (due to modulation). Polarization mode dispersion, another source
of limitation, occurs because although the single-mode fiber can sustain only one transverse
mode, it can carry this mode with two different polarizations, and slight imperfections or
distortions in a fiber can alter the propagation velocities for the two polarizations. This
phenomenon is called fiber birefringence and can be counteracted by polarizationmaintaining optical fiber. Dispersion limits the bandwidth of the fiber because the spreading
optical pulse limits the rate that pulses can follow one another on the fiber and still be
distinguishable at the receiver.
Some dispersion, notably chromatic dispersion, can be removed by a 'dispersion
compensator'. This works by using a specially prepared length of fiber that has the opposite
dispersion to that induced by the transmission fiber, and this sharpens the pulse so that it can
be correctly decoded by the electronics.
Attenuation
Fiber attenuation, which necessitates the use of amplification systems, is caused by a
combination of material absorption, Rayleigh scattering, Mie scattering, and connection
losses. Although material absorption for pure silica is only around 0.03 dB/km (modern fiber
has attenuation around 0.3 dB/km), impurities in the original optical fibers caused attenuation
of about 1000 dB/km. Other forms of attenuation are caused by physical stresses to the fiber,
microscopic fluctuations in density, and imperfect splicing techniques.
Transmission windows
Each effect that contributes to attenuation and dispersion depends on the optical wavelength.
The wavelength bands (or windows) that exist where these effects are weakest are the most
favorable for transmission. These windows have been standardized, and the currently defined
bands are the following:[13]
Band Description
Wavelength Range
O band original
1260 to 1360 nm
E band extended
1360 to 1460 nm
S band short wavelengths
1460 to 1530 nm
C band conventional ("erbium window") 1530 to 1565 nm
L band long wavelengths
1565 to 1625 nm
U band ultralong wavelengths
1625 to 1675 nm
Note that this table shows that current technology has managed to bridge the second and third
windows that were originally disjoint.
Historically, there was a window used below the O band, called the first window, at 800900 nm; however, losses are high in this region so this window is used primarily for shortdistance communications. The current lower windows (O and E) around 1300 nm have much
lower losses. This region has zero dispersion. The middle windows (S and C) around
1500 nm are the most widely used. This region has the lowest attenuation losses and achieves
the longest range. It does have some dispersion, so dispersion compensator devices are used
to remove this.
Regeneration
When a communications link must span a larger distance than existing fiber-optic technology
is capable of, the signal must be regenerated at intermediate points in the link by repeaters.
Repeaters add substantial cost to a communication system, and so system designers attempt
to minimize their use.
Recent advances in fiber and optical communications technology have reduced signal
degradation so far that regeneration of the optical signal is only needed over distances of
hundreds of kilometers. This has greatly reduced the cost of optical networking, particularly
over undersea spans where the cost and reliability of repeaters is one of the key factors
determining the performance of the whole cable system. The main advances contributing to
these performance improvements are dispersion management, which seeks to balance the
effects of dispersion against non-linearity; and solitons, which use nonlinear effects in the
fiber to enable dispersion-free propagation over long distances.
Last mile
Although fiber-optic systems excel in high-bandwidth applications, optical fiber has been
slow to achieve its goal of fiber to the premises or to solve the last mile problem. However, as
bandwidth demand increases, more and more progress towards this goal can be observed. In
Japan, for instance EPON has largely replaced DSL as a broadband Internet source. South
Korea‘s KT also provides a service called FTTH (Fiber To The Home), which provides fiberoptic connections to the subscriber‘s home. The largest FTTH deployments are in Japan,
South Korea, and China. Singapore started implementation of their all-fiber Next Generation
Nationwide Broadband Network (Next Gen NBN), which is slated for completion in 2012
and is being installed by Open Net. Since they began rolling out services in September 2010,
Network coverage in Singapore has reached 60% nationwide.
In the US, Verizon Communications provides a FTTH service called FiOS to select highARPU (Average Revenue Per User) markets within its existing territory. The other major
surviving ILEC (or Incumbent Local Exchange Carrier), AT&T, uses a FTTN (Fiber To The
Node) service called U-verse with twisted-pair to the home. Their MSO competitors employ
FTTN with coax using HFC. All of the major access networks use fiber for the bulk of the
distance from the service provider's network to the customer.
The globally dominant access network technology is EPON (Ethernet Passive Optical
Network). In Europe, and among telcos in the United States, BPON (ATM-based Broadband
PON) and GPON (Gigabit PON) had roots in the FSAN (Full Service Access Network) and
ITU-T standards organizations under their control.
Comparison with electrical transmission
A mobile fiber optic splice lab used to access and splice underground cables
An underground fiber optic splice enclosure opened up
The choice between optical fiber and electrical (or copper) transmission for a particular
system is made based on a number of trades-offs. Optical fiber is generally chosen for
systems requiring higher bandwidth or spanning longer distances than electrical cabling can
accommodate.
The main benefits of fiber are its exceptionally low loss (allowing long distances between
amplifiers/repeaters), its absence of ground currents and other parasite signal and power
issues common to long parallel electric conductor runs (due to its reliance on light rather than
electricity for transmission, and the dielectric nature of fiber optic), and its inherently high
data-carrying capacity. Thousands of electrical links would be required to replace a single
high bandwidth fiber cable. Another benefit of fibers is that even when run alongside each
other for long distances, fiber cables experience effectively no crosstalk, in contrast to some
types of electrical transmission lines. Fiber can be installed in areas with high
electromagnetic interference (EMI), such as alongside utility lines, power lines, and railroad
tracks. Nonmetallic all-dielectric cables are also ideal for areas of high lightning-strike
incidence.
For comparison, while single-line, voice-grade copper systems longer than a couple of
kilometers require in-line signal repeaters for satisfactory performance; it is not unusual for
optical systems to go over 100 kilometers (62 mi), with no active or passive processing.
Single-mode fiber cables are commonly available in 12 km lengths, minimizing the number
of splices required over a long cable run. Multi-mode fiber is available in lengths up to 4 km,
although industrial standards only mandate 2 km unbroken runs.
In short distance and relatively low bandwidth applications, electrical transmission is often
preferred because of its
Lower material cost, where large quantities are not required
Lower cost of transmitters and receivers
Capability to carry electrical power as well as signals (in appropriately designed
cables)
Ease of operating transducers in linear mode.
Optical fibers are more difficult and expensive to splice than electrical conductors. And at
higher powers, optical fibers are susceptible to fiber fuse, resulting in catastrophic destruction
of the fiber core and damage to transmission components.
Because of these benefits of electrical transmission, optical communication is not common in
short box-to-box, backplane, or chip-to-chip applications; however, optical systems on those
scales have been demonstrated in the laboratory.
In certain situations fiber may be used even for short distance or low bandwidth applications,
due to other important features:
Immunity to electromagnetic interference, including nuclear electromagnetic pulses
(although fiber can be damaged by alpha and beta radiation).
High electrical resistance, making it safe to use near high-voltage equipment or
between areas with different earth potentials.
Lighter weight—important, for example, in aircraft.
No sparks—important in flammable or explosive gas environments.
Not electromagnetically radiating, and difficult to tap without disrupting the signal—
important in high-security environments.
Much smaller cable size—important where pathway is limited, such as networking an
existing building, where smaller channels can be drilled and space can be saved in
existing cable ducts and trays.
Resistance to corrosion due to non-metallic transmission medium
Optical fiber cables can be installed in buildings with the same equipment that is used to
install copper and coaxial cables, with some modifications due to the small size and limited
pull tension and bend radius of optical cables. Optical cables can typically be installed in duct
systems in spans of 6000 meters or more depending on the duct's condition, layout of the duct
system, and installation technique. Longer cables can be coiled at an intermediate point and
pulled farther into the duct system as necessary.
Governing standards
In order for various manufacturers to be able to develop components that function compatibly
in fiber optic communication systems, a number of standards have been developed. The
International Telecommunications Union publishes several standards related to the
characteristics and performance of fibers themselves, including
ITU-T G.651, "Characteristics of a 50/125 µm multimode graded index optical fibre
cable"
ITU-T G.652, "Characteristics of a single-mode optical fibre cable"
Other standards specify performance criteria for fiber, transmitters, and receivers to be used
together in conforming systems. Some of these standards are:
100 Gigabit Ethernet
10 Gigabit Ethernet
Fibre Channel
Gigabit Ethernet
HIPPI
Synchronous Digital Hierarchy
Synchronous Optical Networking
Optical Transport Network (OTN)
TOSLINK is the most common format for digital audio cable using plastic optical fiber to
connect digital sources to digital receivers.
3.
Mention the applications of Microwave communication?
Microwaves are widely used for point-to-point communications because their small
wavelength allows conveniently-sized antennas to direct them in narrow beams, which can be
pointed directly at the receiving antenna. This allows nearby microwave equipment to use the
same frequencies without interfering with each other, as lower frequency radio waves do.
Another advantage is that the high frequency of microwaves gives the microwave band a very
large information-carrying capacity; the microwave band has a bandwidth 30 times that of all
the rest of the radio spectrum below it. A disadvantage is that microwaves are limited to line
of sight propagation; they cannot pass around hills or mountains as lower frequency radio
waves can.
Microwave radio transmission is commonly used in point communication on the surface of
the Earth, in satellite communications, and in deep space radio communications. Other parts
of the microwave radio band are used for radars, radio navigation systems, sensor systems,
and radio astronomy.
The next higher part of the radio electromagnetic spectrum, where the frequencies are above
30 GHz and below 100 GHz, are called "millimeter waves" because their wavelengths are
conveniently measured in millimeters, and their wavelengths range from 10 mm down to
3.0 mm. Radio waves in this band are usually strongly attenuated by the Earthly atmosphere
and particles contained in it, especially during wet weather. Also, in wide band of frequencies
around 60 GHz, the radio waves are strongly attenuated by molecular oxygen in the
atmosphere. The electronic technologies needed in the millimeter wave band are also much
more difficult to utilize than those of the microwave band.
Wireless transmission of information
One-way (e.g. television broadcasting) and two-way telecommunication using
communications satellite
Terrestrial microwave radio broadcasting relay links in telecommunications networks
including e.g. backbone or backhaul carriers in cellular networks linking BTS-BSC and BSCMSC.
A parabolic antenna for Erdfunkstelle Raisting, based in Raisting, Bavaria, Germany.
C bandhorn-reflector antennas on the roof of a telephone switching center in Seattle,
Washington, part of the U.S. AT&T Long Lines microwave relay network.
Wireless transmission of power
Proposed systems e.g. for connecting solar power collecting satellites to terrestrial
power grids
Parabolic (microwave) antenna
To direct microwaves in narrow beams for point-to-point communication links or
radiolocation (radar), a parabolic antenna is usually used. This is an antenna that uses a
parabolic reflector to direct the microwaves. To achieve narrow beamwidths, the reflector
must be much larger than the wavelength of the radio waves. The relatively short wavelength
of microwaves allows reasonably sized dishes to exhibit the desired highly directional
response for both receiving and transmitting.
Microwave radio relay
Dozens of microwave dishes on the Heinrich-Hertz-Turm in Germany.
Microwave radio relay is a technology for transmitting digital and analogsignals, such as
long-distance telephone calls, television programs, and computer data, between two locations
on a line of sight radio path. In microwave radio relay, microwaves are transmitted between
the two locations with directional antennas, forming a fixed radio connection between the two
points. The requirement of a line of sight limits the distance between stations to 30 or 40
miles.
Beginning in the 1950s and 1960s networks of microwave relay links, such as the AT&T
Long Lines system in the U.S., carried long distance telephone calls and television programs
between cities. These included long daisy-chained series of such links that traversed
mountain ranges and spanned continents. Much of the transcontinental traffic is now carried
by cheaper optical fibers and communication satellites, but microwave relay remains
important for shorter distances.
How microwave radio relay links are formed
Relay towers on Frazier Mountain, Southern California
Because the radio waves travel in narrow beams confined to a line-of-sight path from one
antenna to the other, they don't interfere with other microwave equipment, and nearby
microwave links can use the same frequencies. Antennas used must be highly directional
(High gain); these antennas are installed in elevated locations such as large radio towers in
order to be able to transmit across long distances. Typical types of antenna used in radio relay
link installations are parabolic antennas, dielectric lens, and horn-reflector antennas, which
have a diameter of up to 4 meters. Highly directive antennas permit an economical use of the
available frequency spectrum, despite long transmission distances.
Danish military radio relay node
Planning considerations
Because of the high frequencies used, a quasi-optical line of sight between the stations is
generally required. Additionally, in order to form the line of sight connection between the
two stations, the first Fresnel zone must be free from obstacles so the radio waves can
propagate across a nearly uninterrupted path. Obstacles in the signal field cause unwanted
attenuation, and are as a result only acceptable in exceptional cases. High mountain peak or
ridge positions are often ideal: Europe's highest radio relay station, the Richtfunkstation
Jungfraujoch, is situated atop the Jungfraujoch ridge at an altitude of 3,705 meters (12,156 ft)
above sea level.
Multiple antennas provide space diversity
Obstacles, the curvature of the Earth, the geography of the area and reception issues arising
from the use of nearby land (such as in manufacturing and forestry) are important issues to
consider when planning radio links. In the planning process, it is essential that "path profiles"
are produced, which provide information about the terrain and Fresnel zones affecting the
transmission path. The presence of a water surface, such as a lake or river, in the mid-path
region also must be taken into consideration as it can result in a near-perfect reflection (even
modulated by wave or tide motions), creating multipath distortion as the two received signals
("wanted" and "unwanted") swing in and out of phase. Multipath fades are usually deep only
in a small spot and a narrow frequency band, so space and/or frequency diversity schemes
would be applied to mitigate these effects.
The effects of atmospheric stratification cause the radio path to bend downward in a typical
situation so a major distance is possible as the earth equivalent curvature increases from
6370 km to about 8500 km (a 4/3 equivalent radius effect). Rare events of temperature,
humidity and pressure profile versus height, may produce large deviations and distortion of
the propagation and affect transmission quality. High intensity rain and snow must also be
considered as an impairment factor, especially at frequencies above 10 GHz. All previous
factors, collectively known as path loss, make it necessary to compute suitable power
margins, in order to maintain the link operative for a high percentage of time, like the
standard 99.99% or 99.999% used in 'carrier class' services of most telecommunication
operators.
The longest microwave radio relay known up to date cross the Red Sea with 360 km hop
between Jebel Erba (2170m a.s.l., 20°44'46.17"N 36°50'24.65"E, Sudan) and Jebel Dakka
(2572m a.s.l., 21° 5'36.89"N 40°17'29.80"E, Saudi Arabia). The link built in 1979 by Telettra
allowed to proper transmit 300 telephone channels and 1 TV signal, in the 2 GHz frequency
band. (Hop distance is the distance between two microwave stations)
Portable microwave rig for Electronic news gathering (ENG) for television news
Over-horizon microwave radio relay
In over-horizon, or tropospheric scatter, microwave radio relays, unlike a standard microwave
radio relay link, the sending and receiving antennas do not use a line of sight transmission
path. Instead, the stray signal transmission, known as "tropo - scatter" or simply "scatter,"
from the sent signal is picked up by the receiving station. Signal clarity obtained by this
method depends on the weather and other factors, and as a result a high level of technical
difficulty is involved in the creation of a reliable over horizon radio relay link. Over horizon
radio relay links are therefore only used where standard radio relay links are unsuitable (for
example, in providing a microwave link to an island).
Usages of microwave radio relay systems
During the 1950s the AT&T Long Lines system of microwave relay links grew to carry the
majority of US long distance telephone traffic, as well as intercontinental television network
signals.[2] The prototype was called TDX and was tested with a connection between New
York City and Murray Hill, the location of Bell Laboratories in 1946. The TDX system was
set up between New York and Boston in 1947. The TDX was improved to the TD2, which
still used klystron tubes in the transmitters, and then later to the TD3 that used solid state
electronics. The main motivation in 1946 to use microwave radio instead of cable was that a
large capacity could be installed quickly and at less cost. It was expected at that time that the
annual operating costs for microwave radio would be greater than for cable. There were two
main reasons that a large capacity had to be introduced suddenly: Pent up demand for long
distance telephone service, because of the hiatus during the war years, and the new medium
of television, which needed more bandwidth than radio.
Though not commonly known, the US Military used both portable and fixed-station
microwave communications in the European Theater during WWII. Starting in the late 1940s,
this continued to some degree into the 1960s, when many of these links were supplanted with
tropospheric scatter or satellite systems. When the NATO military arm was formed, much of
this existing equipment was transferred to communications groups. The typical
communications systems used by NATO during that time period consisted of the
technologies which had been developed for use by the telephone carrier entities in host
countries. One example from the USA is the RCA CW-20A 1–2 GHz microwave relay
system which utilized flexible UHF cable rather than the rigid waveguide required by higher
frequency systems, making it ideal for tactical applications. The typical microwave relay
installation or portable van had two radio systems (plus backup) connecting two LOS sites.
These radios would often provide communication for 24 telephone channels of frequency
division multiplexed signal (i.e. Lenkurt 33C FDM), though any channel could be designated
to carry up to 18 teletype communications instead. Similar systems from Germany and other
member nations were also in use.
Similar systems were soon built in many countries, until the 1980s when the technology lost
its share of fixed operation to newer technologies such as fiber-optic cable and
communication satellites, which offer lower cost per bit.
During the Cold War, the US intelligence agencies, such as the National Security
Agency (NSA), were reportedly able to intercept Soviet microwave traffic using satellites
such as Rhyolite.[3] Much of the beam of a microwave link passes the receiving antenna and
radiates toward the horizon, into space. By positioning a geosynchronous satellite in the path
of the beam, the microwave beam can be received.
At the turn of the century, microwave radio relay systems are being used increasingly in
portable radio applications. The technology is particularly suited to this application because
of lower operating costs, a more efficient infrastructure, and provision of direct hardware
access to the portable radio operator.
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Microwave link
A microwave link is a communications system that uses a beam of radio waves in the
microwave frequency range to transmit video, audio, or data between two locations, which
can be from just a few feet or meters to several miles or kilometers apart. Microwave links
are commonly used by television broadcasters to transmit programmes across a country, for
instance, or from an outside broadcast back to a studio.
Mobile units can be camera mounted, allowing cameras the freedom to move around without
trailing cables. These are often seen on the touchlines of sports fields on Steadicam systems.
Properties of microwave links
Involve line of sight (LOS) communication technology
Affected greatly by environmental constraints, including rain fade
Have very limited penetration capabilities through obstacles such as hills, buildings
and trees
Sensitive to high pollen count[citation needed]
Signals can be degraded[citation needed]during Solar proton events
Uses of microwave links
In communications between satellites and base stations
As backbone carriers for cellular systems
In short range indoor communications
Telecommunications, in linking remote and regional telephone exchanges to larger
(main) exchanges without the need for copper/optical fibre lines.
Frequency ranges from 150 MHz to 150 GHz
Microwave power transmission
Microwave power transmission (MPT) is the use of microwaves to transmit power through
outer space or the atmosphere without the need for wires. It is a sub-type of the more general
wireless energy transfer methods.
History
Following World War II, which saw the development of high-power microwave emitters
known as cavity magnetrons, the idea of using microwaves to transmit power was researched.
In 1964, William C. Brown demonstrated a miniature helicopter equipped with a combination
antenna and rectifier device called a rectenna. The rectenna converted microwave power into
electricity, allowing the helicopter to fly. In principle, the rectenna is capable of very high
conversion efficiencies - over 90% in optimal circumstances.
Most proposed MPT systems now usually include a phased array microwave transmitter.
While these have lower efficiency levels they have the advantage of being electrically steered
using no moving parts, and are easier to scale to the necessary levels that a practical MPT
system requires.
Using microwave power transmission to deliver electricity to communities without having to
build cable-based infrastructure is being studied at Grand Bassin on Reunion Island in the
Indian Ocean.
Common safety concerns
The common reaction to microwave transmission is one of concern, as microwaves are
generally perceived by the public as dangerous forms of radiation - stemming from the fact
that they are used in microwave ovens. While high power microwaves can be painful and
dangerous as in the United States Military's Active Denial System, MPT systems are
generally proposed to have only low intensity at the rectenna.
Though this would be extremely safe as the power levels would be about equal to the leakage
from a microwave oven, and only slightly more than a cell phone, the relatively diffuse
microwave beam necessitates a large receiving antenna area for a significant amount of
energy to be transmitted.
Research has involved exposing multiple generations of animals to microwave radiation of
this or higher intensity, and no health issues have been found.
Proposed uses
MPT is the most commonly proposed method for transferring energy to the surface of the
Earth from solar power satellites or other in-orbit power sources. MPT is occasionally
proposed for the power supply in beam-powered propulsion for orbital lift space ships. Even
though lasers are more commonly proposed, their low efficiency in light generation and
reception has led some designers to opt for microwave based systems.
Current status
Wireless Power Transmission (using microwaves) is well proven. Experiments in the tens of
kilowatts have been performed at Goldstone in California in 1975 and more recently (1997)
at Grand Bassin on Reunion Island. In 2008 a long range transmission experiment
successfully transmitted 20 watts 92 miles (148 km) from a mountain on Maui to the main
island of Hawaii.
4.
Give a brief discussion about satellite and its applications?
A satellite is an object that orbits or revolves around another object. For example, the Moon
is a satellite of Earth, and Earth is a satellite of the Sun. In this document, we will examine
human-made satellites that orbit Earth. They are highly specialized wireless
receiver/transmitters that are launched by a rocket and placed in orbit around the Earth. There
are hundreds of satellites currently in operation.
Satellite communication is one particular example of wireless communication systems.
Similar and maybe more familiar examples of wireless systems are radio and television
broadcasting and mobile and cordless telephones. Systems of this type rely on a network of
ground-based transmitters and receivers. They are commonly referred to as 'terrestrial'
systems as opposed to satellite systems.
Satellite communication systems differ from terrestrial systems in that the transmitter is not
based on the ground but in the sky: the transmitter here consists of a ground-based part called
the uplink, and the satellite-based part that 'reflects' the signals towards the receivers. This
part is called the transponder.
Purpose
Satellites come in many shapes and sizes and have many uses. The first artificial satellite,
called Sputnik, was launched by the Soviet Union in 1957 and was the size of a basketball. Its
purpose was simply to transmit a Morse code signal repeatedly. In contrast, modern satellites
can receive and transmit hundreds of signals at the same time, from simple digital data to
complex television programmes. They are used for many purposes such as television
broadcasting, amateur radio communications, Internet communications, weather forecasting
and Global Positioning Systems (GPS).
Communications satellites act as relay stations in space. One could imagine them as very
long, invisible poles that relay high frequency radio waves. They are used to bounce
messages from one part of the world to another. The messages can be telephone calls, TV
pictures or Internet connections. Certain communications satellites are, for example, used for
broadcasting: they send radio and TV signals to homes. Nowadays, there are more than 100
such satellites orbiting Earth, transmitting thousands of different TV (and radio) programmes
all over the world.
Other applications: remote-sensing satellites
Military, government, weather, environment, scientific, positioning
Remote-sensing satellites study the surface of the Earth. From a relatively low height (480
km) up, these satellites use powerful cameras to scan the planet. The satellite then transmits
valuable data on the global environment to researchers, governments, and businesses
including those working in map making, farming, fishing, mining and many other industries.
Instruments on remote-sensing satellites gather data about features such as the Earth's plant
cover, chemical composition and surface water. Remote-sensing satellites are also used to
study changes in the Earth's surface caused by human activity. Examples of this kind of
observation include investigations into presence of ozone and greenhouse gasses in the
atmosphere, the desertification in West Africa and deforestation in South America.
Weather satellites record weather patterns around the world. Almost all countries use the data
coming from satellites like TIROS (Television Infrared Observational Satellite) ENVISAT to
forecast weather, track storms and carry out scientific research. TIROS is part of a system of
weather satellites operated by the National Oceanic and Atmospheric Administration
(NOAA). There are two TIROS satellites circling Earth over the poles. They work with
another set of satellites in geosynchronous orbit called Geostationary Operational
Environmental Satellites (GOES), such as the Meteosat satellites. Using this group of
satellites, meteorologists study weather and climate patterns around the world.
Many satellites in orbit conduct scientific experiments and observations. SOHO (SOlar and
Heliospheric Observation) for instance is an international project managed by Europe and the
United States. Its very sophisticated instruments can measure activity inside the Sun, look at
its atmosphere or corona and study its surface. SOHO does not orbit Earth. In fact it orbits the
Sun, about a million miles away from Earth. In that position neither the Moon nor the Earth
can block its clear view of the Sun.
The military have developed the Global Positioning System (GPS), but now people are using
these satellite services to determine their exact latitude, longitude and altitude wherever they
are in the world. GPS satellites can be used for navigation almost everywhere on Earth: in an
airplane, boat, or car, on foot, in a remote wilderness, or in a big city. GPS uses radio signals
from at least three satellites in sight to calculate the position of the receiver.
Military and government institutions make extensive use of satellites for a mixture of
communication, remote sensing, imaging, positioning and other services, as well as for more
secret applications such as spying or missile guidance. Extremely useful civilian technology
spin-offs resulted from developments in this sector, although GPS originated as a military
application. The domains of image processing and image recognition also benefited greatly
from Military Research and Development.
Although the purpose of this report is not to train future satellite engineers, there are certain
parts of a satellite system that are worth knowing about and which can help the reader
understand how satellites behave and how they can be used for different purposes. From this
point onwards we will focus almost exclusively on communication satellites, particularly
those parts and elements that are relevant to satellite communications.
The two most important elements of the communications system are the satellite itself and the
Earth station.
Earth station is the common name for every installation located on the Earth's surface and
intended for communication (transmission and/or reception) with one or more satellites. Earth
stations include all devices and installations for satellite communications: handheld devices
for mobile satellite telephony, briefcase satellite phones, satellite TV reception, as well as
installations that are less familiar, eg VSAT stations and satellite broadcast TV stations. The
term Earth station refers to the collection of equipment that is needed to perform
communications via satellite: the antenna (often a dish) and the associated equipment
(receiver/decoder, transmitter).
Antennas vary in size to match the particular service for which they are designed. For
example, a 70 cm antenna may be sufficient for direct reception of satellite TV programmes
in the home, but would not be suitable for, for example, transmitting television programmes.
Handheld satellite telephone, antenna for satellite
TV reception, satellite transmitting Earth station
The other part of the Earth station is the application device which, in the case of reception,
translates radio signals into information that can be displayed on a TV screen or processed by
a computer. In the case of transmission, this device will transform information into a signal
that is suitable for transmission via the antenna, using modulation, amplification and other
processing techniques. In the case of VSAT- type two-way systems, both send and receive
functions can be carried out at the same time.
The two main parts in the sky common to all satellites are called the payload and the bus.
The payload represents all equipment a satellite needs to do its job. This can include
antennas, cameras, radar and electronics. The payload is different for every satellite. For
example, the payload for a weather satellite includes cameras to take pictures of cloud
formations, while the payload for a communications satellite includes large antennas to
transmit TV or telephone signals to Earth.
The transponder is the key component for satellite communications: it is the part of the
payload that takes the signals received from the transmitting Earth station, filters and
translates these signals and then redirects them to the transmitting antenna on board.
Communications satellites carry a large number of transponders on board (normally from six
to more than 24), enabling them to deliver multiple channels of communication at the same
time.
These
channels
are
called
carriers.
There are two main types of transponders. The 'bent pipe repeater' does not actually process
the signal at all. The second type of transponder, the 'onboard processor', can introduce digital
detection for the uplink signal and subsequent digital switching and modulation for the
downlink. Onboard processing is a major step in the implementation of new technologies
onto satellites. In the case of Iridium and many of the Internet access satellites, satellites act
as mini switchboards in the sky.
Communications satellites carry, as part of their payload, antennas that receive the original
signal from the transmitting Earth station and re-transmit this signal to the receive stations on
Earth. The antennas that were used in the past to do this were omni-directional (transmitting
signals in every direction) and not very effective. They were replaced by more efficient highgain antennas (most often dish shaped) pointing quite precisely towards the areas they were
servicing. To allow for flexibility in services or areas covered, later developments allowed
the re-pointing of the so-called steerable antenna to cover a different area or to reshape or
reformat the beam. Future developments will allow for a highly precise and efficient
reshaping of the transmitted beam in order to cover very small areas (pencil beams). This will
greatly facilitate the differentiation of services within large regions. The antennas on board
the satellite are typically limited in size to around 2-3 m by the space that is available on the
satellite structure.
Communications satellite
Bus: physical platform, remote control
The bus is the part of the satellite that carries the payload and all its equipment into space. It
is the physical platform that holds all the satellite's parts together and that provides electrical
power, navigation, control and propulsion to the spacecraft. The bus also contains equipment
that allows the satellite to communicate with Earth, a kind of 'remote control'.
Orbits:
GEO,
MEO,
LEO,
elliptical,
polar
The most common type of communications satellites, particularly the broadcast satellites like
AfriStar, Intelsat, PanAmSat, Eutelsat and ASTRA, are in geosynchronous orbit (from geo =
Earth + synchronous = moving at the same rate). That means that the satellite always stays
over one spot on Earth. It does this by placing the satellite in a position 35,786 km out in
space perpendicularly above the equator. The imaginary ring around the Earth where all
geostationary satellites are stationed for their lifetime is called the Clarke belt. The
consequence of this type of fixed location is that Earth stations (receive as well as transmit
stations on the Earth surface) can almost be permanently fixed because they are constantly
pointed to the same point in the sky where they 'see' the satellite.
A medium Earth orbit (MEO) satellite is one with an orbit from a few hundred miles to a few
thousand miles above the Earth's surface. Satellites of this type are in a higher orbit than low
Earth orbit (LEO) satellites, but lower than geostationary (GEO) satellites. The orbital
periods (the time in between two successive passes over one particular place on Earth) of
MEO satellites range from about 2 to 12 hours. Some MEO satellites orbit in near perfect
circles, therefore they have constant altitude and travel at a constant speed. Others have a
more elliptical shaped orbit, which results in different fly-over times according to the place
on Earth from where they can be seen. A fleet of several MEO satellites with properly
coordinated orbits can provide global coverage. There are several advantages of the use of
MEO satellites: because they are closer to the Earth's surface than geostationary satellites,
they require less power to transmit. The Earth stations (transmitters and receivers) by
consequence can be much smaller and have a small rod-shaped antenna. It is possible to use
mobile and even handheld terminals with such systems.
Low earth orbiting satellite system
A low Earth orbit (LEO) satellite system consists of a large number of satellites each in a
circular orbit at a constant altitude between 320 and 800 km. Because they orbit so close to
Earth, they must travel very fast so gravity does not pull them back into the atmosphere.
Satellites in LEOs circle around the Earth at 27,359 km per hour. The orbits take the satellites
over the geographic poles. Each revolution takes from less than 90 minutes up to a few hours.
The fleet is arranged in such a way that from any point on the surface at any time at least one
satellite is in line of sight. The system operates in a cellular network structure (almost like
mobile phones). The main difference is that in a mobile telephone network the relay towers or
aerials are fixed on the Earth while with satellites these aerials (called transponders or
wireless receiver/transmitters) are moving in space. LEO systems may form the space
segment of future mobile phone systems (such as S-UMTS) that will allow true mobile,
global, broadband multimedia connectivity. But although telecoms experts predicted a bright
future for this technology in the beginning of this century, to date only a few systems have
actually got off the ground.
Footprints: global, regional, spot beams
The area on Earth that the satellite can 'see' (or reach with its antennas) is called the satellite
'footprint'. A satellite's footprint refers to the area over which the satellite operates: the
intersection of a satellite antenna transmission pattern and the surface of the Earth.
Global coverage requires that the pattern of satellite antenna transmission covers the largest
possible portion of the Earth that can be viewed from the satellite. For geostationary
satellites, the beam width for global coverage is about 17.4 degrees.
No satellite can cover the whole surface of the Earth at one time: to achieve a global
coverage, multiple transmission beams from at least 3 different satellites are combined.
Combining footprints from Intelsat APR-1, 511 and 701 providing global coverage
The map above shows examples of how different satellites cover different areas. The
combined Intelsat satellite footprints on this map cover the whole Earth. A person in
Australia can use this satellite to communicate with anyone in Alaska. In combination with
the regional beams from these satellites, communication can be established between many
areas simultaneously.
Regional or zone coverage is the result of a partial illumination of the global coverage area.
The area may have a simple shape such as a circle or ellipse or may be irregularly shaped
(contoured) to cover certain areas most effectively, for example the shape of a continent or
sub-continent. Typical regional beams measure around 5 degrees in width.
Regional coverage of the Eutelsat W1
Spot beam coverage towards South Africa from the Eutelsat W1
Spot beam coverage is an area that is much smaller than global coverage. The beam width is
reduced to around 2 degrees. Spot beams have the advantage of high antenna gain, but are
disadvantaged because they can only cover a smaller area. This drawback can be overcome
by the combination of multiple spot beams.
Most geostationary telecommunication satellites cover large regions (continents or subcontinents). Sometimes satellites cover different areas at the same time from where they are
positioned. For example: the Eutelsat W1 satellite, a typical broadcast satellite, positioned at
10 degrees East, provides a high-power coverage of Europe with a total of 20 channels. In
addition, the satellite provides a high-power steerable narrow coverage carrying another eight
channels directed towards southern Africa (see map above).
Being on the edge of the satellite footprint means the curvature of the Earth starts to disrupt
transmission. It also means being further away from the satellite and therefore having to
transmit or receive over larger distances through the atmosphere than would be required if
transmitting/receiving from the centre of the footprint. Antenna size and power by
consequence are invariably increased at the edge of the footprint. These values can be
deducted from the footprint maps that are published by satellite service operators (see maps
above). The numbers on the circles on the maps above indicate the signal strength received at
that location expressed in dBW. From tables like the one below, users who wish to receive a
transmission can read what size antenna they need. The size varies depending on the
meteorological conditions of the location: places with regular heavy rainfall will need the
larger
dimension.
Antenna Size and Signal Strength
Frequency bands
Satellite communications, like any other means of communication (radio, TV, telephone, etc),
use frequency bands that are part of the electromagnetic spectrum. The electromagnetic
radiation spectrum starts with the longest waves (including those in the audible range) and
extends through radio waves and the visible light, which is effectively a very small part of the
spectrum, all the way to the extremely short wavelengths such as radioactive radiation.
Within this broad range of frequencies, the International Telecommunications Union (the
United Nations institution that regulates worldwide use of airwaves) has allocated parts of the
spectrum that are suitable for and dedicated to transmission via satellite. Some of these bands
are exclusively dedicated to satellite transmission; others are shared with terrestrial
transmission
services.
Satellite communications spectrum
The satellite transmission bands that are of interest to us are the C-, Ku- and Ka-bands.
C-band is the oldest allocation and operates in the frequency range around 6 GHz for
transmission (uplink) and between 3.7 and 4.2 GHz for reception (downlink).
Ku-band is the most common transmission format in Europe for satellite TV and uses around
14 GHz for uplink and between 10.9 and 12.75 GHz for downlink.
Ka-band uses around 30 GHz up- and between 18 and 20 GHz downlink frequency.
C-band and Ku-band are becoming congested by an increasing amount of users, so satellite
service operators are more and more turning to the use of Ka-band.
The selection of the band is not something that individual service providers decide, but is
rather chosen by large satellite operators based on different factors:
Availability: C-band is still the most widely available worldwide. Ku-band is
becoming more available recently in regions which were less covered in the past
(South America, Asia, Africa)
C-band is more prone to interference from other transmission services that share the
same frequencies (adjacent satellites or terrestrial transmissions) than the higher bands
While the C-band technology is cheaper in itself, it requires larger dishes (1 to 3 m)
than Ku- and Ka-band (0.6 to 1.8 m) and therefore imposes relatively higher
(installation) costs on the end-user
Ku- and especially Ka-band make better use of satellite capacity
Higher frequency bands (Ku- and especially Ka-) suffer significantly more from
signal deterioration caused by rainfall: to ensure availability in bad weather
conditions, the signal has to be much stronger. Note that 0.1% of unavailability means
in fact that the service will be interrupted for almost 9 hours over a 1-year period. 1%
unavailability represents 90 hours or almost 4 full days
Satellite control and lifetime
In principle, geostationary satellites occupy a fixed position in space and consequently the
ground-based antennas do not need to be constantly redirected to follow the satellite‘s
movements. The fact that the orientation of ground-based antennas is fixed is a major
advantage of the geostationary satellite orbit used by satellite broadcasters.
In practice however, the satellite wanders slightly around its nominal orbital position under
the gravitational influence of bodies such as the Sun and the Moon, as well as other
influences such as Sun radiation pressure and Earth asymmetry.
It is therefore necessary to take corrective actions in order to keep the satellite within
acceptable margins from its ideal position. This is achieved by activating the so-called
‗thrusters‘ that are mounted on the body of the satellite as part of its propulsion system.
As long as the satellite has enough fuel left to operate its thrusters, it can be kept in the
correct position. As soon as the satellite is out of fuel, it will drift out of control and into
space, which brings an end to its operational life. The satellite service operator can decide to
save on fuel (and by consequence extend the lifetime expectancy of a satellite) by allowing
the satellite to drift a little bit. Although this may bring down the costs for the communication
via the satellite considerably, there is a consequence on the Earth station side. These stations
have to be equipped for tracking (following the drift of) the satellite. The Earth stations that
are used with LEO and GEO systems use omni-directional antennas that make precise
pointing of the antenna unnecessary.
However, for this application, the ability to ‗see‘ the satellite (line of sight should not be
obstructed by walls, roofs, excess foliage) is still required, which means that indoor use is
excluded.
The communication functions of a satellite (antennas, processors) are powered by electricity
provided through a combination of solar energy and batteries. These batteries automatically
take over the power supply from the large wing-shaped solar cell panels at moments when the
satellite
finds
itself
in
the
shadow
of
the
Earth.
LEOs and MEOs spin around the Earth at high speeds in order to resist the Earth‘s
gravitational forces. They are designed to be cheaper and therefore are smaller and lighter
than large GEOs. They take less fuel to correct their flight paths and in most cases have a
shorter life expectancy than GEOs. LEO operators expect to renew their satellite fleet
between 5 and 7 years. GEO operators estimate the lifetime of their satellites to be between
10 and 12 years.
Applications of Satellite Communications Technology
Satellite communications systems differ from terrestrial systems in one obvious and
important aspect - the transmitter is no longer located on the ground but rather in the sky.
Because it's positioned in space, it is able to serve a very large geographical area. This has
several
advantages.
As few as three geostationary satellites can cover almost the whole of the Earth's surface,
with the exclusion of the sparsely populated Polar Regions. To achieve the same coverage by
terrestrial means would require a very large and expensive network of ground-based
transmitters.
Services can be established quickly, since coverage is available for everyone from the day
transmissions start. There is no need for a phased introduction as is the case with groundbased transmissions where antennas need to be added to meet the expansion of the serviced
area. With satellite communications, even users in very remote locations enjoy the same level
of
service
as
any
other
user
in
the
coverage
area.
Satellites can overcome national boundaries, providing possibilities for truly international
services.
Although terrestrial systems may be better suited generally to provide communications
services, in many cases the need to be connected can only be met effectively and rapidly by
the implementation of satellite services.
Radio and TV Broadcasting
The most familiar use of satellites is television broadcasting. TV satellites deliver hundreds of
television channels every day throughout the world. These satellites are even used to supply
television signals to terrestrial transmitters or cable-head end stations for further distribution
to the home, or to exchange signals between television studios. The bandwidth required to
transmit multiple programmes at the same time can easily be provided using satellites. In
addition, developments in broadcast technology (digitalisation, multiplexing and
compression) allow different types of transmissions to be sent sharing the same satellite
signal. To address the largest possible number of viewers, the cost to the viewers must be
small, requiring small receive antennas and cheap receivers.
Satellite service operators such as Intelsat, Eutelsat, ASTRA, PanAmSat, NileSat, AsiaSat
and AfriStar carry the signals for satellite broadcasters such as BSkyB, CanalPLus,
Multichoice, DirecTV and WorldSpace. These in turn bundle programmes from different
public and private broadcasters in order to make them accessible for their viewers in an open
('free-to-air') or closed (restricted) way. Some satellite broadcasters bundle special offers into
so-called 'bouquets of services' that are offered at additional cost.
The importance of satellite TV broadcasting is enormous: at the moment Eutelsat broadcasts
over 900 TV channels and 560 radio stations to more than 84 million satellite and cable
homes, the vast majority of them via the five HOT BIRD satellites at 13 degrees East.
ASTRA, another European direct-to-home satellite system, transmits more than 1,000
television and radio channels in analogue and digital format to an audience of more than 89
million homes throughout Europe. It does this via 12 satellites at the orbital positions of 19.2
degrees, 24.2 degrees and 28.2 degrees East.
In order to make their offer more attractive to broadcasters, satellite service operators try to
place their satellites aimed at the same regional market as far as possible in one single
position. This is why we find the Eutelsat HotBird constellation at 13 degrees East or the
ASTRA position at 19 degrees East, where in each case a number of satellites are clustered.
In consequence, viewers need to point their antennas in one direction only in order to receive
a large number of satellite programmes coming effectively from different satellites but
looking as if they come from only one.
Satellite TV reception antenna
There are many different applications of satellite TV viewing, depending on the needs and
objectives of the broadcaster or the viewers. Direct-to-Home or DTH - also called DBS or
Direct Broadcast via Satellite - speaks for itself: the TV programmes are aimed at the
consumer and transmitted in such a way that residential customers can buy and install the
equipment to receive the programmes at the lowest possible expense. This requires a network
of local resellers that offer the hardware (satellite receive equipment), installers (technicians
that assist the customer in setting up the receive equipment) and service suppliers.
The Eutelsat HotBird position at 13 Degrees East
Programme suppliers can opt for free-to-air programming, where every viewer with a
standard satellite receiver can receive and view the programme without restrictions.
However, some programmes contain information that is not for public viewing. To protect
these programmes so that only those who are the targeted audience will be able to view the
contents, some type of conditional access can be applied. What happens is that programmes
are encrypted and must then be unscrambled with a specific device (usually integrated in the
receiver and therefore often called an Integrated Receiver Decoder or IRD) to view the
contents.
The
move
from
analogue
to
digital
services
The number of analogue channels transmitted and the number of homes receiving analogue
continues to decrease. However, analogue takes up a significant portion of the range of
frequencies available. In addition, even in space, transmission capacity is limited. Where an
analogue signal will occupy a full transponder consuming a bandwidth of 36 or even 72
MHz, digital broadcasting makes it possible to compress signals, vastly increasing the
number of channels available by combining multiple programmes onto one single
transponder.
Nowadays, most digital TV signals are compliant to the MPEG-2/DVB standard and can be
received with standard consumer digital reception equipment that decodes the signal and
separates the different types of content out of the data stream. With transmission bit rates
between 34 and 38 Mbps, a digital signal can carry a combination of up to 12 television
channels,
along
with
numerous
radio
transmissions
and
data.
Consequently, it is digital television that is now driving the satellite TV market, aiming at
large numbers of consumers equipped with small antennae of typically 50 to 80 cm in
diameter in Western Europe and 1.2 to 1.5 m in diameter in other regions. Digital technology
has spurred the development of interactivity and aided the convergence of the worlds of
television,
radio,
personal
computing
and
telephony.
It appeals to the end-user by providing better video and audio quality, improved programme
and
service
choice
and
greater
control
over
content
delivery.
Pay TV is a service where the viewer is charged according to the programmes she/he views,
selected from the TV programme on offer. Video on Demand and Near Video on Demand
enables individual viewers to decide at any given moment (in the case of real Video on
Demand) or at a later time to be scheduled (in the case of Near Video on Demand) to view
the programme of his/her own choice. IP-TV is a Video on Demand application using
compression technologies that allow highly efficient distribution of video and audio using
common multimedia formats such as MPEG-1, 2 and 4. Streaming technologies are based on
the Internet Protocol, which allows delivery over all kinds of networks including the Internet.
The latest development in advanced television applications including delivery via satellite is
the Personal Video Recorder (PVR). These devices are used both to digitally record and play
back programmes: the programme provider sends the content the normal way (TV networks,
cable, satellite). At the receive end the content is fed into the PVR. Compression such as
MPEG-2 and MPEG-4 is used to decrease the bandwidth. The PVR unit is basically a
computer that saves the incoming live TV signal from the cable or antenna onto its large
internal hard drive. In this way, the viewer can play it back with a few seconds delay. The
viewer is then watching off the computer hard drive, instead of straight from the antenna or
cable connection. This allows the viewer to rewind, slow down, stop and pause at any point.
Broadcasters and content providers are able to improve their service offers. New satellite
facilities that are being offered or under development are Pay TV services, (Near) Video on
Demand, IP-TV delivery and Personal TV using devices such as the Personal Video Recorder
(PVR) or the Multimedia Home Platform (MHP), services that are called interactive TV or
enhanced TV. While the concept of interactive TV (iTV) is not new (there have been
numerous interactive TV pilot services and some limited applications have been rolled in
parts of the world), the broadcast world seems to be waiting for final and concrete iTV
standard protocols and definitions to give the public uptake the expected boost. Interactive or
enhanced services such as electronic programme guides, on-screen games, quizzes, enhanced
TV with retrievable background information on demand, recording, rewinding, even Internet
services such as mail and web browsing, are all applications that are not only appealing new
services for leisure TV but will also find applications within the education domain where the
high degree of penetration of TV sets and transmit capacity combined with the low threshold
of the television programmes for the end-user, may well become a strong argument for the
use
of
interactive
TV
in
educational
communities.
The Multimedia Home Platform (MHP) is a software specification that will be implemented
in set-top boxes, integrated digital TV receivers as well as multimedia PCs. The MHP will
connect the broadcast medium with the Internet, television, computer and telecommunication
and enables digital content providers to address all types of terminals ranging from low-end
to high-end set-top boxes, integrated digital TV sets and multimedia PCs.
But even if the broadcasting world is rapidly going digital, analogue TV and radio are set to
remain for several years yet. For some services, analogue is still the most attractive option
due to the large installed audience base and the widespread availability of consumer
equipment that is less expensive than digital equipment. Analogue is particularly popular for
free-to-air broadcasting. Moreover, the capacity to transmit several audio sub-carriers on one
analogue TV signal allows multilingual TV programmes. As for analogue radio broadcasting,
up to eight mono channels or four stereo pairs can be transmitted as sub-carriers of an
analogue television signal. However, due to the heavy use of analogue sub-carriers and the
decreasing number of transponders used for analogue TV, analogue sub-carriers are
becoming less and less available. Service providers considering new audio services should
therefore consider digital. When you take into account the cost of transmission and the
numerous innovative applications that are becoming available, such as the PVR, encryption
and the ability to carry data for multicasting, the appeal of digital broadcasting is hard to
resist.
Business radio and TV
Narrowcasting or business TV and radio is a term used for satellite broadcasters who use
transmission time to reach a very specific audience. Technically speaking, there is no
difference with broadcast satellite TV applications described in the previous section. Digital
television has made it possible to distribute information within organisations and companies
that are geographically dispersed, or to deliver distance education. Similarly, digital radio
allows for the delivery of radio services to relatively small closed user groups.
MPEG-2/DVB technology is the dominant standard for digital television, but other computerbased media coding techniques (such as MPEG-1, Real Video, etc) are also used to embed
video and/or audio into data streams, often integrated with other multimedia or Internet
services. Transmission via satellite requires there to be digital receivers available at relatively
low prices on the consumer market. The advantage is that more advanced or popular audio
coding techniques (for example MP3) can also be used and that the same stream can be used
for other applications, such as data distribution, outside broadcast hours.
Contribution ('backhaul' and SNG)
Satellite transmission technologies can be used to bring the signal that needs to be broadcast
to the place where it can be processed and prepared for re-distribution, for example: to a
broadcaster's main studio; to a number of cable-head end stations; to an Internet Service
Provider where it can be injected into the Internet; or to a network of local Points-of-Presence
for distribution in local networks. These links respond to the need for point-to-point and
point-to-multipoint transmission and are often called a 'hop'. The signal can be digital or
analogue and can include video, audio, data or multimedia.
When used by news companies this type of contribution link is called Satellite News
Gathering (SNG). News and information are sent from a mobile station - a truck equipped
with an antenna or a suitcase uplink - through the satellite to a central point, which is in most
cases the home studio, equipped with an Earth station with a fixed antenna. The home studio
can retransmit it live or record and re-edit it for later use.
Telephony
Thin route or trunk telephony
Telecom operators have been using satellite communications for many years to carry longdistance telephone communications, especially intercontinental, to complement or to bypass
submarine cables. To the end-user this is transparent: the phone calls are routed automatically
via the available capacity at any given moment. However, the 74,000 km round trip, even at
the speed of satellite signals, takes 250 milliseconds causing a delay that makes telephone
conversation rather unnatural, hence the preference for telephony over cable.
In regions where it is not so easy to install terrestrial telephone connections because of the
low density of population or because of the nature of the terrain, satellite is still being used to
connect the local switchboard to the telephone network. This technology is called thin route
or trunk satellite telephone networking. Wireless (microwave, two-way radio, etc) and optical
links, however, are replacing satellite increasingly in this area. With the advent of true mobile
telephony (cellular systems such as GSM, the Global System for Mobile communication),
and new end-user connection technologies such as Fixed Wireless Local Loop where there is
no longer a need to wire up each subscriber, satellite thin route telephony is becoming less
and less popular. In the future, satellite thin route telephony is expected to only hold a small
share of trunk telephony in areas that are otherwise impossible to reach.
Mobile satellite telephony
Mobile telephony allows the user to make telephone calls and to transmit and receive data
from wherever he/she is located. Digital cellular mobile telephony such as GSM has become
a worldwide standard for mobile communications, but its services lack coverage over areas
that are sparsely populated or uninhabited (mountains, jungle, sea), because it is not
economically viable or practical for the network operators to build antennas there. Satellite
telephony seems to be able to provide a possible solution to the problem of providing voice
and data communications services to these other locations.
Inmarsat
Inmarsat was the world's first global mobile satellite communications operator, founded in
the late 1970s. It focuses on communications services to maritime, land-mobile, aeronautical
and other users. Inmarsat now supports links for phone, fax and data communications at up to
64 Kbps to more than 210,000 ship, vehicle, aircraft and portable terminals.
The range of Inmarsat systems includes mobile terminals from handhelds to consoles, with
easy set-up mechanisms that allow users wherever they are to connect via a global fleet of
geostationary Inmarsat satellites to the terrestrial communications network and to carry out
telephone conversations, data transfers, and increasingly multimedia applications and Internet
access. Inmarsat is aimed at professionals who need a reliable communications system
wherever they are: ship owners and managers, journalists and broadcasters, health and
disaster-relief workers, land transport fleet operators, airlines, airline passengers and air
traffic controllers, government workers, national emergency and civil defence agencies, and
peacekeeping forces. The cost is rather high while the capacity is still rather limited:
voice/fax/data systems achieve a maximum data rate of 64 Kbps at connection costs starting
at almost US$ 3 per minute. Dedicated mobile IP systems can achieve a maximum download
speed of up to 144 Kbps.
LEO-based telephony
Another mobile satellite communications system is the Globalstar satellite telephone
network. Globalstar, which was established in 1991 and began commercial service in late
1999, offers service from virtually anywhere across over 100 countries, as well as from most
territorial waters and several mid-ocean regions. Globalstar deploys handheld telephone sets
that switch between the terrestrial wireless telephone network (GSM) and a LEO-based
satellite network in places where no terrestrial GSM network is available.
Globalstar telephony coverage map (April 2003)
Signals from a Globalstar phone or modem are received by one of the 48 LEO satellites and
relayed to ground-based gateways, which then pass the call on to the terrestrial telephone
network.
Globalstar LEO Satellite Telephone Service
A similar LEO satellite communications system is Iridium. Both Iridium and Globalstar are
based on constellations of satellites that can communicate with small handheld telephone sets
as well as between themselves, effectively acting as switchboards in the sky. The satellites
orbit at approximately 800 km above Earth and provide worldwide mobile telephony and
Internet access. Because of the short delay times (thanks to the low height and thus short
distance between Earth station and satellite) it is theoretically possible to introduce
videoconferencing and interactive multimedia to both fixed (with outdoor antenna) and
mobile transceivers at a later stage. It is easy to understand how LEO services would be
suited for urban or rural areas that are not connected to a broadband terrestrial infrastructure
or that cannot be covered economically using traditional terrestrial infrastructures.
GEO-based telephony
An alternative approach to satellite telephony uses a geostationary satellite instead of the
LEO. This results in longer delays (approximately half a second) but switching on board the
satellite reduces this inconvenience as much as possible. The Thuraya mobile satellite system
was launched in 1991, its satellite maintains a geo-synchronous orbit at 44 degrees East.
Thuraya operates effectively in both satellite and GSM environments. Its satellite network
capacity is about 13,750 telephone channels. When within reach of a GSM network,
Thuraya's mobile phone acts as an ordinary GSM handset. Outside this GSM coverage it
seamlessly switches to become a satellite telephone. The system can be used for voice, data,
fax, SMS and location determination (GPS-like). Thuraya handsets and subscription services
are distributed through service providers (mobile telecom companies) located in 106
countries in Europe, Africa, the Middle East, Asia and India. Through roaming agreements,
customers can use their handsets in a number of other countries as well.
Satellite Telephones
Thuraya GSM/Satellite Telephone Service
Satellite based mobile telephony: Conclusion
The deployment of these LEO-based services has not been as successful as had been hoped
by the providers. While initially it took a long time before the first service became available,
the competitors, in this case cellular mobile telephony, eg GSM, had already won a market
share that was lost for the new technology. The receivers were initially too expensive (about
US$ 2,000), the communications costs too high (from less than US$ 1 to more than US$ 5
per minute depending on the call destination and the payment programme, Thuraya being
cheaper in general) and the service had a reputation of not being very reliable: the technology
did not seem to be sufficiently mature and calls were frequently interrupted. The transmission
speed was very low (maximally 9.8 Kbps which is comparable to GSM-based transmissions).
Data, broadband and multimedia services
When we consider that TV and radio, telephone and fax nowadays are all being digitised and
packaged in datagrams (small data packets) to be transported on any type of network, it is
easy to understand that any digital content can be distributed in much the same way. This is
obviously the case with data over satellite communications networks.
Normally, data does not suffer from the small delay caused by the long transmission path via
satellite. Telecoms and global telecommunications carriers have been using satellite data
links to complement existing wire-based data networks for many years. Large, multinational
companies or international organisations in particular have exploited the ability to transfer
data over satellite networks since the time when smaller and cheaper satellite terminals and
more flexible satellite network services became available. Satellite services could support the
different services they were interested in, such as data collection and broadcasting, image and
video transfer, voice, two-way computer transactions and database inquiries.
The development of common datagram and data transport standards, and the digitisation of
voice, image, video and multimedia in general, have led to a shift to Internet Protocol (IP)based satellite communication systems that integrate seamlessly into the Internet world. It is
useful at this point to make a distinction between three different types of applications.
IP over satellite for ISPs
Telecoms and connectivity providers have started using satellite communications to bypass
the increasingly clogged terrestrial and submarine networks to complement their backbone
connectivity or to supplement them where they are not yet available. This approach takes
advantage of the fact that satellite is not a real point-to-point connection like cable, but a
connection that allows the delivery to multiple points at the same time. This allows for
simultaneous updating of multiple caching, proxy or mirroring servers.
IP via satellite for ISPs
In much the same way, it is possible to push Internet content to and even over the edges of
existing networks. When it is necessary to provide large amounts of content to places that are
poorly connected to the Internet, it is now possible to push content to local PoPs (Point of
Presence) edge servers. These can then in turn serve as ISPs to the local users or user
communities.
Although cable may be the preferred way to connect areas with a concentrated demand for
access (like cities or densely populated areas), satellite communications can still assist local
ISPs especially when there is not yet a reliable wired connection to the Network Access
Points or the Metropolitan Access Exchange points on the backbone. This is also practical
when a large quantity of content needs to be transported between two particular points and
high-capacity cable connections are not available.
Corporate or institutional VSAT networks
A particular application of data via satellite is VSAT (Very Small Aperture Terminal)
networks. Organisations with many remote affiliates can create a private high-speed satellite
intranet, which links the main office reliably with all local sites. Within and amongst
institutions there is an ever-growing need to communicate and to enhance the existing
networks, both human and physical. These networks, comparable to the corporate or
institutional networks of large multinational companies or international institutions, today
need high speed, reliable and cost-effective communications. This is especially true when the
locations are dispersed over remote regions and multiple countries, and barely connectable
via a terrestrial network infrastructure. In this case, satellite communications are an effective
way to provide private or secure data networks. VSAT can provide a complete network
capable of connecting all sites and connecting to the Internet, wherever the facilities are
located or wherever facilities will be located in the foreseeable future, including the homes of
staff, members, students etc.
VSAT stands for Very Small Aperture Terminal and refers to combined send/receive
terminals with a typical antenna diameter of 1 to 3.7 m linking the central hub to all remote
offices and facilities and keeping them all in constant immediate contact. VSAT networks
offer solutions for large networks with low or medium traffic. They provide very efficient
point-to-multipoint communication, are easy to install and can be expanded at low extra cost.
VSAT networks offer immediate accessibility and continuous high-quality transmissions.
They are adapted for any kind of transmission, from data to voice, fax and video.
The great advantage of VSAT is its flexibility. It permits any kind and size of network based
around a central hub and remote locations. This makes them particularly useful for corporate
networks or, for example, communication between educational, government or health-care
institutions. Through a VSAT network, a corporation can communicate freely and constantly
with branch offices:
Voice and fax transmissions
Local Area Network interconnection
Data broadcasting
Videoconferencing
In-house training
Various network topologies, protocols and interfaces are available to implement VSAT
communications applications. It is possible to lease satellite capacity on a carrier-per-carrier
basis for any type of VSAT network. VSAT operators offer turnkey solutions including
installation, licensing and maintenance.
VSAT networks are generally 'star' networks. This means there is a central location that acts
as a hub through which remote locations can transmit and receive data to and from each
other. They can be one- or two-directional.
VSAT Star-shaped Networks
A mesh configuration enables remote terminals to contact each other without passing through
the hub and is particularly appropriate for large corporations where local facilities need to be
in
contact
with
other
regions.
VSAT Mesh-shaped Networks
New VSAT technologies and services are being offered to support these demands.
Employing one- or two-way satellite communication, IP-compatible solutions enable private
network operators to provide their network members with enhanced speed and reliability for
institution-wide communication. Networks featuring PC-based user terminals equipped with
data cards linked to a receive/transmit satellite dish ensure fast Internet access and fast,
simultaneous data broadcast to all user terminals via satellite. Intranets, Extranets, Internet
access and email messaging are becoming just as important as the traditional video, voice and
data requirements of videoconferencing, business TV and data-file exchange. Different levels
of VSAT services can deliver various options depending on the requirements of each
network.
Capacity can be booked on a full-time basis with prior reservation for minimum utilisation of
any 24-hour contiguous period per occasion or on an ad-hoc basis according to a pre-arranged
plan of identical transmissions during specific periods, or for occasional use.
PAMA (Permanently Assigned Multiple Access) means having a permanently assigned
frequency channel that provides dedicated bandwidth, through which the network can send
data, voice or video. This may be required when larger amounts of data continuously need to
be transmitted between each element of the network. This can be the case in mission-critical
real-time processes such as process monitoring, distributed processes and data collection, but
also
in
media
streaming
(as
in
TV
and
radio
broadcasting).
DAMA (Demand Assignment Multiple Access) provides intermittent communication or
managed VSAT services on a pay-per-usage basis. With DAMA, satellite capacity is
instantaneously assigned and adapted according to immediate traffic needs. It is available
when needed, and users only pay for what they use. DAMA can support changing or
intermittent image-based or heavy data transfer needs and is best suited where multiple
services are integrated into a single network, since it supports telephony, low- and high-speed
data, video and multimedia applications. In order to be cost effective, DAMA requires the
network to be designed quite precisely to meet the organisation's needs for data distribution
and communications. Peak and minimal usage levels need to be particularly estimated.
DAMA is a highly efficient means of instantaneously assigning telephony and data channels
in
a
transponder
according
to
immediate
traffic
demands.
Advantages of VSAT networks include:
Wide geographic coverage
Independence from terrestrial communication infrastructure
High availability
Communication costs independent of transmission distance
Flexible network configuration
Rapid network deployment
Centralised control and monitoring
Any service can be provided from telephony through to ATM, Frame Relay, and of
course, high speed broadband Internet
Disadvantages include:
VSAT services are generally expensive
VSAT services are not available for single site users, but only to multiple site
networks
The ODU (outdoor unit, antenna) may be prone to vandalism or adverse weather
conditions (lightning, storm, etc)
Requires professional installation, management, monitoring and maintenance
In some countries VSATs are heavily regulated
As with all satellite solutions, there is a latency (delay) in the signal, making
telephone and videoconferencing services more difficult
End-user services for home or small office
Broadband access for end-users is usually considered a 'wired' solution: fibre optic
backbones, cable modems on coax, xDSL and ISDN on twisted copper. ADSL can only be
provided up to a distance of between 4 and 6 km from the local telephone exchanges,
depending on various factors. The cost to upgrade the existing copper network is very high.
This means that many households, particularly those in rural and remote areas, will probably
never be able to receive ADSL. Similarly, the cost of laying bi-directional cables for
interactive TV and Internet means that cable distribution is also unlikely to be available to
those living in small towns or in the countryside.
Satellite broadband connectivity has never been considered seriously, as long as it did not
allow for interactivity. However, nowadays satellites can provide interactivity via either the
satellite return channel or by using a hybrid solution with narrow-band return path via a
telephone line. With Internet via satellite, every user with the correct equipment and living
within the satellite footprint can now have a broadband connection.
Satellite has the capability to reach everywhere, thus effectively removing local loop
difficulties, especially in areas with poorly developed infrastructure. The subscriber requests
(eg the click on a hyperlink in a web page) can still be routed through terrestrial telephone
lines, but the downloaded data can now be routed via satellite directly to the Earth station of
the end-user. The typical asymmetry of home and small business Internet use opens up the
possibility of using a slow, small pipe in one direction and a fast, wide pipe in the other. The
average user does not need in-bound high-bandwidth connectivity around the clock and needs
even less out-bound high-bandwidth. So the hybrid of high-speed satellite for in-bound
matched to a low-cost, low-speed request path may well be the most cost-effective solution.
Using phone lines and a satellite downlink path means that you don't pay for more technology
than you need.
Most Internet-type traffic is asymmetric by nature: on average, the downlink (from ISP to
end-user) is 20 times greater than the uplink (from end-user towards the Internet). It is worth
noting however, that this is not true for certain particular end-users, web builders, content
distributors etc, where the ratio is, of course, different.
One-way Satellite Internet Connection
Recent developments have made it possible to send all requests and return data through the
satellite, which is ideal for areas with a weak telephone infrastructure.
Two-way Satellite Internet Connection
Current configurations can deliver data at a rate of up to 40 Mbps, but in practice, this means
that the hub communicates with the end-user terminals at speeds of up to 4 Mbps. The
terminals have a return link to the hub depending on the set-up of the network via a telephone
modem connection or via the satellite return system with speeds ranging from 16 Kbps to 512
Kbps. The hub is continually listening for data requests from the terminals so, to the user, the
system appears to be 'always on'. It is understood that in order to use public Internet access
through the satellite effectively, the hub needs to be well connected to the Internet backbone.
Not only does this system provide a broadband connection via the satellite downlink, but it
also means that the inherent advantages of satellites in many applications can be exploited,
especially the ability to multicast or broadcast the same data to millions of users over a huge
area. By applying intelligent caching techniques and news group feeds, traffic in the networks
can be reduced and the relatively high bandwidth cost of the space link becomes insignificant
especially
when
compared
to
the
reach.
This allows not only for 'pull' services, such as high-speed web browsing, where a single user
requests a specific item, but more importantly also for 'push' or multicast services, where a
file or stream is transmitted to many users at the same time, for example, real-time
information or streamed video. This type of push service is much harder to accomplish with
traditional wire-based terrestrial Internet connection technologies because it consumes
valuable
bandwidth
on
all
branches
of
the
network.
With satellite transmission, the number of potential users that can receive and decode
broadcast data can be restricted to one user, a group of users or all users. With a point-topoint data transfer, TCP/IP is used to send data addressed to one particular user. With pointto-multipoint file download or video stream, UDP or IP multicasting is used, and the satellite
broadcasts data that can be decoded by a specific group of users.
Only authorised users can connect the base station through the Internet and operate in
interactive mode (eg initiate an online web session). Conditional access is implemented at the
DVB transport level by conventional means, using 'smart cards' or a similar technology. In
addition, every user station has a unique station identification (hardware address) that is used
at link level for individual addressing of stations. Service operators are able to set access
authorisations on a user level so that transmissions may be restricted to a closed user group,
for example, for security reasons, or to allow subscription-based services. The PC board can
also be used for receiving video and audio broadcasting services already available and
transmitted
by
digital
broadcasters
on
the
same
satellite.
Satellite Internet connectivity offers possibilities that are becoming commonly accepted in
many different end-user communities, in regions that are excluded from access until such
time as wired or wireless broadband become available. Because it is impossible to predict
when such services will become available, it may be better to opt for satellite
telecommunications technology as it is immediately available. While financially it may seem
to make sense to wait for the best solution that will become available in the future, in the
short to medium term this could mean a high cost of lost (educational) opportunity.
Educational institutions can communicate across countries, regions and cultures, share
libraries and databases of research information, or offer distance-learning services that are
based on the TCP/IP protocols. Medical institutions can develop networks for telemedicine
applications. Government entities can deliver citizen information services. Push services
enable in all instances the multicasting of video and audio streams, database downloading and
software update distribution. Access to Internet and multimedia becomes available to remote
communities,
effectively
fighting
exclusion.
To conclude, the advantages of two-way satellite Internet connectivity for end-users include:
Reception is possible with a small antenna (one already in use to receive TV can, in
many cases, be sufficient but may require adaptation)
Connection is possible almost anywhere instantly within the footprint of the satellite,
with no cabling work or delays dependent on terrestrial infrastructure, thus effectively
solving the typical 'last mile' problem
Consumer equipment is relatively low cost
Internet connectivity can be combined with traditional broadcast technologies such as
digital TV and radio, enabling content providers to select the most appropriate
delivery means for particular content
In addition, multimedia push services via satellite, such as data broadcasting or
information streaming, are extremely efficient. In these cases, there is no need for a
return link via modem, so there is no additional cost for connectivity to the Internet
Some of the main disadvantages include:
Satellite Internet is generally more expensive than terrestrial access solutions, at least
in regions where they are available
The outdoor unit (antenna and cabling) are more prone to vandalism and weather
conditions
Bandwidth availibility is somewhat limited
Requires professional support
Not the ideal technology for videoconferencing
Mobile data communications
We talk about fixed or mobile services depending on the specific application. Fixed services
are aimed at Earth stations that stay in the same place while operating. The antenna does not
move during transmission and reception. Mobile services in contrast are aimed at users that
need to receive or transmit while moving.
Euteltracs and some Inmarsat applications are examples of mobile satellite data
communication services. Euteltracs equips cars, trucks, ships etc, with a small antenna, an onboard terminal with keyboard and LCD, plus software linking the on-board information
system via the Euteltracs Network Management Centre based in France with the vehicle's
home base. This set-up enables low data-rate services between the mobile vehicle's home
base and the vehicle itself while on the move, which allows for:
Vehicle or vessel localisation with an accuracy of 100 m
Transmission of alarm and distress messages
Message exchange between the mobile terminal and base
Data collection and transmission from the vehicle or vessel
Access to external databases for example, for weather or traffic conditions
This type of system is extremely rugged but allows only for very limited amounts of data to
be transferred. It is therefore not an evident choice for multimedia applications.
5. Explain in detail about the cellular concept?
The cells are normally drawn as hexagonal, but in practice they are irregularly shaped, this is
as a result of the influence of the surrounding terrain, or of design by the network planners
 Advantages of cell structures:
 higher capacity, higher number of users
 less transmission power needed
 more robust, decentralized
 base station deals with interference, transmission area etc. locally
 Problems:
 fixed network needed for the base stations
 handover (changing from one cell to another) necessary
 interference with other cells
•
•
The number of cells in any geographic area is determined by
number of MS subscribers who will be operating in that area,
geographic layout of the area (hills, lakes, buildings).
Large Cells
The maximum cell size for GSM is approximately 70 km in diameter, Size is
dependent on
the terrain the cell is covering
the power class of the MS.
Generally large cells are employed in:
Remote areas, Coastal regions, Areas with few subscribers.
Large areas which need to be covered with the minimum number of cell sites.
Small Cells
support a large number of MSs, in a small geographic region,
low transmission power may be required to reduce the effects of interference.
Small cells currently cover 200 m and upwards.
Uses:Urban areas,Low transmission power required,High number of MSs
The Trade Off – Large vs Small
Network providers would like to use large cells to reduce installation and maintenance
cost, but realize that to provide a quality service to their customers, they have to
consider many factors, such as terrain, transmission power required, number of MSs
etc. This inevitably leads to a mixture of both large and small cells.
Approaches to Cope with Increasing Capacity
Adding new channels
Frequency borrowing – frequencies are taken from adjacent cells by congested cells
Cell splitting – cells in areas of high usage can be split into smaller cells
Cell sectoring – cells are divided into a number of wedge-shaped sectors, each with
their own set of channels
 Microcells – antennas move to buildings, hills, and lamp posts




•
•
•
•
•
•
•
The cells are omni-directional cells. That is each site has a single cell and that cell has
a single transmit antenna which radiates the radio waves to 360 degrees.
The problem of omni-directional cells:
As the number of MSs increases in the same geographical region, need to increase
the number of cells to meet the demand.
( Achieved by decrease the size of the cell and fit more cells into this geographical
area).
introduces co-channel and adjacent channel interference, both of which degrade the
cellular network‘s performance.
splits a single site into a number of cells
each cell behaves as an independent cell to gain a further increase in capacity within
the geographic area (since independent transmit and receive antennas are used)
Each cell uses special directional antennas to ensure that the radio propagation from
one cell is concentrated in a particular direction.
Advantages
all the energy from the cell in a smaller area 60, 120, 180 degrees instead of 360
degrees, we get a much stronger signal, which is beneficial in locations such as ―inbuilding coverage‖.
we can now use the same frequencies in a much closer re-use pattern, thus allowing
more cells in our geographic region which allows us to support more MSs.
 Adjacent cells assigned different frequencies to avoid interference or crosstalk
 Objective is to reuse frequency in nearby cells
– 10 to 50 frequencies assigned to each cell
– transmission power controlled to limit power at that frequency escaping to
adjacent cells
– the issue is to determine how many cells must intervene between two cells
using the same frequency
6.
Explain in detail about the concept of Frequency reuse?
 each cell allocated a group k channels
– a cluster has N cells with unique and disjoint channel
 groups, N typically 4, 7, 12
 total number of duplex channels S = kN
 Cluster repeated M times in a system
 Total number of channels that can be used (capacity)
– C = MkN = MS
 Smaller cells  higher M  higher C
+ Channel reuse  higher capacity
+ Lower power requirements for mobiles
– Additional base stations required
– More frequent handoffs
– Greater chance of ‗hot spots‘
 channels unique in same cluster, repeated over clusters
 keep cell size same
– large N : weaker interference, but lower capacity
– small N: higher capacity, more interference need to maintain certain S/I level
 frequency reuse factor: 1/N
– each cell within a cluster assigned 1/N of the total available channels
 In most of the current networks, frequency reuse factor is 1.
Design of cluster size
 In order to connect without gaps between adjacent cells (to Tessellate)
 N = i2 + ij + j2 where i and j are non-negative integers
 Example i = 2, j = 1
– N = 22 + 2(1) + 12 = 4 + 2 + 1 = 7
 Next page example
– move i cells along any chain or hexagon.
– then turn 60 degrees counterclockwise and move j cells.
 Example 3.1 in page 61
 Fixed Channel Assignments
– Each cell is allocated a predetermined set of voice channels.
– If all the channels in that cell are occupied, the call is blocked, and the
subscriber does not receive service.
– Variation includes a borrowing strategy: a cell is allowed to borrow channels
from a neighboring cell if all its own channels are occupied.
– This is supervised by the Mobile Switch Center: Connects cells to wide area
network; Manages call setup; Handles mobility
 Fixed Channel Assignments
– Each cell is allocated a predetermined set of voice channels.
– If all the channels in that cell are occupied, the call is blocked, and the
subscriber does not receive service.
– Variation includes a borrowing strategy: a cell is allowed to borrow channels
from a neighboring cell if all its own channels are occupied.
– This is supervised by the Mobile Switch Center: Connects cells to wide area
network; Manages call setup; Handles mobility
 Dynamic Channel Assignments
– Voice channels are not allocated to different cells permanently.
– Each time a call request is made, the serving base station requests a channel
from the MSC.
– The switch then allocates a channel to the requested call based on a decision
algorithm taking into account different factors: frequency re-use of candidate
channel and cost factors.
– Dynamic channel assignment is more complex (real time), but reduces
likelihood of blocking
 a call is more annoying than line busy
 Guard channel concept
– Reserve some channels for handoffs
– Waste of bandwidth
– But can be dynamically predicted
 Queuing of handoff requests
– There is a gap between time for handoff and time to drop.
– Better tradeoff between dropping call probability and network traffic.
 Reduce the burden for handoff
– Cell dragging
– Umbrella cell
 major limiting factor in performance of cellular radio systems
 sources of interference:
– other mobiles in same cell
– a call in progress in a neighboring cell
– other base stations operating in the same frequency band
– Non-cellular system leaking energy into the cellular frequency band
 effect of interference:
– voice channel: cross talk
– control channel: missed or blocked calls
 two main types:
– co-channel interference
– adjacent channel interference
 cells that use the same set of frequencies are called co-channel cells.
 Interference between the cells is called co-channel interference.
 Co-channel reuse ratio: Q = D/R=sqrt(3N)
– R: radius of cell
– D: distance between nearest co-channel cells
 Small Q  small cluster size N  large capacity
 large Q  good transmission quality
 tradeoff must be made in actual cellular design
 SINR
 Power: propagation factor 2-6
– Sun, nuclear bomb
 Approximation
 AMPS example
–
=4, S/I=18dB, N needs to be larger than 6.49.
– Reuse factor 1/N small
 Relations: cochannel interference, link quality, reuse factor
 Example 3.2

UNIT III PART-A

Transcription

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