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Chapter 1. INTRODUCTION
A computer network is an interconnected collection of independent computers which aids communication in numerous ways. Apart from providing a good communication medium, sharing of available resources (programs and data on a computer are available to anyone on the network, regardless of the physical location of the computer and user), improved reliability of service (because of the presence of multiple computers), and cost-effectiveness (as small computers have a better price/performance ratio than larger ones) are some of the advantages of computer networking. In the early days, till even the early 1970s, computing and communication were considered to be two separate independent fields. The late 1970s and the early 1980s saw the merger of the fields of computing and communication. A modern-day computer network consists of two major components, namely, distributed applications and networking infrastructure. The distributed applications provide services to users/applications located on other computers. Common examples of distributed applications are Internet, electronic mail, air/train ticket reservation systems, and credit card authentication systems. The networking infrastructure provides support for data transfer among the interconnected computers where the applications reside. There are no restrictions on the physical media used for providing connectivity among the computers. Several physical media such as copper cable, optic fiber, and wireless radio waves are used in the present-day communication infrastructure. The main focus of this book is wireless networking in general, and ad hoc wireless networks in particular. The first half of this chapter consists of a broad discussion of the various concepts and principles involved in wireless communication. The second half of the chapter provides a brief introduction to computer networks, network reference models, and wireless networks.
As mentioned earlier, this book focuses on wireless networks, where electromagnetic radio waves are used for communication (exchange of information). In the following sections, the various characteristics of radio propagation are first discussed. Some of the important signal modulation mechanisms, multiple access techniques, and error control mechanisms are then described. A familiarity with all these fundamental aspects of wireless transmission is essential for
networks.
1.2 THE ELECTROMAGNETIC SPECTRUM
Wireless communication is based on the principle of broadcast and reception of electromagnetic waves. These waves can be characterized by their frequency (f) or their wavelength (λ). Frequency is the number of cycles (oscillations) per second of the wave and is measured in Hertz (Hz), in honor of Heinrich Hertz, the German physicist who discovered radio, and wavelength is the distance between two consecutive maxima or minima in the wave. The speed of propagation of these waves (c) varies from medium to medium, except in a vacuum where all electromagnetic waves travel at the same speed, the speed of light. The relation between the above parameters can be given as
(1.2.1)
Table 1.1 shows the various frequency bands in the electromagnetic spectrum as defined by the International Telecommunications Union (ITU). ITU, located in Geneva and a sub-organization of the United Nations, coordinates wired and wireless telecommunication activities worldwide. There are no official names for the bands in which the very high-frequency X-rays and Gamma rays fall. A pictographic view of the electromagnetic spectrum is given in Figure 1.1 . Table 1.1. Frequency bands and their common uses " id="pdf-obj-1-34" src="pdf-obj-1-34.jpg">
where c is the speed of light (3 × 10 8 m/s), f is the frequency of the wave in Hz, and λ is its wavelength in meters. Table 1.1 shows the various frequency bands in the electromagnetic spectrum as defined by the International Telecommunications Union (ITU). ITU, located in Geneva and a sub-organization of the United Nations, coordinates wired and wireless telecommunication activities worldwide. There are no official names for the bands in which the very high-frequency X-rays and Gamma rays fall. A pictographic view of the electromagnetic spectrum is given in Figure 1.1.
Table 1.1. Frequency bands and their common uses
Figure 1.1. The electromagnetic spectrum.
The low-frequency bands comprised of the radio, microwave, infrared, and visible light portions of the spectrum can be used for information transmission by modulating the amplitude, frequency, or the phase of the waves (the different modulation techniques will be described later in this chapter). The high-frequency waves such as X-rays and Gamma rays, though theoretically better for information propagation, are not used due to practical concerns such as the difficulty to generate and modulate these waves, and the harm they could cause to living things. Also, such high-frequency waves do not propagate well through buildings. As we will see later, the amount of information that can be carried by an electromagnetic wave is
determined by the width of the wavelength band (the corresponding frequency band) and the encoding technique (number of bits encoded/Hz) used.
Radio waves are easy to generate and are widely used for both indoor and outdoor communication due to properties such as their ability to pass through buildings and ability to travel long distances. Since radio transmission is omnidirectional (when radio waves are generated, they spread out from the transmitting antenna in all directions) in nature, the need to physically align the transmitter and receiver also does not arise. The frequency of the radio wave determines many of the characteristics of the transmission. At low frequencies the waves can pass through obstacles easily, but their power falls with an inverse-squared relation with respect to the distance. The higher frequency waves are more prone to absorption by rain drops, and they get reflected by obstacles. Due to the long transmission range of the radio waves, interference between transmissions is a problem that needs to be addressed.
In the VLF, LF, and MF bands the propagation of waves, also called as ground waves, follows the curvature of the Earth. The maximum transmission ranges of these waves are of the order of a few hundred kilometers. They are used for low bandwidth transmissions such as amplitude modulated (AM) radio broadcasting (amplitude modulation will be described in detail later in this chapter). TheHF and VHF band transmissions are absorbed by the atmosphere near the Earth's surface. However, a portion of the radiation, called the sky wave, radiates outward and upward to the ionosphere in the upper atmosphere. The ionosphere contains ionized particles formed due to the sun's radiation. These ionized particles reflect the sky waves back to the Earth. A powerful sky wave may get reflected several times between the Earth and the ionosphere. Sky waves are used by amateur ham radio operators and for military communication.
Microwave transmissions (in the SHF band) tend to travel in straight lines and hence can be narrowly focused. Microwaves were widely used for long-distance telephony, before they got replaced by fiber optics. They are also widely used for mobile phones and television transmission. Since the energy is concentrated into a small beam, these transmissions have a higher signal-to-noise ratio (SNR). SNR is the ratio of the signal power to the noise power on a transmission medium, and is used to categorize the quality of a transmission. However, because of the higher frequency of operation they do not pass through buildings. Hence, proper line-of-sight alignment between the transmitting and the receiving antennas is required.
Straight-line microwave transmission over long distances requires repeaters at regular distances as microwaves get attenuated by objects found in their path. The number of repeaters required is a function of the microwave transmission tower height.
Infrared waves and waves in the EHF band (also known as millimeter waves) are used for short-range communication. They are widely used in television, VCR, and stereo remote controls. They are relatively directional and inexpensive to build. They cannot travel through obstacles, which is a light-like (and not radio-like) property that is observed in very high-frequency transmissions.
The visible light part of the spectrum is just after the infrared portion. Unguided optical signaling using visible light provides very high bandwidth at a very low cost. Recent applications of light-wave transmissions involve the use of lasers to connect LANs on two buildings through roof-top antennas. But the main disadvantage here is that it is very difficult to focus a very narrow unidirectional laser beam, which limits the maximum distance between the transmitter and receiver. Also, such waves cannot penetrate through rain or thick fog.
1.2.1 Spectrum Allocation
Since the electromagnetic spectrum is a common resource which is open for access by anyone, several national and international agreements have been drawn regarding the
usage of the different frequency bands within the spectrum. The individual national
governments allocate spectrum for applications such as AM/FM radio broadcasting, television broadcasting, mobile telephony, military communication, and government usage. Worldwide, an agency of the International Telecommunications Union Radiocommunication (ITU-R) Bureau called World Administrative Radio Conference (WARC) tries to coordinate the spectrum allocation by the various national governments, so that communication devices that can work in multiple countries can be manufactured. However, the recommendations of ITU-R are not binding on any government. Regarding the national spectrum allocation, even when the government sets aside a portion of the spectrum for a particular application/use (e.g., cellular telephony), another issue crops up – the issue of which carrier (company) is to use which set of frequencies. Many methods have been tried out for this frequency
allocation among multiple competing carriers, some of which are described below.
The first and the oldest method, known as comparative bidding and often referred to as the beauty contest method, worked as follows. Each carrier would submit a proposal to the government explaining how it intends to use the frequency bands. The government, represented by officials, would go through the different proposals and come to a conclusion on which carrier would serve the public interest best, and accordingly allocate frequencies to that carrier. However, this method was often found to lead to bribery, corruption, and nepotism. Hence, this method was abandoned.
Governments then moved on to the lottery system, where a lottery would be held among the interested companies. But this method too had a serious problem: Companies which had no real interest in using the spectrum could participate in the lottery and win a segment of the spectrum. They would later resell the spectrum to genuine carriers and make huge profits. So this system also had to go.
The third method, which is still used today, is the auctioning method. The frequency bands would be auctioned off to the highest bidder. This method overcame all the problems associated with the previous two methods. Though governments were able to generate huge revenues by auctioning off frequencies, some companies that bought those frequencies at exorbitantly high rates through bidding are on the verge of going bankrupt.
The simplest method of allocating frequencies is not to allocate them at all. Internationally, the ITU has designated some frequency bands, called the ISM (industrial, scientific, medical) bands, for unlimited usage. These bands commonly used by wireless LANs and PANs are around the 2.4 GHz band. Parts of the 900 MHz and the 5 GHz bands are also available for unlicensed usage in countries such as the United States and Canada.
1.3 RADIO PROPAGATION MECHANISMS
Radio waves generally experience the following three propagation mechanisms:
• Reflection: When the propagating radio wave hits an object which is very large compared to its wavelength (such as the surface of the Earth, or tall buildings), the wave gets reflected by that object.
Reflection causes a phase shift of 180 degrees between the incident and the reflected rays. • Diffraction: This propagation effect is undergone by a wave when it hits an impenetrable object. The wave bends at the edges of the object, thereby propagating in different directions. This phenomenon is termed as diffraction. The dimensions of the object causing diffraction are comparable to the wavelength of the wave being diffracted. The bending causes the wave to reach places behind the object which generally cannot be reached by the line-of-sight transmission. The amount of diffraction is frequency-dependent, with the lower frequency waves diffracting more. • Scattering: When the wave travels through a medium, which contains many objects with dimensions small when compared to its wavelength, scattering occurs. The wave gets scattered into several weaker outgoing signals. In practice, objects such as street signs, lamp posts, and foliage cause scattering.
Figure 1.2 depicts the various propagation mechanisms that a radio wave encounters. When the transmitted wave is received at the receiver, the received power of the signal is generally lower than the power at which it was transmitted. The loss in signal strength, known as attenuation, is due to several factors which are discussed in the next section.
Figure 1.2. Propagation mechanisms.
The wireless channel (transmission medium) is susceptible to a variety of transmission impediments such as path loss, interference,
and blockage. These factors restrict the range, data rate, and the reliability of the wireless transmission. The extent to which these factors affect the transmission depends upon the environmental conditions and the mobility of the transmitter and receiver. Typically, the transmitted signal has a direct-path component between the transmitter and the receiver. Other components of the transmitted signal known as multipath components are reflected, diffracted, and scattered (explained in the previous section) by the environment, and arrive at the receiver shifted in amplitude, frequency, and phase with respect to the direct-path component. In what follows, the various characteristics of the wireless channel such as path loss, fading, interference, and Doppler shift are discussed. Also, two key constraints, Nyquist's and Shannon's theorems, that govern the ability to transmit information at different data rates, are presented.
1.4.1 Path Loss
Path loss can be expressed as the ratio of the power of the transmitted signal to the power of the same signal received by the receiver, on a given path. It is a function of the propagation distance. Estimation of path loss is very important for designing and deploying wireless communication networks. Path loss is dependent on a number of factors such as the radio frequency used and the nature of the terrain. Since several of these factors (in particular, the terrain) cannot be the same everywhere, a single model may not be enough, and hence cannot be used to describe the characteristics of every transmission. Therefore, in designing a network, several models are required to describe the variety of transmission environments.
The free space propagation model is the simplest path loss model in which there is a direct-path signal between the transmitter and the receiver, with no atmospheric attenuation or multipath components. In this model the relationship between the transmitted power P t and the received power P r is given by
respectively, in the direction from the receiver antenna transmitter to the receiver, d is the distance between the transmitter and receiver, and λ = c/f (from Equation 1.2.1 ) is the wavelength of the signal. Realistic path loss models that take into account the propagation " id="pdf-obj-7-20" src="pdf-obj-7-20.jpg">
where G t and G r are
gains, 1 respectively, in the direction from the
receiver antenna transmitter to the
receiver, d is the distance between the transmitter and receiver, and λ = c/f (fromEquation 1.2.1) is the wavelength of the signal. Realistic path loss models that take into account the propagation
effects in specific environments can be obtained by solving Maxwell's equations (for details on Maxwell's equations, the reader can refer to [1]). Since obtaining solutions for such models involves complex algorithms and computation intensive operations, simpler models have been proposed to depict the loss.
1 Antenna gain is defined as the ratio of the power required at the input of a loss-free reference antenna to the power supplied to the input of the given antenna to produce, in a given direction, signals of the same strength at the same distance.
Another popular path loss model is the two-ray model or the two- path model. The free space model described above assumes that there is only one single path from the transmitter to the receiver. But in reality, the signal reaches the receiver through multiple paths (because of reflection, refraction, and scattering). The two-path model tries to capture this phenomenon. The model assumes that the signal reaches the receiver through two paths, one a line-of-sight path, and the other the path through which the reflected (or refracted, or scattered) wave is received. According to the two-path model, the received power is given by
1 ]). Since obtaining solutions for such models involves complex algorithms and computation intensive operations, simpler models have been proposed to depict the loss. Antenna gain is defined as the ratio of the power required at the input of a loss-free reference antenna to the power supplied to the input of the given antenna to produce, in a given direction, signals of the same strength at the same distance. Another popular path loss model is the two-ray model or the two- path model. The free space model described above assumes that there is only one single path from the transmitter to the receiver. But in reality, the signal reaches the receiver through multiple paths (because of reflection, refraction, and scattering). The two-path model tries to capture this phenomenon. The model assumes that the signal reaches the receiver through two paths, one a line-of-sight path, and the other the path through which the reflected (or refracted, or scattered) wave is received. According to the two-path model, the received power is given by (1.4.2) where P is the transmitted power, G and G represent the antenna gains at the transmitter and the receiver, respectively, d is the distance between the transmitter and receiver, and h and h are the heights of the transmitter and the receiver, respectively. In the general case, for isotropic antennas (antennas in which the power of the transmitted signal is the same in all directions), the received power of the signal is given by (1.4.3) where γ is the propagation coefficient that varies between 2 (free- space propagation) and 5 (strong attenuation). For more details on path loss models, the reader can refer to [ 2 ]. 1.4.2 Fading " id="pdf-obj-8-18" src="pdf-obj-8-18.jpg">
where P t is the transmitted power, G t and G r represent the antenna gains at the transmitter and the receiver, respectively, d is the distance between the transmitter and receiver, and h t and h r are the heights of the transmitter and the receiver, respectively.
In the general case, for isotropic antennas (antennas in which the power of the transmitted signal is the same in all directions), the received power of the signal is given by
where γ is the propagation coefficient that varies between 2 (free- space propagation) and 5 (strong attenuation). For more details on path loss models, the reader can refer to [2].
1.4.2 Fading
Fading refers to the fluctuations in signal strength when received at the receiver. Fading can be classified into two types: fast fading/small-scale fading, and slow fading/large-scale fading.
Fast fading refers to the rapid fluctuations in the amplitude, phase, or multipath delays of the received signal, due to the interference between multiple versions (copies) of the same transmitted signal arriving at the receiver at slightly different times. The time between the reception of the first version of the signal and the last echoed signal is called delay spread. The multipath propagation of the transmitted signal, which causes fast fading, is because of the three propagation mechanisms described previously, namely, reflection, diffraction, and scattering. The multiple signal paths may sometimes add constructively or sometimes destructively at the receiver, causing a variation in the power level of the received signal. The received signal envelope of a fast-fading signal is said to follow a Rayleigh distribution (refer to [3]) if there is no line-of-sight path between the transmitter and the receiver (applicable in outdoor environments), and a Ricean distribution (refer to [3]) if one such path is available (characterizes indoor settings).
Slow fading occurs when objects that partially absorb the transmissions lie between the transmitter and receiver. Slow fading is so called because the duration of the fade may last for multiple seconds or minutes. Slow fading may occur when the receiver is inside a building and the radio wave must pass through the walls of a building, or when the receiver is temporarily shielded from the transmitter by a building. The obstructing objects cause a random variation in the received signal power. Slow fading may cause the received signal power to vary, though the distance between the transmitter and receiver remains the same. Slow fading is also referred to as shadow fading since the objects that cause the fade, which may be large buildings or other structures, block the direct transmission path from the transmitter to the receiver.
Some of the common measures used for countering the effects of fading are diversity and adaptive modulation. Diversity mechanisms are based on the fact that independent paths between the same transmitter and receiver nodes experience independent fading effects, and therefore, by providing multiple logical channels between the transmitter and receiver, and sending parts of the signal over each channel, the error effects due to fading can be compensated. The independent paths can be distinct in space, time, frequency, or polarization. Time diversity mechanisms aim at spreading the data over time so that the effects of burst errors are minimized. Frequency diversity mechanisms spread the transmission
over a wider frequency spectrum, or use multiple carriers for transmitting the information. The direct sequence spread spectrum and the frequency hopping spread spectrum techniques (which will be described later in this chapter) are some of the important frequency diversity mechanisms. Space diversity involves the use of different physical transmission paths. An antenna array could be used, where each antenna element receives an independent fading signal, each of the received signals following a different transmission path. The received signals are combined in some fashion such that the most likely transmitted signal could be reconstructed. In the adaptive modulation mechanisms, the channel characteristics are estimated at the receiver and the estimates are sent by the receiver to the transmitter through a feedback channel. The transmitter adapts its transmissions based on the received channel estimates in order to counter the errors that could occur due to the characteristics of the channel. Adaptive techniques are usually very complex to implement.
1.4.3 Interference
Wireless transmissions have to counter interference from a wide variety of sources. Two main forms of interference are adjacent channel interference and co-channel interference. In the adjacent channel interference case, signals in nearby frequencies have components outside their allocated ranges, and these components may interfere with on-going transmissions in the adjacent frequencies. It can be avoided by carefully introducing guard bands 2 between the allocated frequency ranges. Co-channel
interference,
sometimes also referred to as narrow-band
interference, is due to other nearby systems (say, AM/FM broadcast)
the same transmission frequency. As we will see later
in Chapter 3, narrow-band interference due to frequency reuse in cellular systems can be minimized with the use of multiuser
detection mechanisms, 3 directional antennas, and dynamic channel allocation methods.
small frequency
band used
to separate two
adjacent frequency bands in order to avoid interference between
3 Multiuser detection is an effective approach used to combat the multiuser interference problems inherent in CDMA systems.
Inter-symbol interference is another type of interference, where distortion in the received signal is caused by the temporal spreading and the consequent overlapping of individual pulses in the signal.
When this temporal spreading of individual pulses (delay spread) goes above a certain limit (symbol detection time), the receiver becomes unable to reliably distinguish between changes of state in the signal, that is, the bit pattern interpreted by the receiver is not the same as that sent by the sender. Adaptive equalization is a commonly used technique for combating inter-symbol interference. Adaptive equalization involves mechanisms for gathering the dispersed symbol energy into its original time interval. Complex digital processing algorithms are used in the equalization process. The main principle behind adaptive equalization is the estimation of the channel pulse response to periodically transmitted well-known bit patterns, known as training sequences. This would enable a receiver to determine the time dispersion of the channel and compensate accordingly.
1.4.4 Doppler Shift
The Doppler shift is defined as the change/shift in the frequency of the received signal when the transmitter and the receiver are mobile with respect to each other. If they are moving toward each other, then the frequency of the received signal will be higher than that of the transmitted signal, and if they are moving away from each other, the frequency of the signal at the receiver will be lower than that at the transmitter. The Doppler shift fd is given by
where v is
receiver, and λ is the wavelength of the signal.
1.4.5 Transmission Rate Constraints
Two important constraints that determine the maximum rate of data transmission on a channel are Nyquist's theorem and Shannon's theorem. The two theorems are presented below.
Nyquist's Theorem
The signaling speed of a transmitted signal denotes the number of times per second the signal changes its value/voltage. The number of changes per second is measured in terms of baud. The baud rate is not the same as the bit rate/data rate of the signal since each signal value may be used to convey multiple bits. For example, if the voltage values used for transmission are 0, 1, 2, and 3, then each
value can be used to convey two bits (00, 01, 10, and 11). Hence the bit rate here would be twice the baud rate. TheNyquist theorem gives the maximum data rate possible on a channel. If B is the bandwidth of the channel (in Hz) and L is the number of discrete signal levels/voltage values used, then the maximum channel capacity C according to the Nyquist theorem is given by
The above condition is valid only for a noiseless channel.
Shannon's Theorem
Noise level in the channel is represented by the SNR. It is the ratio of signal power (S) to noise power (N), specified in decibels, that is, SNR = 10 log 10 (S/N). One of the most important contributions of Shannon was his theorem on the maximum data rate possible on a noisy channel. According toShannon's theorem, the maximum data rate C is given by
where B is the bandwidth of the channel (in Hz).
1.5 MODULATION TECHNIQUES
Having seen the various radio propagation mechanisms and wireless transmission impairments, the next step is to see how raw bits constituting the information are actually transmitted on the wireless medium. Data (whether in analog or in digital format) has to be converted into electromagnetic waves for transmission over a wireless channel. The techniques that are used to perform this conversion are called modulation techniques. The modulation process alters certain properties of a radio wave, called a carrier wave, whose frequency is the same as the frequency of the wireless channel being used for the transmission. The modulation scheme to be used must be chosen carefully so that the limited radio spectrum is used efficiently. Modulation schemes can be classified under two major categories: analog modulation schemes and digital modulation schemes. This classification is based on the nature of the data, analog or digital, to be transmitted. Some of the commonly used modulation techniques are discussed below.
1.5.1 Analog Modulation
As the name implies, analog modulation techniques are used for transmitting analog data. The analog data signal is superimposed on a carrier signal. This superimposition is aimed at altering a certain property (amplitude or frequency) of the carrier signal. Some of the commonly used analog modulation techniques are amplitude modulation, frequency modulation, and phase modulation. These techniques are described below.
Amplitude Modulation
Amplitude modulation (AM) is one of the simplest modulation schemes. It was the first method to be used for transmitting voice. The transmitter superimposes the information signal x(t), also called the modulating signal, on the carrier signal c(t). The result of AM of the carrier signal of Figure 1.3 (b), by the information/modulating signal of Figure 1.3 (a), is shown in Figure 1.3 (c). It can be seen that the frequency of the modulated wave remains constant, while its amplitude varies with that of the information signal. As the power of a signal depends on its amplitude, the power of the transmitted wave depends on the power of the modulating signal. When a cosine carrier wave is used, the AM wave can be mathematically represented as below:
Figure 1.3 (b) , by the information/modulating signal of Figure 1.3 (a) , is shown in Figure 1.3 (c) . It can be seen that the frequency of the modulated wave remains constant, while its amplitude varies with that of the information signal. As the power of a signal depends on its amplitude, the power of the transmitted wave depends on the power of the modulating signal. When a cosine carrier wave is used, the AM wave can be mathematically represented as below: (1.5.1) Figure 1.3. Analog modulation schemes. " id="pdf-obj-13-26" src="pdf-obj-13-26.jpg">
Figure 1.3. Analog modulation schemes.
where n a ,
known
modulation index, is
amplitude of the information signal to that of the carrier signal, f c is the frequency of the carrier signal, x(t) is the information signal, and c(t) =cos(2π f c t) is the carrier signal.
In modern communication systems, AM is not
the preferred
modulation mechanism. AM creates additional unwanted signals in
frequencies on either side of the carrier during the superimposition process. These signals, called sidebands, lead to poor spectrum utilization, and they consume additional power. Hence, AM is considered to be an inefficient modulation mechanism. A variant ofAM, where the sidebands are stripped away on one side, called single side band (SSB), is often used in place of plain AM. Broadcast radio is one of the applications which still uses AM.
Angle Modulation
category.
modulated
mathematically
represented as
Equation 1.5.2 , with the condition that the derivative of the phase, that is, Φ'( t ), is proportional to the information signal x ( t ). Since frequency can also be defined as the rate of change of phase of the signal, here Φ' ( t ) represents the deviation of the instantaneous frequency of the modulated signal from that of the carrier signal. In FM, this Φ' ( t ) is directly proportional to the information signal/modulating signal. It can be represented as (1.5.3) where n is a constant, known as the frequency modulation index. The instantaneous frequency of the carrier wave is changed according to the amplitude of the information signal, resulting in the " id="pdf-obj-15-60" src="pdf-obj-15-60.jpg">
where A c is the amplitude and f c is the frequency of the carrier signal. Frequency modulation and phase modulation are special cases of angle modulation. Changing the frequency of a wave also changes its phase, and vice versa. The angle modulated signal has a constant amplitude, and as a consequence the transmitter operates at full power constantly, which also maximizes its range.
In frequency modulation (FM), the amplitude of the modulated signal is kept constant, while the instantaneous frequency is altered to reflect the information signal that is being transmitted. Frequency modulation follows Equation 1.5.2, with the condition that the
derivative of the
phase,
is,
Φ'(t),
information signal x(t). Since frequency can also be defined as the
of change of
phase of
signal, here Φ'
(t) represents the
deviation of the instantaneous frequency of the modulated signal from that of the carrier signal. In FM, this Φ' (t) is directly proportional to the information signal/modulating signal. It can be represented as
Equation 1.5.2 , with the condition that the derivative of the phase, that is, Φ'( t ), is proportional to the information signal x ( t ). Since frequency can also be defined as the rate of change of phase of the signal, here Φ' ( t ) represents the deviation of the instantaneous frequency of the modulated signal from that of the carrier signal. In FM, this Φ' ( t ) is directly proportional to the information signal/modulating signal. It can be represented as (1.5.3) where n is a constant, known as the frequency modulation index. The instantaneous frequency of the carrier wave is changed according to the amplitude of the information signal, resulting in the " id="pdf-obj-15-136" src="pdf-obj-15-136.jpg">
where n f is a constant, known as the frequency modulation index.
The instantaneous frequency of the carrier wave is changed according to the amplitude of the information signal, resulting in the
stretching or the compressing of the carrier wave depending on the value of the modulating voltage. FM is more resistant to noise. Random interference is more likely to affect the amplitude of a
signal rather than its frequency. Thus the SNR for an FM wave is
higher than that
for an AM wave. Some common applications
where FM is used are radio broadcasts and first-generation cellular phones. Figure 1.3 (d) shows the FM modulated signal.
Phase Modulation
In phase modulation (PM), the phase of the modulated signal Φ(t) is directly proportional to the information signal x(t). It is represented as
Figure 1.3 (d) shows the FM modulated signal. Phase Modulation In phase modulation (PM), the phase of the modulated signal Φ( t ) is directly proportional to the information signal x ( t ). It is represented as (1.5.4) where n is a constant, known as the phase modulation index. It can be seen from Equation 1.5.2 that [2π f t + Φ ( t )] is the phase of the modulated signal. The instantaneous phase deviation of the modulated signal from the carrier signal is Φ( t ). In PM, this Φ( t ) is proportional to the information signal/modulating signal. 1.5.2 Digital Modulation Digital modulation schemes are used for transmitting digital signals that consist of a sequence of 0 and 1 bits. As in analog modulation, digital modulation also changes a certain property of the carrier signal. The main difference between analog and digital modulation is that while the changes occur in a continuous manner in analog modulation, they occur at discrete time intervals in digital modulation. The number of such changes per second is known as the baud rate of the signal. Some of the basic digital modulation techniques such as amplitude shift keying, frequency shift keying, and phase shift keying are described below. Amplitude Shift Keying In amplitude shift keying (ASK), when a bit stream is transmitted, a binary 1 is represented by the presence of the carrier signal c ( t ) for a specified interval of time, and a binary 0 is represented by the absence of the carrier signal for the same interval of time. Mathematically, ASK can be represented as " id="pdf-obj-16-26" src="pdf-obj-16-26.jpg">
where n p is a constant, known as the phase modulation index. It can be seen from Equation 1.5.2 that [2πf c t + Φ(t)] is the phase of the modulated signal. The instantaneous phase deviation of the modulated signal from the carrier signal is Φ(t). In PM, this Φ(t) is proportional to the information signal/modulating signal.
1.5.2 Digital Modulation
Digital modulation schemes are used for transmitting digital signals that consist of a sequence of 0 and 1 bits. As in analog modulation, digital modulation also changes a certain property of the carrier signal. The main difference between analog and digital modulation is that while the changes occur in a continuous manner in analog modulation, they occur at discrete time intervals in digital modulation. The number of such changes per second is known as the baud rate of the signal. Some of the basic digital modulation techniques such as amplitude shift keying, frequency shift keying, and phase shift keying are described below.
Amplitude Shift Keying
In amplitude shift keying (ASK), when a bit stream is transmitted, a binary 1 is represented by the presence of the carrier signal c(t) for a specified interval of time, and a binary 0 is represented by the absence of the carrier signal for the same interval of time. Mathematically, ASK can be represented as
where A c is the amplitude of the carrier signal and f c is its frequency. The result of ASK when applied to the bit pattern shown in Figure 1.4 (a) is shown in Figure 1.4 (b).
Figure 1.4. Digital modulation schemes.
Figure 1.4 (a) is shown in Figure 1.4 (b) . Figure 1.4. Digital modulation schemes. Frequency Shift Keying " id="pdf-obj-17-16" src="pdf-obj-17-16.jpg">
Frequency Shift Keying
Frequency shift keying (FSK) is done as follows. If f c is the frequency of the carrier signal and if k is a small frequency offset, then transmission of a binary 1 is represented by the presence of a
carrier signal
frequency f c + k for
a specified
interval of time.
Transmission of
carrier signal of frequency f c - k for the same interval of time. Since two frequency levels are used in this technique, it is also known as
two-level FSK or
binary FSK (BFSK). FSK can
represented as follows:
Figure 1.4 (a) is shown in Figure 1.4 (c) . In the FSK mechanism described above, that is, BFSK, since only two frequency levels are used, each frequency level encodes just a single bit, 1 or 0. By using multiple frequency levels (by using multiple frequency offsets), a single frequency level can be used to encode multiple bits. An example is the four-level FSK. A four- level FSK can be used to transmit two bits per frequency shift. It can be mathematically represented as where A is the amplitude and f is the frequency of the carrier signal. The spacing between the frequencies used for representing bits 0 and 1 (also known as tone distance) is an important parameter in the design of FSK systems. The frequency spacing is directly proportional to the bandwidth occupied by the FSK signal. In order for a receiver to detect the 1 and 0 bits without any error, the tone distance must be chosen carefully. FSK modulation with a minimum tone distance of 1/2 T , where T is the duration of each transmitted bit, is referred to as minimum shift keying (MSK). In order to reduce side bands, the baseband signal to be transmitted is filtered before the frequency shift keying process. Gaussian filters are the most commonly used filters for this purpose, and the associated modulation mechanism is referred to as Gaussian MSK (GMSK). The filtering smoothens the rapid transmissions and reduces the bandwidth occupied by the signal. GMSK is widely used in second- " id="pdf-obj-18-68" src="pdf-obj-18-68.jpg">
be mathematically
where A c and f c are the amplitude and frequency, respectively, of the cosine carrier signal. FSK when applied to the bit pattern of Figure 1.4 (a) is shown in Figure 1.4 (c).
In the FSK mechanism described above, that is, BFSK, since only two frequency levels are used, each frequency level encodes just a single bit, 1 or 0. By using multiple frequency levels (by using multiple frequency offsets), a single frequency level can be used to encode multiple bits. An example is the four-level FSK. A four- level FSK can be used to transmit two bits per frequency shift. It can be mathematically represented as
where A c is the amplitude and f c is the frequency of the carrier signal.
The spacing between the frequencies used for representing bits 0 and 1 (also known as tone distance) is an important parameter in the design of FSK systems. The frequency spacing is directly proportional to the bandwidth occupied by the FSK signal. In order for a receiver to detect the 1 and 0 bits without any error, the tone distance must be chosen carefully. FSK modulation with a minimum tone distance of 1/2T, where T is the duration of each transmitted bit, is referred to as minimum shift keying (MSK). In order to reduce side bands, the baseband signal to be transmitted is filtered before the frequency shift keying process. Gaussian filters are the most commonly used filters for this purpose, and the associated modulation mechanism is referred to as Gaussian MSK (GMSK). The filtering smoothens the rapid transmissions and reduces the bandwidth occupied by the signal. GMSK is widely used in second-
generation cellular systems such as GSM, described in detail in Chapter 3.
Gaussian FSK (GFSK) is an FSK technique in which the data to be transmitted is first filtered in the baseband by means of a Gaussian filter, and is then modulated by simple frequency modulation. A two- level or four-level GFSK is used in the IEEE 802.11 standard (described in detail in the next chapter). An n-level GFSK technique is similar to an n-level FSK technique, the only difference being that the baseband signal is filtered using a Gaussian filter before being frequency modulated in the n-levelGFSK.
Phase Shift Keying
In phase shift keying (PSK), change in phase of the carrier signal is
used to represent
bits.
represented by the presence of the carrier for a specific interval of time, while the transmission of bit 1 is represented by the presence of a carrier signal with a phase difference of π radians for the same interval of time. PSK using a cosine carrier wave with amplitude A c and frequency f c can be mathematically represented as
Chapter 3 . Gaussian FSK (GFSK) is an FSK technique in which the data to be transmitted is first filtered in the baseband by means of a Gaussian filter, and is then modulated by simple frequency modulation. A two- level or four-level GFSK is used in the IEEE 802.11 standard (described in detail in the next chapter). An n-level GFSK technique is similar to an n-level FSK technique, the only difference being that the baseband signal is filtered using a Gaussian filter before being frequency modulated in the n-levelGFSK. Phase Shift Keying In phase shift keying (PSK), change in phase of the carrier signal is used to represent the 0 and 1 bits. The transmission of bit 0 is represented by the presence of the carrier for a specific interval of time, while the transmission of bit 1 is represented by the presence of a carrier signal with a phase difference of π radians for the same interval of time. PSK using a cosine carrier wave with amplitude A and frequency f can be mathematically represented as This technique is also known as binary PSK (BPSK) or two- level PSK since a single phase difference is used for representing 0 and 1 bits. Figure 1.4 (d) shows the BPSK modulation of the bit pattern o f Figure 1.4 (a) . Just as multiple frequency levels are used in FSK, multiple phase deviations can be used in PSK. This enables encoding of multiple bits by each phase representation. Quadrature PSK (QPSK), for example, uses four different phases each separated by π/ 2 radians. This would enable transmission of two bits per phase shift. The mathematical representation of QPSK is given below. π/ 4 shifted QPSK ( π/ 4-QPSK) is a QPSK mechanism where the maximum phase deviation is limited to ± 135 degrees. The main advantage of π/ 4 shifted QPSK is that it can be received non- coherently, that is, the receiver need not lock to the phase of the " id="pdf-obj-19-44" src="pdf-obj-19-44.jpg">
binary PSK (BPSK)
two-
level PSK since a single phase difference is used for representing 0
bits. Figure
(d) shows
the BPSK modulation of
pattern ofFigure 1.4 (a).
Just as multiple frequency levels are used in FSK, multiple phase deviations can be used in PSK. This enables encoding of multiple bits by each phase representation. Quadrature PSK (QPSK), for example, uses four different phases each separated by π/2 radians. This would enable transmission of two bits per phase shift. The mathematical representation of QPSK is given below.
Chapter 3 . Gaussian FSK (GFSK) is an FSK technique in which the data to be transmitted is first filtered in the baseband by means of a Gaussian filter, and is then modulated by simple frequency modulation. A two- level or four-level GFSK is used in the IEEE 802.11 standard (described in detail in the next chapter). An n-level GFSK technique is similar to an n-level FSK technique, the only difference being that the baseband signal is filtered using a Gaussian filter before being frequency modulated in the n-levelGFSK. Phase Shift Keying In phase shift keying (PSK), change in phase of the carrier signal is used to represent the 0 and 1 bits. The transmission of bit 0 is represented by the presence of the carrier for a specific interval of time, while the transmission of bit 1 is represented by the presence of a carrier signal with a phase difference of π radians for the same interval of time. PSK using a cosine carrier wave with amplitude A and frequency f can be mathematically represented as This technique is also known as binary PSK (BPSK) or two- level PSK since a single phase difference is used for representing 0 and 1 bits. Figure 1.4 (d) shows the BPSK modulation of the bit pattern o f Figure 1.4 (a) . Just as multiple frequency levels are used in FSK, multiple phase deviations can be used in PSK. This enables encoding of multiple bits by each phase representation. Quadrature PSK (QPSK), for example, uses four different phases each separated by π/ 2 radians. This would enable transmission of two bits per phase shift. The mathematical representation of QPSK is given below. π/ 4 shifted QPSK ( π/ 4-QPSK) is a QPSK mechanism where the maximum phase deviation is limited to ± 135 degrees. The main advantage of π/ 4 shifted QPSK is that it can be received non- coherently, that is, the receiver need not lock to the phase of the " id="pdf-obj-19-110" src="pdf-obj-19-110.jpg">
π/4 shifted QPSK (π/4-QPSK) is a QPSK mechanism where the maximum phase deviation is limited to ± 135 degrees. The main
advantage of π/4 shifted
QPSK is
coherently, that is, the receiver need not lock to the phase of the
transmitted signal, which simplifies the receiver design. It provides the bandwidth efficiency of QPSK along with lesser fluctuations in amplitude. π/4shifted QPSK is used in the North American digital cellular TDMA standard, IS-136, and also in the Japanese digital cellular (JDC) standard.
Differential PSK (DPSK) is a variation of the basic PSK mechanism. Here, binary 1 is represented by the presence of a carrier signal whose phase has been changed relative to the phase of the carrier used for representing the previous bit. A binary 0 is represented by
the presence of a carrier wave whose phase is the same as that of
the carrier used (e) depicts the
for transmitting the previous bit. Figure 1.4
DPSK pattern when applied on the bit pattern
of Figure 1.4 (a). DPSK comes with an added advantage. Since phase differences occur continuously for long runs of 1s, this can be used for self-clocking. If there are only two possible phase differences, used for representing bit 0 and bit 1, then the modulation technique is called differential binary PSK (DBPSK). If four phase differences are used, for representing the bit sequences 00, 01, 10, and 11, then the scheme is called differential quadraturePSK (DQPSK). If a greater number of phase differences, say, 8 or 16, is used, then the corresponding systems are called 8-DPSK and 16-DPSK, respectively. π/4-DQPSK is another modulation technique, which is a variant of the DQPSK technique. In π/4-DQPSK, an additional phase shift of π/4 radians is inserted in each symbol. In standard DQPSK, a long run of 0s at the data input would result in a signal with no phase shifts at all, which makes synchronization at the receiver very difficult. If π/4-DQPSK is used in such a situation, the phase shift of π/4 ensures that there is a phase transition for every symbol, which would enable the receiver to perform timing recovery and synchronization.
Another variant of PSK is the π/4 shifted PSK (π/4-PSK). This is a four- level PSK technique, hence each phase shift represents two bits. A pair of bits are represented by varying the phase of the carrier signal relative to the phase of the carrier signal used for representing the preceding pair of bits. Table 1.2 shows the phase changes used for various pairs of bits. This method also provides for self-clocking since there is always a phase shift between the transmissions of
1.5 illustrates
the π/4 shifted PSK mechanism for a bit string 110001.
Table 1.2. Phase changes used in π/4 shifted PSK
Figure 1.6 depicts the constellation pattern for a system using four phase changes with two amplitude values. By using a large number of phase changes/amplitude combinations, it would be possible to encode multiple bits using each of those combinations. Hence, spectrum utilization is greatly improved inQAM. Several higher level QAM schemes which are more bandwidth-efficient are in use today. 16-QAMand 64-QAM are two such schemes where 16 and 64 different phase changes/amplitude combinations, respectively, are used. The main drawback of QAM systems is that, as the number of phase changes/amplitude combinations are increased, the system becomes more complex and is more susceptible to errors caused due to noise and distortion. Figure 1.6. Constellation pattern in 8-QAM. " id="pdf-obj-21-2" src="pdf-obj-21-2.jpg">
Figure 1.5. Operation of π/4 shifted PSK.
The next modulation mechanism to be discussed here is quadrature amplitude modulation (QAM). Both the amplitude and the phase are varied in QAM in order to represent bits. In QPSK, each shift in phase would be used to code a pair of bits. But if two amplitude values were used in combination with regular QPSK, it would be possible to obtain eight different combinations. And hence it would now be possible for each of those combinations to encode three bits. This effectively increases the data rate of the system. The different combinations possible are also known as constellation patterns. Figure 1.6depicts the constellation pattern for a system using four phase changes with two amplitude values. By using a large number of phase changes/amplitude combinations, it would be possible to encode multiple bits using each of those combinations. Hence, spectrum utilization is greatly improved inQAM. Several higher level QAM schemes which are more bandwidth-efficient are in use today. 16-QAMand 64-QAM are two such schemes where 16 and 64 different phase changes/amplitude combinations, respectively, are used. The main drawback of QAM systems is that, as the number of phase changes/amplitude combinations are increased, the system becomes more complex and is more susceptible to errors caused due to noise and distortion.
Figure 1.6. Constellation pattern in 8-QAM.
of signals, each signal represented as a function of time, frequency, and code. Hence, multiplexing can be performed with respect to one of these three parameters; the respective techniques are termed frequency division multiple access, time division multiple access, and code division multiple access. Apart from these techniques, the medium can be multiplexed with respect to space also, and this technique is called space division multiple access. A brief discussion of each of the above techniques is presented below. Two vectors are said to be orthogonal if the normalized inner product of the two vectors, which is given by the cosine of the angle between the two vectors, is zero. The normalized inner product of two vectors is a measure of similarity between the two vectors. If it is zero, then it implies that the two vectors are totally dissimilar. 1.6.1 Frequency Division Multiple Access " id="pdf-obj-22-2" src="pdf-obj-22-2.jpg">
1.6 MULTIPLE ACCESS TECHNIQUES
The previous section discussed techniques used for the transmission of raw bits on the channel. Since the transmission medium in wireless networks is broadcast in nature, a node (wireless communication device) cannot transmit on the channel whenever it wants to. Multiple access techniques are used to control access to the shared channel. They determine the manner in which the channel is to be shared by the nodes. Some of the basic multiple access techniques used in wireless networks are presented in this section.
Multiple access techniques are based on the orthogonalization 4 of signals, each signal represented as a function of time, frequency, and code. Hence, multiplexing can be performed with respect to one of these three parameters; the respective techniques are termed frequency division multiple access, time division multiple access, and code division multiple access. Apart from these techniques, the medium can be multiplexed with respect to space also, and this technique is called space division multiple access. A brief discussion of each of the above techniques is presented below.
4 Two vectors are said to be orthogonal if the normalized inner product of the two vectors, which is given by the cosine of the angle between the two vectors, is zero. The normalized inner product of two vectors is a measure of similarity between the two vectors. If it is zero, then it implies that the two vectors are totally dissimilar.
1.6.1 Frequency Division Multiple Access
The frequency division multiple access (FDMA) mechanism operates as below. The available bandwidth is divided into multiple frequency channels/bands. A transmitter–receiver pair uses a single dedicated frequency channel for communication. Figure 1.7 depicts the principle behind the operation of FDMA. The frequency spectrum is in effect divided into several frequency sub-bands. Transmissions on the main band of a channel also result in the creation of additional signals on the side bands of the channel. Hence the frequency bands cannot be close to each other. Frequency bands are separated from each other by guard frequency bands in order to eliminate inter- channel interference. These guard bands result in the under- utilization of the frequency spectrum. This is the main disadvantage ofFDMA. FDMA has been widely adopted in analog systems for portable telephones and automobile telephones.
Figure 1.7. Illustration of FDMA.
Figure 1.7 depicts the principle behind the operation of FDMA. The frequency spectrum is in effect divided into several frequency sub-bands. Transmissions on the main band of a channel also result in the creation of additional signals on the side bands of the channel. Hence the frequency bands cannot be close to each other. Frequency bands are separated from each other by guard frequency bands in order to eliminate inter- channel interference. These guard bands result in the under- utilization of the frequency spectrum. This is the main disadvantage ofFDMA. FDMA has been widely adopted in analog systems for portable telephones and automobile telephones. Figure 1.7. Illustration of FDMA. In a cellular network, a central controller known as the base station (BS) dynamically allocates a different carrier frequency to each node, known as the mobile station (MS). Each node is in fact allocated a pair of frequencies for communication, one for the traffic from the MS to the BS (uplink frequency), and the other for carrying traffic from the BS to the MS (downlink frequency). This system, " id="pdf-obj-23-8" src="pdf-obj-23-8.jpg">
In a cellular network, a central controller known as the base station (BS) dynamically allocates a different carrier frequency to each node, known as the mobile station (MS). Each node is in fact allocated a pair of frequencies for communication, one for the traffic from the MS to the BS (uplink frequency), and the other for carrying traffic from the BS to the MS (downlink frequency). This system,
used for two-way communication between a pair of stations (MS and BS here), is called frequency division duplexing (FDD). Since high-frequency transmissions suffer greater attenuation when compared to low-frequency transmissions, high transmission power is required for high-frequency channels for compensating the transmission losses. Power available at an MS is limited. Hence, in order to conserve energy at the MS, the uplink frequency is always lower than the downlink frequency.
Orthogonal Frequency Division Multiplexing
Orthogonal frequency division multiplexing (OFDM) is a multi-carrier transmission mechanism. It resembles FDMA in that both OFDM and FDMA split the available bandwidth into a number of frequency channels. OFDM is based on the spreading of the data to be transmitted over multiple carriers, each of them being modulated at a low rate. The data signal is split into multiple smaller sub- signals that are then transmitted to the receiver simultaneously at different carrier frequencies (sub-carriers). Splitting the carrier into multiple smaller sub-carriers and then broadcasting the sub-carriers simultaneously reduce the signal distortion at the receiver caused due to multipath propagation of the transmitted signal. By appropriately choosing the frequency spacing between the sub- carriers, the sub-carriers are made orthogonal to each other. Though spectral overlapping among sub-carriers occurs, the orthogonality of the sub-carriers ensures error-free reception at the receiver, thereby providing better spectral efficiency. OFDM is sometimes also referred to as discrete multi-tone (DMT) modulation. OFDM is currently used in several applications such as wireless local area networks (WLANs) and digital broadcasting.
1.6.2 Time Division Multiple Access
Time division multiple access (TDMA) shares the available bandwidth in the time domain. Each frequency band is divided into several time slots (channels). A set of such periodically repeating time slots is known as the TDMA frame. Each node is assigned one or more time slots in each frame, and the node transmits only in those slots. Figure 1.8 depicts the concept behind TDMA. For two-way communication, the uplink and downlink time slots, used for transmitting and receiving data, respectively, can be on the same frequency band (TDMA frame) or on different frequency bands. The former is known as time division duplex - TDMA (TDD-TDMA), and the latter as frequency division duplex - TDMA (FDD-TDMA). Though TDMA is essentially a half-duplex mechanism, where only one of the two communicating nodes can transmit at a time, the
small duration of time slots creates the illusion of a two-way simultaneous communication. Perfect synchronization is required between the sender and the receiver. To prevent synchronization errors and inter-symbol interference due to signal propagation time differences, guard intervals are introduced between time slots. Since the sizes of slots are already small, the introduction of guard intervals results in a significant overhead for the system.
Figure 1.8. Illustration of TDMA.
Chapter 3 . 1.6.3 Code Division Multiple Access Unlike other systems such as TDMA and FDMA, code division multiple access (CDMA) does not assign a specific frequency to each user. Instead, every channel uses the entire spectrum. Individual conversations are encoded with a pseudorandom digital sequence. As the narrow-band transmission frequency is spread over the entire wideband spectrum, the technique is also called as spread spectrum. The transmissions are differentiated through a unique " id="pdf-obj-25-6" src="pdf-obj-25-6.jpg">
FDMA requires the device to have the capability of simultaneously receiving and transmitting signals, which leads to increased cost. But when TDMA is used, the device can switch between slots and hence use the same transmitter for receiving also. Hence, the equipment cost in TDMA is less. TDMA is widely used in second- generation cellular systems such as GSM, described in detail in Chapter 3.
1.6.3 Code Division Multiple Access
Unlike other systems such as TDMA and FDMA, code division multiple access (CDMA) does not assign a specific frequency to each user. Instead, every channel uses the entire spectrum. Individual conversations are encoded with a pseudorandom digital sequence. As the narrow-band transmission frequency is spread over the entire wideband spectrum, the technique is also called as spread spectrum. The transmissions are differentiated through a unique
code that is independent of the data being transmitted, assigned to each user. The orthogonality of the codes enables simultaneous data transmissions from multiple users using the entire frequency spectrum, thus overcoming the frequency reuse limitations seen in FDMA and TDMA.
CDMA was first used during World War II by the English Allies to foil attempts by the German army to jam the transmissions of the Allies. The Allies decided to transmit signals over several frequencies in a specific pattern, instead of just one frequency, thereby making it very difficult for the Germans to get hold of the complete signal.
Two types of spread spectrum systems are widely in use today, namely, frequency hopping spread spectrum and direct sequence spread spectrum, which are described below.
Frequency Hopping Spread Spectrum
Frequency hopping spread spectrum (FHSS) is a simple technique in which the transmission switches across multiple narrow-band frequencies in a pseudo-random manner, that is, the sequence of transmission frequencies is known both at the transmitter and the receiver, but appears random to other nodes in the network. The process of switching from one channel to the other is termed frequency hopping. The radio frequency signal is de-hopped at the receiver by means of a frequency synthesizer controlled by a pseudo-random sequence generator. Figure 1.9 illustrates the concept behind FHSS. The figure shows two simultaneous transmissions. The first transmission (darker shade in figure) uses the hopping sequence f 4 f 7 f 2 f 1 f 5 f 3 f 6 f 2 f 3 and the second transmission uses the hopping sequence f 1 f 3 f 6 f 2 f 4 f 2 f 7 f 1 f 5 . Frequency hopped systems are limited by the total number of frequencies available for hopping. FHSS can be classified into two types: fast FHSS and slow FHSS. In fast FHSS, the dwell time on each frequency is very small, that is, the rate of change of frequencies is much higher than the information bit rate, resulting in each bit being transmitted across multiple frequency hops. In slow FHSS, the dwell time on each frequency is high, hence multiple bits are transmitted on each frequency hop. FHSS is now used mainly for short-range radio signals, particularly those in the unlicensed bands.
Figure 1.9. Illustration of FHSS.
of the vector representations of any two codes is zero. Each node transmits using its code. At the receiver, the transmission is received and information is extracted using the transmitter's code. For transmitting a binary 1, the sender transmits its code; for a binary 0, the one's complement of the code is transmitted. Hence, transmission of a signal using CDMA occupies n times the bandwidth that would be required for a narrow-band transmission of the same signal. Since the transmission is spread over a large bandwidth, resistance to multipath interference is provided for inCDMA. Figure 1.10 illustrates the DSSS transmission. DSSS is used all CDMA cellular telephony systems. in The normalized inner product of two vectors is a measure of similarity between the two vectors. It is given by the cosine of the " id="pdf-obj-27-2" src="pdf-obj-27-2.jpg">
The principle behind direct sequence spread spectrum (DSSS) can
be explained easily by the following analogy. Suppose conversations in several languages are occurring in a room. People who
only a particular language
listen to
and follow
conversation taking place in that language alone. They ignore and
principle applies to DSSS. In DSSS, each node is assigned a specific n-bit code, called a chipping code. n is known as the chipping rate of the system. These codes assigned to the nodes are orthogonal to each other, that is, the normalized inner product 5 of the vector representations of any two codes is zero. Each node
transmits using its code. At the receiver, the transmission is received and information is extracted using the transmitter's code. For transmitting a binary 1, the sender transmits its code; for a binary 0, the one's complement of the code is transmitted. Hence, transmission of a signal using CDMA occupies n times the bandwidth that would be required for a narrow-band transmission of the same signal. Since the transmission is spread over a large bandwidth, resistance to multipath interference is provided for inCDMA. Figure
1.10 illustrates the DSSS transmission. DSSS is used all CDMA cellular telephony systems.
5 The normalized inner
product of two
a measure
similarity between the two vectors. It is given by the cosine of the
angle between the two vectors. If the normalized inner product is zero, then it implies that the two vectors are totally dissimilar.
Figure 1.10. Illustration of DSSS.
Complementary code keying (CCK) is a modulation technique used in conjunction with DSSS. In CCK, a set of 64 8-bit code words is used for encoding data for the 5.5 Mbps and 11 Mbps data rates in the 2.4 GHz band of the IEEE 802.11 wireless networking standard (described in detail in the next chapter). The code words used in CCK have unique mathematical properties that allow them to be correctly distinguished from one another by a receiver even in the presence of significant noise and multipath interference. In CCK, sophisticated mathematical formulas are applied to the DSSS codes, which permit the codes to represent a greater amount of information in each clock cycle. The transmitter can now send multiple bits of information through each DSSS code, which makes it possible to achieve the 11 Mbps data rate.
1.6.4 Space Division Multiple Access
The fourth dimension in which multiplexing can be performed is space. Instead of using omnidirectional transmissions (as in FDMA, TDMA, and CDMA) that cover the entire circular region around the transmitter, space division multiple access (SDMA) uses directional transmitters/antennas to cover angular regions. Thus different areas/regions can be served using the same frequency channel. This method is best suited to satellite systems, which often need a narrowly focused beam to prevent the signal from spreading
too widely and in the process becoming too weak. A satellite can thus reuse the same frequency to cover many different regions of the Earth's surface. Figure 1.11 shows how SDMAcould be used for satellite communication.
Figure 1.11. Illustration of SDMA.
Figure 1.11 shows how SDMAcould be used for satellite communication. Figure 1.11. Illustration of SDMA. 1.7 VOICE CODING The primary objective of most wireless networks in existence today is to support voice communication. If just a data signal consisting of binary digits is to be transmitted, the bits to be transmitted can be directly used to modulate the carrier signal to be used for the transmission. This process is not so direct in the case of analog voice signals. The voice coding process converts the analog signal into its equivalent digital representation. The analog speech information to be transmitted is first converted into a sequence of digital pulses. It should then be transmitted without any noticeable distortion. The devices that perform this analog to digital conversion (at the sender) and the reverse digital to analog signal conversion (at the receiver) are known as codecs (coder/decoder). The main goal of a codec is to convert the voice signal into a digital bit stream that has the lowest possible bit rate, while maintaining an acceptable level of quality of the signal, that is, it should be possible to reproduce the original analog speech signal at the receiver without any perceivable distortion or disturbances. Pulse position modulation (PPM) is one of " id="pdf-obj-29-8" src="pdf-obj-29-8.jpg">
1.7 VOICE CODING
The primary objective of most wireless networks in existence today is to support voice communication. If just a data signal consisting of binary digits is to be transmitted, the bits to be transmitted can be directly used to modulate the carrier signal to be used for the transmission. This process is not so direct in the case of analog voice signals. The voice coding process converts the analog signal into its equivalent digital representation. The analog speech information to be transmitted is first converted into a sequence of digital pulses. It should then be transmitted without any noticeable distortion. The devices that perform this analog to digital conversion (at the sender) and the reverse digital to analog signal conversion (at the receiver) are known as codecs (coder/decoder). The main goal of a codec is to convert the voice signal into a digital bit stream that has the lowest possible bit rate, while maintaining an acceptable level of quality of the signal, that is, it should be possible to reproduce the original analog speech signal at the receiver without any perceivable distortion or disturbances. Pulse position modulation (PPM) is one of
the techniques used for converting an analog signal into its digital representation. In PPM, the amplitude and width of the pulse are kept constant. The position of each pulse, with respect to the position of a recurrent reference pulse, is varied according to the instantaneous sampled amplitude of the information signal. PPM requires constant transmitter power since all pulses have the same constant amplitude and duration. PPM is used in the infrared physical layer specification of the IEEE 802.11 standard. The main disadvantage of PPM is that perfect synchronization is required between the transmitter and receiver. Pulse code modulation is another technique which is commonly used for conversion of an analog signal into its corresponding digital form.
1.7.1 Pulse Code Modulation
Pulse code modulation (PCM) essentially consists of three stages, namely, sampling of the analog signal, quantization, and binary encoding. Each of these stages is described below.
The codec converts the analog speech signal to its digital representation by sampling the signal at regular intervals of time. The higher the sampling rate, that is, the shorter the time interval between successive samples, the better the description of the voice signal. The frequency of the samples to be taken, such that quality of the voice signal is preserved, is given by the sampling theorem. According to the sampling theorem, if the original voice signal has a limited bandwidth, that is, it does not contain any component with a frequency exceeding a given value B, then all information in the original signal will be present in the signal described by the samples, if the sampling frequency is greater than twice the highest frequency in the original signal, that is, it is greater than 2 × B. This theorem is also known as Nyquist's theorem. The series of pulses produced after sampling the analog signal, known as pulse amplitude modulation (PAM) pulses, have their amplitudes proportional to that of the original signal.Figure 1.12 (b) shows the PAM pulses generated for the analog waveform of Figure 1.12 (a).
Figure 1.12. Pulse code modulation.
During the quantization phase, the amplitudes of the PAM pulses are
measured and each pulse is assigned a numerical value. In order to
avoid having
thePAM pulses, a fixed number of amplitude levels are used. The amplitude of the analog signal at each sampling instant is rounded
off to the nearest level. Figure 1.12 (c) depicts the output of
quantization applied to the PAM pulses
in Figure
(b). Since
some of the information is lost due to the approximation of amplitude levels, quantization distorts the original analog signal. This distortion is also known as quantization error. But at the same time, since only a fixed number of amplitude levels are used, only a fixed number of numerical values need to be transmitted. This reduces the complexity of the equipment and also the probability of transmission errors.
Binary Encoding
In this final phase, the quantized PAM pulses are encoded into the binary format, which forms the final output of the analog-to-digital conversion process. The number of bits used to represent each amplitude level is based on the total number of amplitude levels that need to be represented. The quantized PAM pulses in Figure 1.12 (c) use seven quantization levels. Therefore, at least three bits would be required for representing each level. The encoded output of the quantized pulses in Figure 1.12 (c) would be 110 110 100 010 011 110 110 101 100 100 101. This bit pattern is then modulated and transmitted by the sender.
1.7.2 Vocoders
The PCM mechanism described above requires high bit rates. For example, if a sampling rate of 8,000 samples/second is used and each sample is encoded into eight bits, then the corresponding data rate would be 64 Kbps. Hence, PCM is not very suitable for wireless networks where the limited bandwidth needs to be utilized efficiently. Vocoders are devices that operate making use of knowledge of the actual structure and operation of human speech production organs. They do not modulate the actual voice signal; instead they work by modeling the mechanics by which sound is produced (e.g., lip movement, and the effect of the throat and nose on speech). This information is encoded and transmitted by the sender, and is received by the receiver, which then reconstructs the original signal. Vocoders achieve voice transfer at low bit rates. Hence they are more suitable compared to PCM for
voice communication in wireless networks. But a drawback of using vocoders is that since the actual voice signals are not transmitted, the signals received and reconstructed at the receiver do not appear to be natural. The artificiality in the reconstructed signal makes it difficult to identify the actual speaker on the other side.
In order to overcome the above problem, certain hybrid codecs are used, which transmit both vocoding information and also the PCM signal information. Certain other codecs are also available, which vary the bit rate of the outgoing signal according to the characteristics of the speech signals fed to them. For more details on vocoders, the reader can refer to [4].
1.8 ERROR CONTROL
On any transmission channel, and especially on the wireless channel, it is never guaranteed that the transmitted data will be delivered as it is without any errors at the destination. There is always a non-zero probability that the bit stream gets altered while in transmission. The bit error rates (BERs – fraction of bits that are received in error) for the wired channel and the wireless channel can be as high as 10 -9 and 10 -4 , respectively. The unreliable nature of the wireless channel makes error control even more important in the wireless networks context.
Several coding techniques are available, which try to provide resistance against errors by adding redundant bits to the transmitted bits. These redundant bits help the receiver to either detect errors and request for a retransmission (error detection), or to identify and correct the faulty bits (error correction). We have different coding schemes for error detection and for error correction. The process of adding the redundant bits to the bit stream is known as channel coding. Some of the commonly used coding techniques are discussed below.
1.8.1 Parity Check
Parity check, which can detect single-bit errors, is the simplest of the error detection techniques. The parity check mechanism works as below. Initially, the transmitter and receiver come together in an agreement on whether the number of 1 bits in the messages they exchange would be odd or be even. If the number of 1s is odd, then the scheme is called odd parity; if it is even, the scheme is called even parity. Each message has an additional bit, called a parity bit, which is the last bit of the message. The parity bit is set to 1 or 0 based on the prior agreement on whether the message should
contain an even or odd number of 1s. Consider the following example. If the message to be transmitted is 10111 and the odd parity scheme is used, the sender appends a parity bit (last bit) of 1 to make the number of 1s odd. Thus the message transmitted would be 101111. If no error occurs in transmission, the receiver would receive the message as it is, it would verify and find that the number of 1 bits is odd in number, and hence conclude that there was no error in transmission. On the other hand, if a single-bit error had occurred (a single bit would have been flipped from 0 to 1 or 1 to 0), the number of ones in the receivedmessage would not be odd. The receiver hence concludes that the message became corrupted in transmission and requests the sender for a retransmission.
The parity check mechanism cannot detect a multiple of two-bit errors in the same message. In the above example, if the received message had been 101001, the receiver would have wrongly concluded that the message had been received correctly, though actually two bits are in error. This scheme is guaranteed to detect only single-bit errors. Another disadvantage of this scheme is that the receiver can only detect the error and not correct it since it does not know the exact bit in error.
For correcting errors, the Hamming distance of the binary streams/binary words used in the communication becomes important. The Hamming distance of a set of binary streams is the minimum number of bits that need to be inverted for converting one stream from the set into another valid stream of the set. For example, if the set of binary streams used in the communication consists of only two patterns 000000 and 000111, the Hamming distance becomes three. Hence, up to two-bit errors can be detected successfully, and all one-bit errors in the received messages can be corrected. For example, if the message received is 100111, since the Hamming distance is three, the receiver would conclude that if a one-bit error had occurred, then the actual transmitted message should have been 000111. In general, to detect d errors, a Hamming
distance of at least d+1 is required,
and for correcting d errors a
Hamming distance of at least 2d + 1 is required. This would make the valid bit streams to be far enough from each other so that even when up to d bits go in error, the original transmitted bit stream is closer to one of the valid bit streams than any other.
1.8.2 Hamming Code
The Hamming code works as follows. Each valid binary stream of length n is said to consist of d data bits and r redundant/check bits (n = d + r). The bits of the bit stream are numbered consecutively
from bit 1, the most significant bit (MSB), at the left end. Bits that
are powers
(1,
16, etc.)
check bits. The
remaining positions (3, 5, 6, 7, 9, etc.) are taken up by the data bits. Each check bit is set to 1 or 0 such that the parity of the bits it represents, including itself, is even or odd (depending on whether
even parity or odd parity is used). In order to determine to what check bits a data bit contributes to, the position of the data bit is written in powers of 2. The data bit is checked by the check bits occurring in the expansion. For example, 11 can be written as 1 + 2
+ 8. Hence it is checked by the check bits in positions 1, 2, and 8. When a bit stream arrives at the receiver, it initializes a counter to
zero. It then examines each check bit in position i (i = 1, 2, 4, 8, and checks whether it satisfies the parity condition (even or odd). If
not, it adds the position of that check bit to the counter. After all check bits have been examined, if the counter value is zero, then it
implies
Otherwise, the counter contains the position of the incorrect bit. For example, if check bits 1 and 2 are in error, it implies that the third bit has been inverted. This is because it is only bit 3 that is checked by bits 1 and 2 alone. Hamming codes can correct only single-bit errors.
1.8.3 Cyclic Redundancy Check
Cyclic redundancy check (CRC) is one of the widely used error detection techniques. The CRCtechnique is used for protecting blocks of data called frames. The transmitter appends an additional n-bit sequence called frame check sequence (FCS) to each frame it transmits. The FCS holds redundant information regarding the frame, which helps the receiver detect errors in the frame. The CRCalgorithm works as follows. It treats all bit streams as binary polynomials. The transmitter generates the FCS for a frame so that the resulting frame (FCS appended to the original frame) is exactly divisible by a certain pre-defined polynomial called the divisor or CRC polynomial. At the receiver, the received frame is divided by the same CRC polynomial. The frame is considered to be free of errors if the remainder of this division is zero; if not, it implies that the received frame is in error. CRC has the advantages of very good error detection capabilities, less overhead, and ease of implementation.
1.8.4 Convolutional Coding
Techniques such as the Hamming code and CRC work on fixed blocks of data. They would not be suitable for a continuous data stream. Convolutional coding is well-suited for long bit streams in noisy
channels. Convolutional codes are referred to, based on their code rates and constraint length. The code rate r of a convolutional code gives the ratio of the number of bits k that were fed as input to the convolutional coder to the number of bits n at the output of the coder, that is, r = k/n. The constraint length K of a convolutional code denotes the length of the convolutional coder, that is, the number of stages used in the coder. The larger the constraint length, the lesser the probability of a bit suffering an error. Operation of the encoding process can be best explained using an example. Consider
the convolutional coder shown in Figure 1.13. It consists of four stages, that is, K = 4, and a code rate r = 1/3. Initially, the first bit
(MSB) is
coder,
and three bits are
obtained as the result of coding this bit. This process continues till
the last bit of the input bit stream leaves the last stage of the coder.
For a bit pattern
1101, the output
the coder in Figure
1.13 would be 111 100 001 110 000 010 011 000. The operation of the coder depends on various factors such as the number of stages used, the number of XOR adders used, and the way connections are made from the stage outputs to the adders.
Figure 1.13. A convolutional coder.
convolutional codes. The first mechanism called sequential decoding performs well with convolutional codes having a large K value, but it has a variable decoding time. The second type of decoding, called Viterbi decoding, has a fixed decoding time, but the computational complexity for this decoding is high. For more details on convolutional coding, the reader can refer to [5].
1.8.5 Turbo Codes
Turbo codes are very powerful error-correcting codes. Turbo codes were first proposed by C. Berrou et al. in 1993 [6]. Turbo codes are
considered to be one of the very important breakthroughs in channel coding theory. The turbo codes exhibit performance closer to the theoretical Shannon limit than any other code. They have very good error-control properties, and at the same time the decoding process is computationally feasible. These two features make the turbo codes very powerful. Figure 1.14 shows a basic turbo code encoder. It uses two recursive 6 systematic 7 convolutional (RSC) codes with parallel concatenation. In Figure 1.14, the first RSC/convolutional encoder uses the bit stream as it arrives. The second convolutional encoder receives the bit stream through the interleaver. The interleaverrandomizes the input sequence of the second encoder, such that the inputs of the two encoders are uncorrelated. The two encoders introduce redundant bits for the same block of input data bits. However, because of the interleaver which alters the bit pattern input to the second encoder, the redundant bits generated by the two encoders are different. The received data would have two different coded sequences (generated by the two convolutional encoders). An iterative decoder is used at the receiver. A turbo decoder that uses a single iteration is shown in Figure 1.15. The first decoder decodes the input sequence to give a soft output. This soft output is a measure of the reliability of each decoded information bit (whether or not any errors had occurred). This reliability information is called extrinsic information, that is, it does not depend on the current inputs to the decoder. The extrinsic information from the first decoder, after interleaving, is passed to the second decoder. The second decoder uses this information to decode the interleaved input bit sequence. From the soft outputs of the second decoder the new extrinsic information, which again is a measure of the reliability of the decoded bits, is fed back to the first decoder, and the process continues. The final output is taken from the output of the second decoder. The performance of the turbo coding scheme improves as the number of decoder iterations is increased.
6 A code is said to be recursive if one of the feed forward paths in the encoder is fed back into its input.
7 A systematic code is one in which the original data bits
modified; the redundant bits are just appended to the data bits.
Figure 1.14. Turbo code encoder.
Figure 1.15. Turbo code decoder.
1.9 COMPUTER NETWORKS
Having discussed the fundamentals of wireless communication technology, it would now be appropriate to know how wireless networks operate. But before going into the specifics of wireless networks, it would do well to understand the principles behind the operation of a general computer network. A computer network interconnects autonomous computers/nodes. Computer networks are almost indispensable in today's world. They have become very essential for day-to-day life and are used by a wide gamut of applications. Some of the very common applications are electronic mail systems, ticket reservation systems, credit card systems, and video conferencing.
Computer networks can be broadly classified under two main categories, namely, client/server networks and peer-to-peer networks. A client/server network involves processes, client processes and server processes, which usually reside on different machines. The client processes request and receive services from the server processes. The client/server model is widely used for
sharing resources such as files, printers, and other applications. For example, a Web server could be a server process receiving requests for Web pages from Web browsers (client processes) running on different machines, and sending them back the required Web pages. Other popular applications such as electronic mail clients and file transfer session clients (FTP clients) also follow the client/server model. A peer-to-peer network, on the other hand, does not require any dedicated server. Every computer on the network can share its resources (e.g., files or printers) with any other computer on the network, provided sufficient access privileges are granted for those computers. In effect, every computer connected to the network is both a client and a server. The music file-sharing system, Napster, was once the most popular peer-to-peer application on the Internet.
Based on the transmission technology used, networks can be classified under two broad categories, namely, broadcast networks and point-to-point networks. A broadcast network uses a single shared communication channel. A message transmitted by a node can be heard by all other nodes in the network. Each message carries the address of its intended destination. The destination node, on finding its address on the message, accepts the message. The other nodes just ignore the message. A broadcast network is usually used to connect machines within a small geographic area. In a point- to-point network, the communication channel is not shared. Two nodes are connected through a dedicated link. Data transmitted by the source node can be received only by the intended next hop receiver node. Hence, the transmitted data might have to traverse through multiple hops in order to reach the final destination. Such point-to-point links are used to connect machines that are physically separated by large geographic distances. Based on the geographic span of the network, networks can be classified as LANs, MANs, or WANs.
Local area networks (LANs) are the most common networks one can find today. As the name suggests, a LAN is used to connect computers that are located within a small geographical area, such as within office buildings or homes. In a LAN, all the computers are attached to a common broadcast medium. Ethernet is the most common physical medium used for connecting computers in a LAN. A metropolitan area network (MAN) covers a much larger area compared to that covered by a LAN. AMAN may be used to connect computers that are located at different offices or those that are located in the same city or town. Some of the common technologies used in MANs are asynchronous transfer mode (ATM) (which will be described later in this chapter) and fiber distributed data interface (FDDI). 8 Networks that are used to connect computers located in a
large geographical area, which may span different cities or even countries, are called wide area networks (WANs). WANs may be connected through phone lines, satellite links, or optical fibers. The transmission medium in a WAN may include several point-to-point links. A network that connects two or more of the above three types of networks, that is, LANs, WANs, or MANs, is called an internetwork.
8 FDDI
a token-based LAN standard
that uses
network in which nodes are interconnected by optical fiber links.
Any network requires network hardware and software in order to operate. Network hardware may include devices such as network cards, switches, routers, and gateways. Network software was not given much importance in the early networks. But today, the network software plays an extremely important role in any type of network.
1.10 COMPUTER NETWORK SOFTWARE
In order to reduce its complexity, the computer network software is organized as a set of layers. Every layer has a specific function to perform. It offers its services to its next higher layer. In order for layers to access the services provided by their lower layers, each layer (except the bottom-most layer) is provided with a service access point (SAP). The SAP is an interface through which the higher layer can access the services offered by its immediately next lower layer. The lower ranking layer is referred to as the service provider, and the higher ranking layer as the service user. The information passed through the SAP is known as an interface data unit (IDU). The IDU typically consists of a service data unit(SDU), which is the actual data, and associated control information. For its convenience, a layer may split the received SDU into several units before sending it to the next lower layer. Each such unit is termed as a protocol data unit (PDU). The higher layer is in no way aware of the manner in which the services are actually implemented in the lower layer. Hence, effecting changes in a layer does not affect any of the other layers. Each layer at the sender machine is said to logically communicate with its peer layer on the destination machine. The rules used for this communication between peer layers, that is, layer n on the sender and the same layer n at the receiver, are collectively termed the layer n protocol.But the actual information flow between two connected machines occurs as shown in Figure 1.16.Figure 1.16 depicts a four-layer model. The collection of protocols used by the system, one protocol per network layer, is referred to as the protocol stack.
Figure 1.16. Different layers in computer network software.
At the sender side, the top-most layer, on receiving data from the application, adds its own header to the data packet (a packet refers to the basic unit of data that is transmitted on the medium) and sends it to the next lower layer. Each intermediate layer, on receiving this data, appends its own layer-specific header and sends it to the next lower level layer. This process continues till the data reaches the lower-most layer. This lowest layer transmits the received data packet on the physical medium. When the packet transmitted by a node reaches its destination, each layer at the destination node strips off its corresponding peer layer header from the packet, and sends the packet to the next higher layer. This process continues till the data reaches the top-most layer, which then delivers it to the appropriate process. The layers, along with their respective protocols, constitute the computer network architecture. The computer network architecture does not pertain to any specific hardware or software. Using the network architecture, an implementer would be able to design his/her hardware and software for the network.
1.11 COMPUTER NETWORK ARCHITECTURE
As defined previously, the network layers,
respective protocols, constitute the computer network architecture. Several such computer network architectures/models are in existence today. Prominent models such as the OSI reference model, the TCP/IP reference model, and the ATMreference model will be discussed now.
1.11.1 The OSI Reference Model
This model was proposed
by the International Organization for
Standardization (called ISO in short),
national standards bodies representing more than 140 countries. The objective of this standard was to standardize the protocols used in the various network layers. This model was named as the open systems interconnection (OSI) reference model, since the main objective of the model was to specify mechanisms for communication between systems (machines) in a telecommunication network that are open for communication with other systems. The OSI reference model is depicted in Figure 1.17. It consists of seven layers, namely, the physical layer, data link layer, network layer, transport layer, session layer, presentation layer, and the application layer. Each layer is associated with a unique set of functions and responsibilities. The following is a brief description of the seven layers.
Figure 1.17. OSI reference model.
The physical layer, which is the lowest layer, is responsible for the transmission of the bit stream over the physical medium. It deals with the mechanical and electrical specifications of the network hardware and the physical transmission medium to be used for the transmissions. The mechanical specifications refer to the physical dimensions of the devices such as cables, connectors, and pins used for interconnection. The electrical specifications include details such as voltage levels to be used for representing digital information (binary 0s and 1s) and the duration of each bit. The physical layer acts as a direct interface to the actual physical transmission medium.
The main objective of the next higher layer, the data link layer, is to ensure error-free transmission of data across a physical link. It receives data from the higher layer, splits the data into several frames/packets, and transmits the frames. It may have mechanisms for detecting and retransmitting damaged or lost frames. The data link layer is also responsible for implementing link-level congestion control mechanisms that ensure that a fast sender does not flood a slow receiver. In broadcast networks, the data link layer performs another important function, which is that of medium access control. It determines when, that is, at what point of time, the current node can access the channel for its transmissions so that collisions arising due to simultaneous transmissions by multiple nodes are minimized.
A detailed description of the data link layer is provided in the next section.
The network layer is responsible for routing data packets from the source node to the destination node. The network layer is also responsible for node addressing. A packet may have to traverse across several networks in order to reach its destination. Each of these networks might follow different addressing schemes. The network layer has to make sure that this heterogeneity does not affect end-to-end packet delivery. Congestion control in the local network is also the responsibility of the network layer.
Next comes the transport layer. The transport layer provides the higher layers a network independent interface to the lower layers. The functions of the transport layer include segmentation and reassembly of messages, end-to-end error recovery, monitoring of quality of service (QoS), and end-to-end flow control.
The fifth layer in the OSI reference model is the session layer. A session can be defined as a connection between two presentation layer (the next higher layer) processes. It can be thought to be a connection between two users on two different machines. The services provided by the session layer to the presentation layer are with respect to these sessions. They include establishment and release of session connections, interaction management (which determines whether the session is two-way simultaneous, or two- way alternate, or a one-way interaction session), and synchronization between the two ends of the session.
The presentation layer, the sixth layer from the bottom, is concerned with the syntax and semantics of the information exchanged between the two end systems. The presentation layer ensures that the messages exchanged between two processes that reside on the different computers that use different data representations, have a common meaning. In order for such computers to communicate, the data structures that need to be transmitted can be defined in a common abstract manner. The presentation layer takes care of the management of such abstract data. Other functions of the presentation layer include data encryption and data compression.
The top-most layer in the OSI protocol stack is the application layer. The application layer acts as an interface to the application processes that require communication support. It provides mechanisms for supporting tasks such as data transmission between users, distributed database access, running processes on remote machines, controlling distributed systems, and electronic mail
communication.Some examples of protocols used in this layer are virtual terminal (TELNET), which provides remote login facility for users, file transfer protocol (FTP), used for transferring files between machines, simple mail transfer protocol (SMTP), that is used for electronic mail applications, and hypertext transfer protocol (HTTP), which is used for transferring Web pages over the Internet.
Shortcomings of the OSI Reference Model
Though the OSI model was elaborately designed, it never took off in reality. By the time the OSI model appeared, other protocols, in
particular the TCP/IP protocol,
already in use widely. All
products in the market were designed according to those protocols. The manufacturers were hesitant to again invest in products based on this new OSI model. Hence, the OSI model was never implemented.
Regarding the technical aspects, it was felt that the OSI model had too many layers. The session and presentation layers were not required for most applications. The model itself was thought to be very complex and difficult to implement efficiently. Several layers in the model were performing redundant operations. Tasks such as flow control and error control were being performed by many layers in the model. As such, the model was not very efficient. The few initial implementations of the model were bulky and slow. Since other better models such as the TCP/IP were already available, people preferred such models.
1.11.2 The TCP/IP Reference Model
The TCP/IP reference model is the network model that is used in today's Internet. The model is widely referred to as TCP/IP protocol stack since it was originally designed as a protocol stack for the Advanced Research Project Agency Network (ARPANET) 9 in the late sixties, rather than as a reference model. It is named so after two of its main protocols, transmission control protocol (TCP) and Internet protocol (IP). In contrast to the OSI reference model, the TCP/IP reference model has only four layers. This model was far better than the OSI model because, unlike the OSI model, the protocols that are part of this model were already in use before the model was actually designed. Hence, they fitted perfectly into the model. Figure 1.18 shows the layers present in the TCP/IPreference model. The figure also shows some of the protocols used in each layer. The four layers of theTCP/IP model are the host-to-network layer, internet/network layer, 10 transport layer, and the application layer. The session and presentation layers that found place in
the OSI reference model were not included in the TCP/IP model since they were found to be not very significant.
9 The ARPANET, sponsored by the U.S. Department of Defense, was the first major effort at developing a network to interconnect computers over a wide geographical area.
10 The term internet (i in lowercase) refers to the interconnection of individual networks using devices such as routers and gateways. The term Internet (I in uppercase) refers to a specific worldwide network.
Figure 1.18. TCP/IP reference model.
The ARPANET, sponsored by the U.S. Department of Defense, was the first major effort at developing a network to interconnect computers over a wide geographical area. The term internet (i in lowercase) refers to the interconnection of individual networks using devices such as routers and gateways. The term Internet (I in uppercase) refers to a specific worldwide network. Figure 1.18. TCP/IP reference model. The lowest layer, the host-to-network layer, is used to interface the TCP/IP protocol stack with the physical transmission medium. The operation of this layer is not very well-specified in the TCP/IPmodel. The host machine should connect to the network using some protocol in order to transmit the packets sent to it by the next higher layer, the internet layer. The internet layer is similar in functionality to the network layer in the OSI reference model. The main functions of the internet layer are routing and congestion control. The internet layer should provide mechanisms for the host to inject packets into any network and have those packets travel independently to the destination. The internet layer defines a protocol called Internet protocol (IP). The packets sent by the internet layer to the lower layer follow this IP format. The routing of packets is done on an individual packet-by-packet basis. Hence the packets may arrive out of order at the destination. It is the responsibility of the next higher layer, the transport layer, to reorder the received packets. The transport layer in the TCP/IP model performs similar operations as the transport layer in the OSIreference model. The fact is that the " id="pdf-obj-46-14" src="pdf-obj-46-14.jpg">
The lowest layer, the host-to-network layer, is used to interface the TCP/IP protocol stack with the physical transmission medium. The operation of this layer is not very well-specified in the TCP/IPmodel. The host machine should connect to the network using some protocol in order to transmit the packets sent to it by the next higher layer, the internet layer.
The internet layer is similar in functionality to the network layer in the OSI reference model. The main functions of the internet layer are routing and congestion control. The internet layer should provide mechanisms for the host to inject packets into any network and have those packets travel independently to the destination. The internet layer defines a protocol called Internet protocol (IP). The packets sent by the internet layer to the lower layer follow this IP format. The routing of packets is done on an individual packet-by-packet basis. Hence the packets may arrive out of order at the destination. It is the responsibility of the next higher layer, the transport layer, to reorder the received packets.
The transport layer in the TCP/IP model performs similar operations as the transport layer in the OSIreference model. The fact is that the
design of the transport layer in the OSI model was actually influenced by the operation and performance of TCP. Two end-to-end communication protocols, the transmission control protocol (TCP) and the user datagram protocol (UDP), have been defined in this layer. TCP is a reliable connection-oriented protocol. Here, establishment of an end-to-end connection that may traverse multiple hops is required before the actual data transfer can begin. All packets belonging to a session follow the same path, and are received at the destination in the same order as they were transmitted by the source node. The incoming byte stream at the source node is split into several messages and sent to the internet layer. The received messages are reassembled by the transport layer at the destination. In order that a fast sender does not swamp a slow receiver, TCP also performs flow control. UDP is an unreliable connectionless protocol. It does not have the flow control and sequencing mechanisms present in TCP. UDP is used by applications such as voice/video transmissions where fast delivery of information is more important than the accuracy of the information delivered.
The application layer is the
topmost layer in the TCP/IP model. It
consists of higher-level protocols, which include the FTP, TELNET, SMTP, and HTTP.
Shortcomings of the TCP/IP Reference Model
One of the main disadvantages of the TCP/IP model is that it is too specific to the TCP/IP protocol stack. It can be said that the model is not generic enough to describe other protocol stacks. It is difficult to design new networks with new technologies using the TCP/IP reference model. Another shortcoming of the TCP/IP model is the host-to-network layer. This layer is actually an interface connecting the machine to the network. An interface does not really require an exclusive layer for itself. Also, the data link layer which performs important functions such as framing, error control, link-level flow control, and medium access control (in broadcast networks) is absent in the TCP/IP model. Irrespective of all these shortcomings, the TCP/IP reference model still remains the most successful model and even forms the backbone of today's Internet.
1.11.3 The ATM Reference Model
The asynchronous transfer mode (ATM) model described below was developed by the ITU for broadband-integrated services digital network (B-ISDN). The ATM model is quite different from theOSI and TCP/IP reference models described previously. Here, all
information is transmitted in the form of short fixed-sized packets called cells. The size of each cell is 53 bytes, a 5-byte header followed by a 48-byte payload. ATM uses a switching technology called cell switching. Before going into details of cell switching, the two other commonly used switching technologies, namely, packet switching and circuit switching, are described briefly below.
Circuit switching requires the setting up of an end-to-end path between the source and destination nodes before the actual data transmission begins. A request packet sent by the source node travels till the destination node. The destination responds with an acknowledgment. When the source node receives this acknowledgment, implying that the path has been set up, it starts transmitting packets. The request-response process also reserves the required resources (bandwidth) on the path. Once a path is set up, all packets belonging to the flow follow the same path. The packets are received at the destination in the same order they were transmitted at the source. Since resources are reserved a priori, the problem of link congestion rarely occurs in circuit switched networks. But if a reserved path is sparingly used or not used at all, then the reserved resources remain not utilized, resulting in under-utilization of the available network capacity. Circuit switching is widely used in telephone networks where once a connection is established between the two end users, the circuit (connection) continuously remains open from the start till completion of the call. Circuit switching may not be suitable for data or other non-conversational transmissions. Non-voice transmissions tend to occur in bursts, and if circuit switching is used for such transmissions, the transmission line often remains idle, resulting in under-utilization of the available network capacity.
Another switching technique, which is not very popular and is not used commonly, is message switching. Here the sender transmits data in blocks. There is no limit on the size of the blocks. An intermediate node receives the entire block of data from its uplink node (uplink node refers to the next hop neighbor node on the path from the current node to the source node of the data session), stores the block in its disk, checks for errors, and then transmits it to the next downlink node (downlink node refers to the next hop neighbor node of the current node on the path from the current node to the destination node). Since there is no limit on the size of blocks transmitted, a single large block of data may hog the channel for a long period of time. Hence, message switching is not suitable for interactive applications and time-sensitive applications such as voice and video transmissions. The third and most widely used switching technique, packet switching, solves the above problems.
In packet switching, a strict bound is set on the maximum size of the blocks transmitted. Hence the packets can now be held in the main memory of intermediate routers rather than in their disks. Since a single connection cannot hog a transmission channel indefinitely, it is suitable for non-conversational transmissions. A node can transmit each packet independently. Unlike message switching, it need not wait for a whole block of data (consisting of many packets) to be received before the transmission of the received block can be started. Hence, the end-to-end delay involved is less, and the throughput of the system is also high. Since resources are not reserved beforehand, the links may become congested when the network load increases, resulting in packet loss. In packet switching, each packet is routed (and not switched) independently at each intermediate node. Packets belonging to the same flow may hence take different paths to reach the destination. Due to this, it is highly probable that the packets are received out of order at the destination node.
Continuing with ATM and cell switching, cell switching is similar to packet switching in that it splits a data stream into multiple equal- sized cells which are then transmitted on a channel that is shared by other nodes also. The main difference between cell and packet switching techniques is that while cells have a fixed size, packets can have different sizes. One of the main reasons for having fixed- length cells is to facilitate the building of fast hardware switches. The design of fast and highly scalable hardware switches would be made simple if all packets were of fixed length. The processing of packets would be made simple if the size of the packets were known beforehand. Also, parallelism could be incorporated into the switching system if all packets were of the same length; multiple switching elements could be working in parallel performing the same operation on different packets. This improves the scalability of the switching system. Hence, the ATM design had fixed length cells.
Though an ATM network uses cell switching, it is in some ways similar to a circuit switched network. Every connection goes through a set-up phase, during which a virtual circuit 11 is established. The virtual circuit has several advantages over a physical circuit. Reservations can make use of statistical multiplexing (multiplexing in which channels are established on a statistical basis, that is, connections are established on demand according to the probability of need, using the statistical information about the system, that is, information regarding the data arrival pattern at nodes). Also, once a virtual circuit is established for a connection, the switching time for packets belonging to that connection gets minimized. This increases the network throughput significantly. An ATM switch is never allowed
to reorder the cells/packets. It may drop the packets, though, when the network becomes overloaded. Virtual circuits are unidirectional. The ATM model is depicted in Figure 1.19. Unlike the previous two models, the ATM model has been defined as a three-dimensional model consisting of layers and planes. The three main layers are the physical layer, the ATM layer, and the ATM adaptation layer. The three planes used in this model are the user plane, the control plane, and the management plane.
11 A virtual circuit is a path between two points in the network that appears to be a dedicated physical path, but in reality is a managed pool of network resources from which specific flows/paths are allocated resources based on the need, such that the traffic requirements of the flows are met.
Figure 1.19. ATM reference model.
Figure 1.19 . Unlike the previous two models, the ATM model has been defined as a three-dimensional model consisting of layers and planes. The three main layers are the physical layer, the ATM layer, and the ATM adaptation layer. The three planes used in this model are the user plane, the control plane, and the management plane. A virtual circuit is a path between two points in the network that appears to be a dedicated physical path, but in reality is a managed pool of network resources from which specific flows/paths are allocated resources based on the need, such that the traffic requirements of the flows are met. Figure 1.19. ATM reference model. The user plane is responsible for operations such as cell transport, error control, and flow control. The control plane deals with connection management. Typical operations on this plane include call setup and call tear-down. The third plane, the management plane, maintains the network and carries out operational functions. This plane is subdivided into layer management and plane management to manage the different layers and planes. The concept of user and control planes enables streamlining of the " id="pdf-obj-50-12" src="pdf-obj-50-12.jpg">
The user plane is responsible for operations such as cell transport, error control, and flow control. The control plane deals with connection management. Typical operations on this plane include call setup and call tear-down. The third plane, the management plane, maintains the network and carries out operational functions. This plane is subdivided into layer management and plane management to manage the different layers and planes. The concept of user and control planes enables streamlining of the
implementation in each plane, which improves the performance of the system.
As in the OSI reference model, the physical layer deals with the actual physical medium used, which includes the voltage representation to be used for bits, the duration of each bit, and other physical medium dependent issues. This layer is divided into two sublayers. The lower sublayer, called the physical medium dependent (PMD) sublayer, directly interfaces with the physical medium and takes care of the actual transmission of bits and the bit timing. The upper sublayer of the physical layer is called the transmission convergence (TC) sublayer. At the sender side, the TC sends the incoming cells as a string of bits to the PMD, which transmits the bit stream. At the receiver, the TC sublayer receives the bit stream from the PMD sublayer. The TC sublayer, which is aware of details such as where cells begin and end (cell boundaries), then converts the incoming bit stream into cells and delivers them to the next higher layer, the ATM layer.
The ATM layer is the
the ATM reference model. It is responsible for routing cells across the
network. Functions of the ATM layer include definition of the cell layout, establishment and release of virtual circuits, and congestion control. The ATM layer is equivalent to the combined network and data link layers in the OSI reference model.
The next higher layer is the ATM adaptation layer (AAL). This layer is where user information is created and received as 48-byte payloads. It consists of two sublayers, a segmentation and reassembly (SAR) sublayer and a convergence sublayer (CS). The CS sublayer which is the upper sublayer packages the higher layer's PDUs with any additional information required for adapting them to the ATMservice. On receiving a higher-layer PDU, the CS sublayer adds a header and trailer to the receivedPDU, resulting in a CS-PDU, which is passed on to the lower SAR sublayer. As the name implies, on the sender side the SAR sublayer segments/breaks the PDUs received from the higher layers into cells, and at the receiver side it reassembles/puts them back together again. When a CS-PDU is received by theSAR sublayer at the sender, it breaks it into small units, adds its own header and trailer to each unit, and the resultant 48-byte unit is known as SAR-PDU. This SAR-PDU is sent to the ATM layer, which adds a 5-byte header, resulting in a 53-byte ATM-PDU or what is known as a cell. The reverse process takes place at the receiver. The SAR sublayer collects multiple SAR-PDUs, strips off the headers and trailers, assembles a CS-PDU, and sends it to the CS sublayer. The CS sublayer in turn removes its own header and trailer and
gives the resulting higher-layer PDU to the higher layer. Several types of AALs have been recommended by the ITU Telecommunication Standardization Sector (ITU-T) for supporting different types of data. AAL1 and AAL2 provide real-time service guarantees to connections. AAL1 provides support for constant-bit-rate (CBR) connections, mostly used for voice and video transmissions. AAL2 supports variable-bit-rate (VBR) connections. AAL3/4 is used for supporting conventional packet switching services. AAL5, which is sometimes also known as simple and efficient adaptation layer (SEAL), is used by applications that can use ATM services directly.
The above described layers are the ones actually defined by the model. Additional upper layers may include optional protocol layers used for further encapsulating ATM services for use with other protocols such as TCP/IP. Figure 1.20 shows a comparison of the ATM reference model with the other two reference models described previously in this chapter, namely the OSI reference model and theTCP/IP reference model.
1.20. Comparison
reference models.
ATM,
OSI, and TCP/IP
of ATM is
can provide
differentiated services through the use of four service classes.
The CBR class provides services similar to
communication line. It offers the best guarantees of packet delivery. The second service class is the VBR service class. This class is further divided into two subclasses, real-time (VBR-RT) and non-real- time (VBR-NRT). VBR-RT is commonly used for time-sensitive transmissions such as voice and real-time video. VBR-RTprovides better congestion control and delay guarantees than VBR-NRT. The next service class, available-bit-rate (ABR) class, is sort of a hybrid between the CBR and the VBR classes. The ABR class delivers cells at a minimum guaranteed rate. If more bandwidth is available, the minimum rate can be exceeded. This class ideally suits bursty traffic. The fourth service class, called the unspecified-bit-rate(UBR) class, supports best-effort delivery of cells. It provides no delay or bandwidth guarantees.
Shortcomings of the ATM Reference Model
The ATM reference model is not widely in use today. It has several shortcomings. Each 53-byte cell has a 5-byte header, which is nearly ten percent of the cell/frame. This constitutes a significant control overhead on the system. Complex mechanisms are required for ensuring fairness among connections and provisioning quality of service. The delay jitter due to the varying delays faced by the packets, requires complex packet scheduling algorithms at the ATM switches. The high cost and complexity of devices based on the ATM technology, and the lack of scalability are the other reasons that contributed to the failure of ATM.
1.12 IEEE 802 NETWORKING STANDARD
The Institute of Electrical and Electronics Engineers (IEEE) has defined several standards for LANs. Such standards collectively come under the IEEE 802 standard. The following are some of the originalIEEE standards: 802.1 - internetworking; 802.2 - logical link control; 802.3 - Ethernet or CSMA/CD; 802.4 - token bus LANs; 802.5 - token ring LANs; 802.6 - MANs; 802.7 - broadband LANs; 802.8 - fiber optic LANs and MANs; 802.9 - integrated (voice/data) services LANs and MANs; 802.10 - security in LANs and MANs; 802.11 - wireless LANs; 802.12 - demand priority access LANs; 802.15 - wireless PANs; and 802.16 - broadband wireless MANs. The IEEE 802 standard deals with the data link layer and the physical layer of the OSI reference model. It defines rules for cabling, signaling, and media access control, which assure interoperability between network products manufactured by different vendors. The physical layer and the data link layer which assume prominence in the 802 standard context are described below, followed by discussions on
standard,
widely
used LAN standard, and the 802.11 standard for wireless networks.
1.12.1 Physical Layer
In simple terms, the role of the physical layer is to transmit and receive data in the form of bits. The functions of the physical layer in a LAN that uses 802 standards are the same as those of the physical layer in the OSI reference model. The physical layer operates with raw bits. It is responsible for bit encoding, determining the voltage to be used for the 0/1 bit transmissions, and the time duration of each bit. The time required for the transmission of a character is dependent on the encoding scheme used and the signaling speed, which is the number of times per second the signal changes its value/voltage. The physical layer deals with the actual physical transmission medium used for communication. Hence, the implementation of the physical layer varies depending on the physical medium used. Some of the commonly used physical transmission media are twisted pair, coaxial cable, optical fiber, and radio waves (wireless). The MAC sublayer of the data link layer uses the services of the physical layer to provide its own services to the LLC sublayer of the data link layer. The MAC andLLC sublayers are explained in the following section.
1.12.2 Data Link Layer
One of the most important layers in any network model is the data link layer (DLL). As mentioned earlier, the data link layer performs several important functions such as error control, flow control, addressing, framing, and medium access control. The data link layer consists of two sublayers, the logical link control sublayer (which takes care of error control and flow control) and the medium access control sublayer (which is responsible for addressing, framing, and medium access control). Figure 1.21depicts the two sublayers. The following sections describe in detail the two sublayers.
Figure 1.21. Data link layer.
LLC provides physical medium independent data link layer services to the network layer. Below the LLC is the media access control sublayer. As mentioned previously, error control and flow control are some of the important services offered by the LLC. As defined in the IEEE 802.2 standard, the LLC adds the destination service accesspoint (DSAP) and the source service access point (SSAP) labels to each packet it receives from the network layer. The SAPs identify the applications/users involved in the data transfer. The IBM token ring architecture led to the popular IEEE 802.5 token ring standard. All nodes in a token ring network are connected in a ring. A special bit pattern, called a token, travels around the ring. In order to transmit a message, the sender node must be in receipt of the token. The token is attached to each message sent by a node. Once the sender completes its transmission, it releases the token to the next node that needs to transmit. The process continues and is used as the basis for resolving contention between multiple nodes waiting for transmission. LLC provides three types of services, namely, unacknowledged connectionless service, connection-oriented service, and acknowledged connectionless service. The unacknowledged connectionless data transfer service is commonly known as LLC type 1, or LLC1 service. A connectionless service does not require the establishment of an end-to-end connection; each packet is routed independently to the destination. Once service access points (SAPs) are enabled at the hosts, data transfer can begin. The network layer " id="pdf-obj-55-2" src="pdf-obj-55-2.jpg">
Logical Link Control Sublayer
The logical link control sublayer (LLC) is defined by the IEEE 802.2 standard. It was originally designed by IBM as a sublayer in the IBM token ring architecture. 12 LLC provides physical medium independent data link layer services to the network layer. Below the LLC is the media access control sublayer. As mentioned previously, error control and flow control are some of the important services offered by the LLC. As defined in the IEEE 802.2 standard, the LLC adds the destination service accesspoint (DSAP) and the source service access point (SSAP) labels to each packet it receives from the network layer. The SAPs identify the applications/users involved in the data transfer.
12 The IBM token ring architecture led to the popular IEEE 802.5 token ring standard. All nodes in a token ring network are connected in a ring. A special bit pattern, called a token, travels around the ring. In order to transmit a message, the sender node must be in receipt of the token. The token is attached to each message sent by a node. Once the sender completes its transmission, it releases the token to the next node that needs to transmit. The process continues and is used as the basis for resolving contention between multiple nodes waiting for transmission.
LLC provides three types of services, namely, unacknowledged connectionless service, connection-oriented service, and acknowledged connectionless service. The unacknowledged connectionless data transfer service is commonly known as LLC type 1, or LLC1 service. A connectionless service does not require the establishment of an end-to-end connection; each packet is routed independently to the destination. Once service access points (SAPs) are enabled at the hosts, data transfer can begin. The network layer
at the source node sends its packets through the SAP provided by the LLC. When the packets are received at the destination node, the LLC of the destination delivers them to the network layer through its SAP. Connection-oriented data transfer service is referred to as LLC type 2, or LLC2 service. LLC2 guarantees delivery of data using sequence numbers on packets and acknowledgments. A connection-oriented service requires the establishment of an end- to-end path (that may traverse multiple hops) before the actual data transfer can begin. All packets belonging to a session are routed along the same path, and are received at the destination in the same order as they were transmitted by the source node. LLC2 is useful when the end nodes (source and destination) do not use a transport layer protocol for reliable packet delivery. LLC3 provides acknowledged connectionless service, which is nothing but connectionless transfer of individual packets, along with acknowledgments.
The LLC plays the role of a software bus. It enables different protocols of the higher layers of the protocol stack to access different types of physical networks. For example, a host may have multiple network interface cards, such as an Ethernet or a token ring card. It is the responsibility of the LLC to forward packets belonging to different higher-layer protocols to the appropriate network interfaces. Hence, the upper layers can function independently without worrying about the type of the physical network in use.
Medium Access Control Sublayer
The medium access control sublayer (MAC) forms the lower half of the data link layer. It directly interfaces with the physical layer. It provides services such as addressing, framing, and medium access control. Unlike the LLC, these services vary with the physical medium in use. Of these, medium access control is considered to be the most important service. It is relevant to networks (such as LANs) where a single broadcast transmission channel needs to be shared by multiple competing machines. Some of the important medium access control mechanisms are discussed below.
The ALOHA protocol [7] is one of the oldest multiple access mechanisms. ALOHA had its origins in the Hawaiian Islands. It was borne out of the need for interconnecting terminals at the campuses of the University of Hawaii, which were located on different islands. It was devised in the 1970s at the University of Hawaii. The original ALOHA system is referred to as pure ALOHA. Roberts extended this
system and developed what is called the slotted ALOHA system [8], which doubled the throughput of the pure ALOHA system.
The pure ALOHA system is very simple. A radio transmitter is attached to each terminal. A user (terminal) transmits whenever data is ready for transmission. If more than one user transmits simultaneously, the transmitted packets end up getting collided and are lost. The sender would be able to detect collisions because of the feedback property of broadcast transmissions. If a sender detects that the frame transmitted by it got destroyed due to collisions, it waits for a random period of time and retransmits the frame. If the waiting time chosen is not random, the competing nodes would wait for the same period of time and so the frames transmitted by them will get collided time and again. In pure ALOHA the vulnerable period for a transmission by a node, which is the period during which no other nodes transmit so that a transmitted frame does not suffer a collision, is equal two twice the frame length/frame period. The throughput achieved in this scheme was found to be around 18 percent.
Slotted ALOHA reduces the probability of collisions by having the nodes transmit in a synchronized fashion. It requires the channel to be divided in time into discrete intervals/slots. The length of each slot is equal to the frame length used. The slot boundaries are the same for all users, that is, time is synchronized. Here a node does not transmit as soon a packet becomes ready for transmission. Instead, it waits till the beginning of the next slot interval and then transmits. If the transmitted frame suffers a collision, the node waits for a random period of time, and then retransmits the frame at the beginning of the next interval. It can be seen that the vulnerable period in the slotted ALOHA scheme is equal to the slot interval, which is equal to the frame length. This is half of that of the pure ALOHA scheme. Hence, the throughput achieved in slotted ALOHA is around 37 percent, which is almost double that of pure ALOHA.
Carrier Sense Multiple Access
The maximum achievable throughput in the ALOHA protocols is low because of wastage of bandwidth due to packet collisions. Packet collisions could be reduced by having the nodes sense for the carrier signal on the channel before they actually start transmitting. Carrier sense multiple access (CSMA) protocols are those in which nodes, before transmitting, first listen for a carrier (i.e., transmission) on the channel, and make decisions on whether or not to transmit based on the absence or presence of the carrier. Several CSMA protocols have
discussed below.
the important CSMA protocols are
The first protocol discussed here is the 1-persistent CSMA. Upon receiving a packet for transmission, a node first senses the channel to determine whether it is free. If free, the packet is transmitted immediately. Otherwise, it keeps sensing the channel till it becomes free. Once the channel becomes free, it immediately transmits the ready frame. This scheme is called 1-persistent CSMA, since the probability with which a ready node starts transmitting once it finds the channel to be idle, is 1. Propagation delay plays an important role here. Consider the case when the propagation delay is high. The sender has started transmitting the packet, but a ready node near the destination might sense the channel and find it to be idle since the already transmitted packet has not yet arrived at the destination. So this new ready node will start its own transmission which would finally result in a collision. The 1- persistent CSMA scheme performs well when the propagation delay is low.
In the non-persistent CSMA scheme, a ready node first senses the channel. If it finds the channel to be busy, it goes into a wait state. The wait period is randomly chosen. After waiting for the random time duration, it again senses the channel, and the algorithm is repeated. Consider the case when a transmission had already started, and two other nodes in the network receive packets for transmission from their higher layers. In the 1- persistent CSMA scheme, both of the ready nodes will wait till the channel becomes idle (current transmission ends) and immediately transmit. Hence, the packets suffer collisions. In the same situation, if the non-persistent scheme is followed, the two ready nodes will wait for different randomly chosen time periods before sensing the channel again. Hence, the probability of collisions is minimized.
The third scheme, p-persistent CSMA, combines the best features of the above two schemes. Here, the channel is assumed to be slotted as in the slotted ALOHA scheme. A ready node first senses the
channel. If
it finds
channel to
be busy,
it keeps sensing
channel until it can find an idle slot. Once it finds an idle slot, it transmits in the same slot with probability p, or defers the transmission to the next slot with probability q = 1 - p. If the next slot is again idle, then the node transmits in that slot or defers its transmission to the next slot with probabilities p and q, respectively. If a transmitted packet gets destroyed due to collisions, then the
node waits for an idle slot and repeats the same algorithm.
The performance of all the CSMA protocols described above is greatly dependent on the end-to-end propagation delay of the medium, which constitutes the vulnerable period when collisions are most likely to occur.
Carrier Sense Multiple Access with Collision Detection
In CSMA with collision detection (CSMA/CD), the nodes, apart from sensing the channel, are also capable of detecting collisions in the channel. In the previous schemes, suppose two neighbor nodes start transmitting simultaneously; though the packets start getting collided and garbled at the very beginning, the two nodes still transmit the entire packets. In CSMA/CD, the moment a node detects a collision on the channel, it aborts its current transmission. Hence time, and therefore bandwidth, is saved. After aborting its current transmission, the node transmits a brief jamming signal. On hearing this jamming signal, all other nodes learn about the collision; any other simultaneously transmitting node stops its transmission once it hears the jamming signal. After transmitting the jamming signal, the node waits for a random period of time, and then restarts the process. A collision can be detected by a node by comparing the power or pulse width of the received signal with that of the transmitted signal, that is, if a node finds out that what it reads back from the channel is different from what it put onto the channel, it concludes that its current packet got collided. Assume that a node starts its transmission at time t 0 and let the propagation duration (delay) be t d . Suppose the destination node starts transmitting at time t d — δ (where δ is very small). It will immediately detect a collision and stop transmitting. But, the sender node would come to know of this collision only at time 2t d — δ. In the worst case, for a sender node to determine whether its packet suffered any collision or not, it would take 2t d units of time. Therefore, the channel here is slotted as in slotted ALOHA, with the slot length equal to 2t d . The IEEE 802.3 Ethernet standard follows CSMA/CD.
1.12.3 IEEE 802.3 Standard
IEEE 802.3 is a standard for CSMA/CD networks. It is commonly referred to as the Ethernet standard. Ethernet was so named based on the word ether so as to describe an essential feature of the system, which was the transfer of bits by the physical medium to all nodes in the LAN, in a manner similar to that of the old luminiferous ether which was once thought to propagate electromagnetic waves through space. Ethernet has proven itself for many years now as a reasonably fast and relatively inexpensiveLAN technology, and
hence is one of the most popular LAN technologies in use today. IEEE 802.3 defines the physical layer and the MAC sublayer for CSMA/CD LANs. The traditional Ethernet specification supports data transmission speeds of 10 Mbps. Additional specifications such as fast Ethernet (IEEE 802.3u) and gigabit Ethernet (IEEE 802.3z) have been published to extend the performance of traditional Ethernet up to 100 Mbps and 1,000 Mbps speeds, respectively. The physical layer and MAC sublayer of the IEEE 802.3 standard are described below.
Physical Layer
Four types of cabling have been specified in the 802.3 standard. They use the following physical transmission media: thick coaxial cable, thin coaxial cable, twisted pair, and optic fiber.
The first type of cabling called 10Base2 or thin Ethernet uses thin coaxial cable. Until recently, this was the most popular 802.3 cable because it was cheap and since standard BNC (Bayonet Neil- Concelman, or sometimes British Naval Connector) connectors, rather than vampire taps, were used to form T-junctions, the quality of the connection between the computer and the coaxial cable was good. As evident from its name, it operates at 10 Mbps, uses baseband signaling, and can support LAN segments (a LAN segment is a single section of network media that connects computers/nodes) of lengths up to 200 m. Each segment can have up to 30 nodes.
The second type of cabling, which uses thick coaxial cable, is called 10Base5 cabling. It is commonly referred to as thick Ethernet. Connections are made using vampire taps, that is, the connection is made by forcing a pin halfway into the coaxial cable's core. This results in a poor connection between the computer and the coaxial cable. This system is called 10Base5 as it operates at 10 Mbps, uses baseband signaling (i.e., the signal is transmitted at its original frequency without using any modulation), and supports LAN segments of lengths up to 500 meters. It can support up to 100 nodes per segment. In the above two cabling techniques, detecting cable breaks, bad taps, or loose connections could be very tedious. These problems with coaxial cabling systems made people switch to the 10Base-T cabling system.
In the 10Base-T system each node has a separate cable connecting
a central hub. Unshielded twisted pair (UTP) is the most
commonly used cable for this system. 10Base-T supports a maximum segment length of 100 m with a maximum of 1,024 nodes per segment.
The fourth cabling technique, called 10Base-F, uses fiber optic cable. 10Base-F is costlier compared to the other three schemes. But it offers excellent noise immunity. Hence, it is the preferred cabling system when the distance between two connecting points is large. It can support segments of lengths up to 2, 000 m, with a maximum of 1,024 nodes per segment.
IEEE came up with the 802.3u standard for supporting a faster LAN with speeds of 100 Mbps. It was backward compatible with the 802.3 standard. The 802.3u standard is commonly referred to as fast Ethernet. 802.3u uses three types of physical transmission media, namely category 3 UTP (100Base-T4), category 5 UTP (100Base-TX), and optic fiber (100Base-F). The IEEE 802.3z gigabit Ethernet standard extends fast Ethernet to provide speeds of 1, 000 Mbps. It uses shielded twisted pair (STP) (1000Base- CX), UTP (1000Base-T), and optic fiber (1000Base-SX and 1000Base- LX). Table 1.3summarizes the above discussion. The reader can refer to [9] for more details on fast Ethernet and gigabit Ethernet cabling.
Table 1.3. Physical layer cabling
Table 1.3 summarizes the above discussion. The reader can refer to [ 9 ] for more details on fast Ethernet and gigabit Ethernet cabling. Table 1.3. Physical layer cabling MAC Sublayer As mentioned previously, the MAC sublayer uses CSMA/CD technology. When a node wants to transmit a frame, it listens to the channel (cable) to check whether there are any on-going transmissions. If the channel is busy, the node waits for a random duration of time and attempts again. If the channel is idle, the node starts transmitting. It is possible that two nodes end up transmitting simultaneously. In order to minimize the duration of collisions in such " id="pdf-obj-61-28" src="pdf-obj-61-28.jpg">
MAC Sublayer
As mentioned previously, the MAC sublayer uses CSMA/CD technology. When a node wants to transmit a frame, it listens to the channel (cable) to check whether there are any on-going transmissions. If the channel is busy, the node waits for a random duration of time and attempts again. If the channel is idle, the node starts transmitting. It is possible that two nodes end up transmitting simultaneously. In order to minimize the duration of collisions in such
circumstances, a transmitting node listens to what it is sending. If what it hears is different from what it is sending, then it implies that a collision has occurred. On detecting a collision, a transmitting node stops its data transmission and transmits a brief jamming signal. The jamming signal ensures that all other nodes get to know that there has been a collision. Any other simultaneously transmitting node, on hearing this jamming signal, stops its transmission. After transmitting the jamming signal, the node waits for a random period of time, and then attempts to transmit again. A receiver node operates as below. It listens to all frames on the cable. If the address of the frame is the same as that of its own address, or is the same as that of the group address of nodes of which the current node is a member, it copies the frame from the cable. Otherwise it just ignores the frame.
The 802.3 frame format is shown in Figure 1.22. The frame starts with a 7-byte preamble. It is used for synchronization purposes. Each byte in the preamble contains the bit pattern 10101010. 802.3 uses an encoding technique called Manchester encoding. Manchester encoding of the 10101010 produces a 10 MHz square wave for a duration of 5.6 μsec. This enables the receiver node's clock to synchronize with that of the sender. For more details on Manchester encoding and other encoding techniques, the reader can refer to [10]. The preamble field is followed by the destination address and source address fields. Each manufacturer of Ethernet network cards assigns a unique 48-bit Ethernet address to every Ethernet card manufactured. If all the bits are 1 in the destination address, the message is received by all computers connected to the cable, that is, broadcast. If the first bit is 1, then the rest of the bits indicate a multicast group address; the frame would be delivered only to those nodes that are part of that particular multicast group. If the first bit is 0, the rest of the bits make up a normal node address, which is unique to that node on that LAN. The length field denotes the number of bytes present in thedata field. It can vary from a minimum of 0 to a maximum of 1,500 bytes. Though a data field of length 0 bytes is perfectly legal, it would cause a problem. On detecting a collision, a transmitting node stops its transmission. Thus, collisions lead to the appearance of corrupted frames and stray bits on the cable all the time. In order to make it easier to distinguish valid frames from corrupted frames (frames that suffered collisions), 802.3 requires the length of valid frames to be at least 64 bytes from the destination address to the checksum. Another important reason for having a minimum frame length is to enable a node to detect whether the frame it transmitted was actually received without error at the destination or suffered collisions. Assume that the frame length is very small. The node would have
completed transmitting the frame before initial bits of the frame reach the destination. If the destination node starts its own transmission just before the previously transmitted frame reaches it, it results in a collision. But since the source node has already completed its transmission, it does not get to know about the collision. Therefore, to avoid such situations, the transmission duration of a frame must be equal to at least twice the round-trip time of the cable. Hence, a minimum frame length is required. For a 10 Mbps LAN with a maximum length of 2,500 m and four repeaters (802.3 specification), the minimum frame duration must be 51.2 μsec. This corresponds to 64 bytes. If the frame length is less than 64 bytes, the Pad field is used to fill up the frame to 64 bytes. The Checksumfield is used to detect if any data bits have been corrupted during transmission.
Figure 1.22. IEEE 802.3 frame format.
As described earlier, when a collision is detected, the transmitting node aborts its transmission and retries after a random duration of time. This randomization is achieved using an algorithm called thebinary exponential back-off (BEB) algorithm. The BEB algorithm adjusts what is called the contention window size by indirectly estimating the traffic load in the communication channel at individual nodes. This estimation of traffic load is done by counting the number of consecutive collisions involving the same packet transmitted by the node. The contention window size of a node is doubled every time the packet transmitted by the node experiences a collision, and is reset to its minimum value after every successful transmission. On detecting a collision, the node must wait for a random period of time before re-attempting to transmit. This waiting period is chosen based on the current value of the contention widow of the node. After k consecutive collisions, a random number between 0 and 2 k - 1 (where 2 k is the current size of the contention window) is chosen, and the node remains idle, not attempting to transmit, for that number of slots (time is assumed to be slotted,
with the slot length equal to the worst-case end-to-end propagation delay). For example, if two nodes collide, after the first collision, each node waits for 0 or 1 slot time before attempting to transmit again. If they collide again (second collision), each node picks 0, 1, 2, or 3 slots and waits for that number of slots. Thus the randomization interval grows exponentially after each collision. The randomization interval (0 to 2 k - 1) is frozen once the maximum number of slots (1,023) is reached. If a node suffers repeated collisions, it retries for a certain maximum number of times (16), and then reports failure to the node. It is up to the higher layers to take recovery action.
1.12.4 IEEE 802.11 Standard
The IEEE 802.11 standard is one of the most popular standards for wireless LANs. Wireless LANs are used for providing network services in places where it may be very difficult or too expensive to lay cabling for a wireline network. The IEEE 802.11 standard comes under the IEEE 802.x LAN standards, and specifies the physical layer and the MAC layer, adapted to the specific requirements of wireless LANs. The objective of this standard is to provide wireless connectivity to wireless devices/nodes that require rapid deployment, which may be portable, or which may be mounted on moving vehicles within a local area. The IEEE 802.11 standard also aids the regulatory bodies in standardizing access to one or more radio frequency bands for the purpose of local area communication. The interfaces offered by 802.11 to the higher layers are the same as those offered in other 802.x standards. The MAC layer should be able to work with multiple physical layers catering to multiple transmission techniques such as infrared and spread spectrum.
The basic 802.11 standard supports three different physical layers. Two of them, frequency hopping spread spectrum (FHSS) and direct sequence spread spectrum (DSSS) (which were described previously in Section 1.6.3) are based on radio transmissions, and the third is based on infrared. The physical layer provides mechanisms for sensing the wireless channel and determining whether or not it is idle. This mechanism is also called clear channel assessment (CCA).
The 802.11 standard follows carrier sense multiple access with collision avoidance (CSMA/CA), which is a random access scheme with carrier sensing and collision avoidance through random back-
off.
Because of
radio environment, it is
difficult for a transmitting node to detect packet collisions in the network. Hence, CSMA/CD is not preferred in wireless LANs.
802.11 Task Groups
Several task groups have been working on different networking aspects of wireless LANs. The various 802.11 task groups are listed below.
• 802.11: This was the first 802.11 task group. The objective of this group was to develop MAC layer and physical layer specifications for wireless connectivity for fixed, portable, and mobile nodes within a local area. The 802.11 standard was first published in 1997. • 802.11a: This group created a standard for wireless LAN operations in the 5 GHz frequency band, where data rates of up to 54 Mbps are possible. The 802.11a standard was ratified in 1999. • 802.11b: This task group created a standard for wireless LAN operations in the 2.4 GHz Industrial, Scientific, and Medical (ISM) band, which is freely available for use throughout the world. This standard is popularly referred to as Wi-Fi, standing for Wireless-Fidelity. It can offer data rates of up to 11 Mbps. This standard was ratified in 1999. • 802.11c: This group was constituted for devising standards for bridging operations. Manufacturers use this standard while developing bridges and access points. The standard was published in 1998. • 802.11d: This group's main objective is publishing definitions and requirements for enabling the operation of the 802.11 standard in countries that are not currently served by the standard. This
standard was published in 2001. • 802.11e: The main objective of this group is to define an extension of the 802.11 standard for quality of service (QoS) provisioning and service differentiation in wireless LANs. Work is in progress in this regard. • 802.11f: This group was created for developing specifications for implementing access points and distribution systems following the
802.11 standard, so that interoperability problems between devices
manufactured by different vendors do not arise. This standard was published in 2003. • 802.11g: This group was involved in extending the 802.11b standard to support high-speed transmissions of up to 54 Mbps in the 5 GHz frequency band, while maintaining backward compatibility with current 802.11b devices. The 802.11g standard was published
• 802.11h: This is supplementary to the 802.11 standard. It was developed in order for the MAC layer to comply with European regulations for 5 GHz wireless LANs, which require products to have mechanisms for transmission power control and dynamic frequency selection. The standard was published in 2003. • 802.11i: This group is working on mechanisms for enhancing security in the 802.11 standard. • 802.11j: This task group is working on mechanisms for enhancing the current 802.11 MAC physical layer protocols to additionally operate in the newly available Japanese 4.9 GHz and 5 GHz bands. • 802.11n: The objective of this group is to define standardized modifications to the 802.11 MAC and physical layers such that modes of operation that are capable of much higher throughputs at the MAClayer, with a maximum of at least 100 Mbps, can be enabled. Work on this is in progress.
A detailed description of the 802.11 standard is provided in the next chapter.
Wireless networks are computer networks that use radio frequency channels as their physical medium for communication. Each node in the network broadcasts information which can be received by all nodes within its direct transmission range. Since nodes transmit and receive over the air, they need not be physically connected to any network. Hence, such networks offer data connectivity along with user mobility.
The world's first wireless radio communication system was invented by Guglielmo Marconi in 1897. In 1901, he successfully demonstrated his wireless telegraph system to the world by transmitting radio signals across the Atlantic Ocean from England to America, covering more than 1,700 miles. Through his system, two end users could communicate by sending each other alphanumeric characters that were encoded in an analog signal. This signaled the beginning of the radio communications era.
Today, very advanced communication systems are available. Radio and television broadcasting are some of the very common applications that cannot do without wireless technologies. Voice communication has been the biggest beneficiary of the developments that took place, and are still taking place, in wireless communication technologies. Beginning in the 1960s, several
communication satellites have been launched, and today a huge proportion of long-distance voice communication takes place through the numerous satellites orbiting around the Earth.
Wireless communications is one of the fastest growing industries in the world. The wireless communications industry has several segments such as cellular telephony, wireless LANs, and satellite- based communication networks. However, the major portion of the growth in wireless industry has been due to cellular networks. The early 1980s saw the commercial deployments of the world's first mobile cellular networks. Cellular networks provide two-way simultaneous voice communication. A fixed base station serves all mobile phones in its coverage area, also called a cell. The entire service area is divided into a number of non-overlapping cells (in actual deployments the cells overlap partially), communication within each cell being coordinated by a separate base station. The first-generation (1G) cellular networks used analog signal technology. They used frequency modulation for signal transmission. The mobile phones were bulky and not very compact. The coverage provided by the 1G networks was also not good. The 1G system deployed in the United States and Canada was known as the advanced mobile phone system (AMPS). Second-generation (2G) cellular systems used digital transmission mechanisms such as TDMA and CDMA. The global system for mobile communication (GSM) used in Europe, the IS-136 (Telecommunications Industry Association Interim Standard) system used in the United States, and the personal digital communication (PDC) system used in Japan are some of the popular 2G cellular systems. The 1G and 2G systems were designed primarily for voice communication. The present generation of wireless communication networks is often called 2.5G. 2.5G is usually associated with the general packet radio services (GPRS) system. GPRS has been commercially deployed in many countries. The third-generation (3G) systems are expected to provide services such as enhanced multimedia, bandwidth up to 2 Mbps, and roaming capability throughout the world. Commercial large scale deployments of 3G systems have not yet started. It is estimated that communication service providers would shift to 3G by 2005. However, the commercial viability of such a system which requires fresh investments is still in question. Wideband code division multiple access (W-CDMA) and universal mobile telecommunications system (UMTS) are some of the important 3G standards. Standardization efforts are on for the fourth-generation (4G) wireless networks. The 4G systems are expected to provide further improvements in the services provided by 3G, such as enhanced multimedia, universal access, and portability across all types of devices. 4G services are expected to be commercially
introduced by 2010. The deployment of 4G services would transform the world truly into a global village.
Though the 2G and 2.5G systems were well-suited for voice communication, what fueled the development of newer technologies such as 3G and 4G, was the realization that wireless data services would also become very important in the next few years. Wireless data services are projected to be another huge market for the telecommunications companies. Wireless Internet forms the core of these data services. Wireless access to the Internet using mobile phones would require several modifications to the already existing Internet protocols, since such devices have limited display and input capabilities. Work is in progress for developing such protocols.
Wireless local area networks (WLANs) are another type of wireless networks that are increasingly popular today. WLANs are used for providing network services in places where it is very difficult or is too expensive to lay cabling for a wireline network. In a WLAN, a stationary node called an access point (AP) coordinates the communication taking place between nodes in the LAN. The two main standards for WLANs are the IEEE 802.11 standard and the European Telecommunications Standards Institute (ETSI) HIPERLAN standard.
Wireless personal area networks (WPANs) are short-distance wireless networks that have been specifically designed for interconnecting portable and mobile computing devices such as laptops, mobile phones, pagers, personal digital assistants (PDAs), and a number of other consumer electronic devices. Bluetooth is a popular WPAN specification and it defines how data is managed and physically carried over the WPAN. The typical range of a Bluetooth network is around 10 m. The Bluetooth Special Interest Group (SIG), comprised of several leading companies such as Ericsson, Intel, IBM, Nokia, and Toshiba, is driving the development of the Bluetooth technology and bringing it to the market. The IEEE 802.15 standard defines specifications for the media access control and physical layers for wireless devices in WPANs.
Another emerging type of wireless network is the ad hoc wireless network. An ad hoc wireless network is an autonomous system of mobile nodes connected through wireless links. It does not have any fixed infrastructure (such as base stations in cellular networks). The mobile stations/nodes in the network coordinate among themselves for communication. Hence, each node in the network, apart from being a source or destination, is also expected to route packets for other nodes in the network. Such networks find varied applications
in real-life environments such as communication in battlefields, communication among rescue personnel in disaster-affected areas, law enforcement agencies, and wireless sensor networks.
The networks mentioned so far are either infrastructure-based where fixed base stations/access points are used, or infrastructure- less where no fixed infrastructure support is available. Several new hybrid networking architectures are emerging that are a combination of the above two types of networks, taking advantage of the best features of both. Some examples of such hybrid wireless network architectures are multi-hop cellular network (MCN), integrated cellular and ad hoc relaying system (iCAR), and multi- power architecture for cellular networks (MuPAC).
The rest of the book is organized into thirteen chapters. Chapters 2- 4 address several issues concerning the design of wireless LANs and PANs, wireless WANs and MANs, and wireless Internet, respectively. Chapters 5-11 deal with all important design issues — medium access control, routing, multicasting, transport layer, security, quality of service provisioning, energy management — in ad hoc wireless networking in considerable depth. Chapters 12- 14 cover some recent related important topics.Chapter 12 is devoted to wireless sensor networks. Chapter 13 describes several hybrid wireless architectures, and routing protocols and load balancing schemes for hybrid wireless networks. Finally,Chapter 14 provides recent advances in wireless networking such as ultra wideband technology, Wi-Fi systems, optical wireless networks, and multimode 802.11.
The first half of this chapter provided a brief introduction to the fundamental aspects of wireless transmission, which include the characteristics of the wireless channel, the various modulation mechanisms, multiple access techniques, and coding and error control mechanisms. The second half of the chapter discussed the basic concepts and principles involved in computer networking. Some of the important computer network models such as the ISO OSI reference model, TCP/IP reference model, and the ATM reference model were described. This was followed by a detailed discussion of the IEEE802.3 wired LAN standard and a brief introduction to the IEEE 802.11 wireless LAN standard. A brief discussion on the advent of wireless networks was also presented.
High-frequency X-rays and Gamma rays are not normally used for wireless communication. Explain why.
2. What is multipath propagation? Explain how it affects signal quality.
3. What is inter-symbol interference? Give a mechanism that is used for overcoming problems arising due to inter-symbol interference.
4. What is the maximum data rate that can be supported on a 10 MHz noise-less channel if the channel uses eight-level digital signals?
5. Calculate the maximum achievable data rate over a 9 KHz channel whose signal to noise ratio is 20 dB.
6. What is the main difference between analog modulation and digital modulation schemes?
7. Which modulation mechanism is better, amplitude modulation or frequency modulation? Give reasons to support your answer.
8. Explain the advantages and disadvantages of the quadrature amplitude modulation schemes.
9. Show the phase changes corresponding to the bit pattern 100001 when π/4 shifted PSK is used. UseTable 1.2.
10. With respect to
modulation,
quantization error? Explain how it is caused.
11. Consider the convolutional encoder shown in Figure 1.23.
What would be the output of the encoder for an input of 1001?
Figure 1.23. Convolutional coder.
e Table 1.2 . 10. With respect to pulse code modulation, what is quantization error? Explain how it is caused. 11. Consider the convolutional encoder shown in Figure 1.23 . What would be the output of the encoder for an input of 1001? Figure 1.23. Convolutional coder. 12. Why does ATM use fixed sized cells? 13. What is meant by the vulnerable period of a transmission? What is the value of this vulnerable period for the ALOHA protocols? 14. Why is a minimum frame length required in CSMA/CD LANs that follow the IEEE 802.3 standard? Why is CSMA/CD not generally used in wireless LANs? 15. " id="pdf-obj-70-52" src="pdf-obj-70-52.jpg">
12. Why does ATM use fixed sized cells?
13. What is meant by the vulnerable period of a
transmission? What is the value of this vulnerable period for
the ALOHA protocols?
14. Why
a minimum
frame length required in CSMA/CD
LANs that follow the IEEE 802.3 standard?
Why is CSMA/CD not generally used in wireless LANs?
[1] D. J. Griffiths, Introduction to Electrodynamics, Prentice Hall PTR, New Jersey, 1998.
Pahlavan
Krishnamurthy, Principles
Networks, Prentice Hall PTR, New Jersey, 2002.
[3] T. S. Rappaport, Wireless Communications: Principles and Practice, Prentice Hall PTR, New Jersey, 2001.
[4] N. Morgan, Speech and Audio Signal Processing: Processing and Perception of Speech and Music, John Wiley & Sons, 1999.
Johannesson
Zigangirov, Fundamentals
Convolutional Coding, Wiley-IEEE Press, 2001.
[6] C. Berrou, A. Glavieux, and P. Thitimajshima, "Near Shannon Limit Error-Correcting Coding and Decoding: Turbo-Codes," Proceedings of IEEE ICC 1993, pp. 1064-1070, May 1993.
[7] N. Abramson, "The Aloha System – Another Alternative for Computer Communications,"Proceedings of AFIPS Fall Joint Computer Conference 1970, vol. 37, pp. 281-285, November 1970.
[8] L. Roberts, "ALOHA Packet System with and without Slots and Capture," Computer Communications Review, vol. 5, pp. 28-42, April
[9] Andrew S. Tanenbaum, Computer Networks, Prentice Hall PTR, New Jersey, 2002.
[10] W. Stallings, Data and Computer Communications, Prentice Hall PTR, New Jersey, 1999.
Channel Access Method
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