CSN200 Introduction to Telecommunications, Winter 2000 Lecture-18 Analog Transmission and Modulation Analog Transmission of Digital Data (Broadband Transmission): There are many occasions when data need to be transmitted over a voice communication network. Many people use a computer to connect to their Internet Service Provider via the telephone lines to transmit and receive data, check their email, surf the Web, etc. Telephone networks were originally built for human speech, not for computer data and is unsuitable for digital communication in its raw form. The transmission impairments and the bandwidth limitation makes digital signalling (have a wide spectrum) unsuitable for transmission over the local loop (designed for analog transmission). Transmission Impairments: Transmission lines suffer from three major problems: Attenuation - is the loss of energy as the signal propagates outward. The amount of energy lost depends on the frequency. Each Fourier component is attenuated by a different amount. Delay distortion - It is caused by the fact that different Fourier components travel at different speeds. For digital data, fast components from one bit may catch up and overtake slow components from the bit ahead, mixing the two bits and increasing the probability of incorrect reception. Noise - unwanted energy from the sources other than the transmitter. Thermal noise, cross talk, impulsive noise. Problems of using Voice Channels for Digital Transmission A digital signal is comprised of a number of signals. Specifically, the signal is represented as follows, signal = f + f3 + f5 +f7 +f9 +f11 +f13 ....f(infinity) This means a digital signal has a base frequency, plus another at three times the base frequency, plus another at five times the base frequency etc. f3 is called the third harmonic, f5 the fifth harmonic and so on. The third harmonic is one third of the amplitude of the base frequency (also called the fundamental frequency), the fifth harmonic is one fifth the amplitude of the fundamental and so on. In order to send a digital signal across a voice channel, the bandwidth of the channel must allow the fundamental plus third and fifth harmonic to pass without affecting them too much. As can be seen, this is what such a signal looks like, and is the minimum required to be correctly detected as a digital signal by the receiver. 81900005 Page 1 (14) CSN200 Introduction to Telecommunications, Winter 2000 Lecture-18 Analog Transmission and Modulation Lets consider sending a 2400bps binary digital signal down a voice channel rated with a bandwidth of 3.1KHz. The base frequency of the digital signal is 1200Hz (it is always half the bit rate), so the fundamental frequency will pass through the channel relatively unaltered. The third harmonic is 3600Hz, which will suffer attenuation and arrive severely altered (if at all). The fifth harmonic has no chance of passing the channel. In this case it can be seen that only the base frequency will arrive at the end of the channel. This means the receiver will not be able to reconstruct the digital signal properly, as it will require f3 and f5 for proper reconstruction. This results in errors in the detection process by the receiver. The Public Switched Telephone Network has, for a long time, provided users with dial up telephone connections on demand. Each connection has supported analogue speech in the voice range of 300Hz to 3400Hz. The signals provided by computers are digital, and are not designed to travel across the dial up telephone connections. In order for digital signals to be sent across a telephone connection, they must be converted to analogue voice tones within the frequency range 300Hz to 3400Hz (this is done by a modem). To get around the problems associated with the dc signalling or digital signalling, especially on telephone lines, analog signalling is used. A continuous tone in the range 1000Hz to 2000Hz, called a sine wave carrier is used. Its amplitude, frequency, or phase can be modulated (changed with the information) to transmit information. Frequency = Number of waves or Cycles per Sec = 1/(Time Period in Sec) = (1/T) Cycles per Sec or Hertz Every sound wave has two parts, half above the zero point (positive), and half below (negative) and three important characteristics. The height of the wave is called amplitude. The length of the sound wave is expressed as the number of waves per second or frequency, 81900005 Page 2 (14) CSN200 Introduction to Telecommunications, Winter 2000 Lecture-18 Analog Transmission and Modulation expressed in Hertz (Hz). The phase is the direction in which the wave begins. Bandwidth refers to a range of frequencies. Human hearing ranges from about 20 Hz to about 14,000 Hz (some up to 20,000 Hz). Human voice ranges from 100 Hz to about 3,100 Hz. The bandwidth of a voice signal is 3000 Hz (3 kHz). The bandwidth of a voice grade telephone circuit is 0 to 4000 Hz or 4000 Hz (4 kHz). Guardbands prevent data transmissions from interfering with other transmission when these circuits are multiplexed using FDM. It is important to note that the limit on bandwidth is imposed by the equipment used in the telephone network. The actual capacity of bandwidth of the wires in the local loop depends on what exact types of wires were installed, and the number of kilometers in the local loop. Actual bandwidth of local loops in North America varies from 300 kHz to 1.5 MHz depending on distance. Longer the loop, lower the bandwidth. Modulation: Modulation is the technique that modifies the form of an electrical signal so the signal can carry information on a communications media. Types of Modulation: Analog Modulation (analog encoding of digital information) Digital Modulation (digital encoding of analog information) Analog Modulation: Modifies the form of an analog electrical signal so the signal can carry information. The signal that does the carrying is the carrier wave. Modulation changes the shape of the carrier wave (a sine wave) to transmit 0s and 1s. There are three fundamental methods of analog modulation of an analog signal: Amplitude Modulation (AM) Frequency Modulation (FM) Phase Modulation(PM) These three fundamental methods of analog modulation are called differently when the information is a digital signal: Amplitude Shift Keying (ASK) Frequency Shift Keying (FSK) Phase Shift Keying (PSK) All analog modulation techniques employ a carrier signal. A carrier signal is a single frequency that is used to carry the intelligence (data). For digital data, the intelligence is either a 1 or 0. When we modulate the carrier , we are changing its characteristics to correspond to either a 1 or 0. 81900005 Page 3 (14) CSN200 Introduction to Telecommunications, Winter 2000 Lecture-18 Analog Transmission and Modulation Amplitude Modulation (AM) varies the amplitude or height of the signal to carry the information. (Fig 4-9) One amplitude level is used to represent a 1 and another for a 0. This is also called Amplitude Shift Keying or ASK. Not commonly used because it is too susceptible to noise, causing more errors in transmission. 81900005 Page 4 (14) CSN200 Introduction to Telecommunications, Winter 2000 Lecture-18 Analog Transmission and Modulation Frequency Modulation (FM) varies the frequency of the signal to send data. (Fig 4-10) One frequency represents 1s and another frequency represents 0s. For example, a 1200 Hz tone may send a 0 bit while a 2400 Hz tone sends a 1 bit. This is also called Frequency Shift Keying or FSK. It is not as widely used now as Phase Modulation is more efficient. Phase Modulation (PM) varies the phase of the carrier signal to send data. (Fig 4-11) Phase refers to the relationship of one wave to another. One cycle of a sine wave has 360 so if a sine wave has a phase of 180 every time a zero bit was sent and a phase of 0 every time a one bit was sent, it would appear as in Fig. 5-10. A 180 phase shift of a signal is one half a full cycle. One phase of the carrier sinewave is used to represent 1’s and another phase 0’s. PSK (Phase Shift Keying) (Fig 4-11) Uses 180 phase shift of the carrier every time there is a change from a 1 to a 0 or a 0 to a 1. DPSK (Differential Phase Shift Keying) The phase of the carrier shifts 180 every time a 1 is transmitted. Phase Modulation is a very common form of modulation used in high speed modems today. 81900005 Page 5 (14) CSN200 Introduction to Telecommunications, Winter 2000 Lecture-18 Analog Transmission and Modulation Modulator: The circuit or device does the process of modulation is called the modulator. Demodulation: Demodulation is the process of converting back the modulated signal to its original information (in our case the digital signal, 1s and 0s). Demodulator: The circuit or device does the process of demodulation is called the demodulator. MODEMS: Modem is an acronym for Modulator/demodulator, and takes digital electrical pulses from a computer, terminal, or microcomputer and converts them into a continuous analog signal, for transmission over an analog voice grade circuit. It then re-converts the analog signal to its original digital format. Most modems accept commands from a microcomputer keyboard. 81900005 Page 6 (14) CSN200 Introduction to Telecommunications, Winter 2000 Lecture-18 Analog Transmission and Modulation Bit Rate versus Baud Rate versus Symbol Rate: Two different terms were used to describe the rate at which data is transmitted: Bit Rate - Bits per second (bps) and Baud Rate - Baud per second (Not known as bps, but confusion may happen). Bit Rate: The rate at which information bits are transmitted over a communication path. Normally expressed in bits per second, (bps). Baud Rate: Is a unit of signalling speed, indicating the number of signal events or signal changes per second. A signal change occurs when the amplitude, frequency or phase of a carrier signal changes. In the early days of modems, signalling units (tokens) were called "baud" in honour of the French inventor Emile Baudot who, in 1875, invented a 5 bit code for representing the alphabet. Each 5 bits were a token communicating a letter of the alphabet or a control code. If each signal event represents only one bit, then bits per second and baud are identical. When each signal event represents more than one bit of information, then bps no longer equals baud. The terms bit rate (the number of bits per second) and baud rate are used incorrectly much of the time. They are not the same. A bit is a unit of information, a baud is a unit of signaling speed, the number of times a signal on a communications circuit changes. To remove the confusion between the bps and baud, ITU-T now recommends the term baud rate be replaced by the term Symbol rate (symbols per second). The bit rate and the symbol rate (or baud rate) are the same only when one bit is sent on each symbol. If we use QAM or TCM, the bit rate would be four to eight times the baud rate. Multibit Encoding of Information: Before high speed modems (above 2400 bps) were possible a technique had to be developed to encode bits so the bit rate could be larger than the signalling rate or baud. All high speed modems use multibit encoding techniques so more than one bit of information is sent each time the carrier changes state. If each change of the carrier signal sends more than one bit of information then bps > baud For example, if we have 4 possible amplitude levels for the carrier sine wave then each level can be used to represent 2 bits of information, (Fig 4-12) Sending Multiple Bits Simultaneously: Each of the three modulation techniques can be refined to send more than one bit at a time. It is possible 81900005 Page 7 (14) CSN200 Introduction to Telecommunications, Winter 2000 Lecture-18 Analog Transmission and Modulation to send two bits on one wave by defining four different amplitudes. Amplitude 1 volt 2 volts 3 volts 4 volts Bit value 00 01 10 11 For 2 bits per baud, we need 22 or 4 unique states of the carrier. If the carrier sinewave amplitude was changing 1000 times per second but each amplitude represented 2 bits of information, then baud = 1000 Hz bps = 2000 bps This technique could be further refined to send three bits at the same time by defining 8 different amplitude levels or four bits by defining 16, etc. The same approach can be used for frequency and phase modulation. For 3 bits per baud we need 23 or 8 states. For 4 bits per baud we need 24 or 16 states. In general to transmit n bits per baud you require 2n states. baud = bps / (# of bits/baud) or bps = baud * (# of bits/baud) A 14,400 bps modem may use 2400 baud with 6 bits/baud encoding. 26 = 64 so there must be 64 unique states of the carrier for each of the 64 combinations of 6 bits to be identified. 81900005 Page 8 (14) CSN200 Introduction to Telecommunications, Winter 2000 Lecture-18 Analog Transmission and Modulation It is not practical to use amplitude modulation with 64 different amplitude levels, or frequency modulation with 64 different frequencies. As the number of different amplitudes, frequencies or phases becomes larger, it becomes very difficult for a receiving modem to differentiate among them correctly. Errors are made in interpreting the signal state. In practice, the maximum number of bits that can be sent with any one of these techniques is about five bits. The solution is to combine modulation techniques. One popular technique is quadrature amplitude modulation (QAM) involves splitting the signal into eight different phases, and two different amplitude for a total of 16 different possible values. QAM: Quadrature Amplitude Modulation involves modulating the carrier with 8 different phases (3 bits) and two different amplitudes (1 bit) for 16 possible states of the carrier. So one signal change can represent 4 bits (4 bits/baud). This is used in 9600 bps modems with a 2400 baud rate. The problem with high speed modulation techniques such as QAM is that they are more sensitive to imperfections in the communications circuit. Trellis coded modulation (TCM) is an enhancement of QAM that combines a noise reduction technique. Various forms of TCM encode 6,7, 8 or 10 bits per baud. At 8 bits/baud a modem using a 2400 baud signaling rate would transmit at 19,200 bps and require 256 unique states of the carrier. Capacity of a Voice Circuit: The capacity of a voice circuit (the maximum data rate) is the fastest rate at which you can send your data over the circuit. The maximum symbol rate in any circuit depends upon the bandwidth available and the signal-to-noise ratio. Voice grade lines provide a bandwidth of 3000 (=3300 - 300) Hz and a signal-to-noise of 30dB. Maximum Symbol (Baud) Rate: The maximum symbol rate in any circuit depends upon the bandwidth available and the amount of noise in the circuit. As a rule, the maximum baud rate is usually the same as the bandwidth. The maximum baud rate in a dial-up voice circuit (with a bandwidth) is usually 3000 bauds. Overtime, modem designers began to realize that the telephone network was getting better and that more bandwidth was available. The baud rate has been extended to 3429 in newer v.34 and v.34+ modems. (Uses a carrier of 1959 Hz, this implies that less than one cycle of the carrier is being used for each signaling event. Needs a bandwidth from 244 [=1959-3429/2] Hz to 3674 [=1959 +3429/2] Hz.) V.34+ uses a symbol rate of 3429 and the number of bit per symbol is 9.8 resulting a maximum data rate of 33600 bps. Which is the maximum data rate limit according to shanon' theorem. 81900005 Page 9 (14) CSN200 Introduction to Telecommunications, Winter 2000 Lecture-18 Analog Transmission and Modulation Shanon's Theorem: There is a limit to the amount of information that could be communicated over a band limited channel in the presence of noise. This is known as the channel capacity (maximum number of bits per sec). One of the most fundamental laws used in telecommunications is Shanon's law. In 1948 Shanon proved that the maximum data rate of a noisy channel whose bandwidth is B Hz, and whose signal-to-noise ratio is S/N, is given by Channel Capacity = B log2 (1+S/N) bps where: bps = bits per second B = channel bandwidth S/N = signal to noise power ratio For a bandwidth of 3.1 kHz and a signal to noise ratio of 30 dB (a ratio of 1000/1), the maximum data rate is 31 kbps. Channel Capacity = B log2 (1+S/N) bps = 3100 log2 (1+1000) bps = 3100 log2 (1001) bps = 3100 log2(29.967) bps = 3100 x 9.967 bps = 30,898 bps = 31 kbps A telephone circuit has a signal-to-noise ratio of about 30dB and a newer bandwidth of about 3429 Hz. A signal-to-noise ratio of 30 dB = a ratio of 1000 to 1. log2 (1+1000) is roughly equal to 10. So, the bps = 3429 x 10 = 34290 = 33.6 kbps. Todays Modems reached that limit of data rate on a old telephone line (it is not the wire itself, it includes the circuitry at the local telephone offices). Nyquist Theorem (Noiseless channel): Max. data rate = 2 B log2 M bps where B is the bandwidth, M is signal levels Shannon Theorem (Random noise): Max. data rate = B log2 ( 1 + S/N ) bps where B is the bandwidth, S/N is the signal-to-noise ratio. Signal-to-noise ratio (SNR): SNR = 10 log10 S/N decibels(dB) Nyquist Sampling Theorem: If a signal f(t) is sampled at regular intervals of time and at a rate higher than twice the highest significant signal frequency, then the samples contain all the information of the original signal. The function f(t) may be reconstructed from these samples by the use of a low-pass filter. 81900005 Page 10 (14) CSN200 Introduction to Telecommunications, Winter 2000 Lecture-18 Analog Transmission and Modulation QAM (Quadrature Amplitude Modulation): QAM operates by modulating a carrier sine wave signal in both amplitude and phase. Each unique combination of amplitude and phase is known as a "symbol". And when you talk of amplitude and phase, you immediately think of vectors. It turns out that modem designers use the concept of vectors to visualize the symbols being transmitted. For example, the figure below shows one representative symbol, a combination of amplitude and phase. Figure: Graph of a single symbol, showing the amplitude and phase As you try to represent a large number of symbols, however, the graph can get fairly busy. Additionally, some symbols will have the same phase angle but different values of amplitude. These two vectors will lay atop one another and it will be hard to see the one with the lesser amplitude. Because of this, modem designers graph only the end points of the vectors, using a dot to represent the end point location of the vector. When a large number of symbols are drawn on a graph, the resulting figure is known as a "constellation" because it begins to look like a star map. A simple constellation of only four symbols is shown in the figure below. Transmitted Symbol Received Symbol 01 00 Decision Region 10 Error Vector 11 Figure: A four point constellation showing the values assigned to each symbol and the receiver decision region 81900005 Page 11 (14) CSN200 Introduction to Telecommunications, Winter 2000 Lecture-18 Analog Transmission and Modulation This figure also illustrates the decision region, another aspect of the transmission of a symbol from the transmitter to the receiver. Although the transmitter sends the symbol accurately, the transmission channel can modify the symbol in ways that cause the symbol received by the receiver to be different than the one transmitted. For example, suppose a noise transient occurs during the transmission of the symbol. This may cause the amplitude to be larger or smaller than what was transmitted, which makes the constellation point move compared to what was transmitted. This difference is called the "error vector". However, as long as the received constellation point is within the decision region, it will be interpreted as the correct symbol and the proper bits will be communicated. As the modem uses more and more symbol points, the decision region shrinks, leading to a higher error rate in the presence of noise. Thus, a modem designer can’t just keep increasing the number of symbols to gain a higher data rate – at some point you can no longer communicate data at an acceptable error rate. To a large degree, this is why your V.34 modem doesn’t connect at 33.6 Kbps very often. To communicate at 33.6 Kbps with V.34, the modem has to use a very large number of symbol points (1,664 symbols, 10.7 bits per symbol). Since the transmit power is limited, the overall size of the constellation can only be so large. This means that the symbol points must be very close together, making it very difficult for the receiver to accurately distinguish which symbol was sent. To operate, therefore, the modem must "downrate" to a constellation which uses a smaller number of symbol points. In reality, there are other types of channel impairments, such as available bandwidth, which also limit operation at these higher rates, but noise, both Gaussian and quantization, plays a large part. Phase Reference Signal Quadrature Amplitude Modulation produces 8 signal states from a combination of 4 angles and 2 amplitude levels. Similarly the number of signal states can be increased to 16 signal states from a combination of 4 angles and 4 amplitude levels. And so on upto a practical limit imposed by noise and detection errors. 81900005 Page 12 (14) CSN200 Introduction to Telecommunications, Winter 2000 Lecture-18 Analog Transmission and Modulation TCM (Trellis coded modulation): TCM is an enhancement of QAM that combines a noise reduction technique. Trellis Coded Modulation (TCM) encoding is a forward, error-correcting technique that allows a modem to more readily recover data corrupted by noise and other transmission impairments on the telephone line. Trellis encoding allows error detection and correction without retransmission of the data. This is different to and separate from the error-control methods of the V.42 LAPM and MNP protocols. Trellis encoding is available when the modem is operating in the ITU-T V.32 or V.32bis mode and connected to another V.32 or V.32bis modem. In the case of V.32bis, TCM is unconditionally active at 7200 bps and above and cannot be disabled. In the case of V.32, TCM can be activated only at 9600 bps. The factory default is for TCM to be enabled. It can be disabled with the AT&U1 command. LAPM – Link Access Protocol for Modems. A type of error control used by modems. It is included in the V.42 and V.42bis protocol. This trellis coding technique developed about the time of V.32, generally credited to Gottfried Ungerboeck at IBM’s Zurich Research Laboratory. Trellis codes explanation: If you look back at the above figure, you’ll see how the decision region surrounds the transmitted symbol point. As more and more points are defined, the decision region shrinks until a small amount of noise can move the symbol point into another symbol’s decision region, causing an incorrect decode by the receiver. Suppose, however, that you had a way to subset the constellation points, removing half or more of the points to maximize the decision region around each remaining point. This would allow the receiver to more accurately decode which symbol point was sent, in the presence of noise. This is essentially what’s done with trellis (frame or lattice) codes. A number of constellations are created, each with the maximum sized decision region around each symbol. In trellis code, the constellation is divided into 2, 4, 8, ... subsets with size Total Points/2, Total Points/4, Total Points/8, etc. That results in larger distances between signal points in the new subset constellations. The following figure illustrates the partitioning of constellation corresponds to 16-QAM. The distances between signal points increases from successive partitioning of the original constellation. To send n bits/symbol with quadrature modulation, we start with a constellation of 2n+1 signal points. Two incoming bits per symbol enter a 2/3 convolutional encoder; the resulting three codded bits per symbol determine the selection of a particular subset. The remaining uncoded data bits determine which particular point from the selected subset is to be signaled. 81900005 Page 13 (14) CSN200 Introduction to Telecommunications, Winter 2000 Lecture-18 Analog Transmission and Modulation Since each symbol point represents a unique string of bits, and if the receiver knew what constellation was being used, it could do a better job of decoding what symbol had actually been sent. The problem, of course, is to tell the receiver what subset constellation to use. [ This is accomplished in an "after the fact" manner in the receiver by allowing only certain transitions from one state to another. Suppose that we create four different subset constellations. For each starting state, we then establish valid transitions to only two other states and use an extra bit per symbol to force legal transitions, only, through a convolutional encoder. The receiver decodes the symbols as it normally would, assuming only a single constellation. The receiver then does a "traceback" and examines the sequence of state transitions. If an invalid state transition occurred, the receiver then goes back and examines the error vectors for the symbols, computing the distance from the actually received symbol point to valid symbol points (Viterbi decoding). The closest valid symbol point, which also creates a legal sequence of state transitions, is selected. Nothing is free, however. To accomplish this bit of magic, an extra bit per symbol (for two dimensional codes) or every two symbols (for four dimensional codes) is required. Sending an extra bit per symbol requires that we double the number of symbols and we lose 3 dB of signal to noise ratio (SNR) performance by doing this. The trellis code, however, provides about 6 dB of coding gain so we achieve about 3 dB better SNR performance. ] Ref: 81900005 Text Book: Chapter-4, Fitzgerald Communication System -by Simon Haykin Rockwell: www.nb.rockwell.com/K56flex/whitepapers/k56whitepaper.html Netexpress: http://www.netexpress.net/techsupp/flex.html v90: http://www.v90.com/ AT Commands: http://www.internetpro.net/~phred/files/refman.html Brian Brown: http://www.cit.ac.nz/smac/dc100www/dc_006.htm#analogue Chen: http://www.tec.hkr.se\chen\courses\index.html Page 14 (14)
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