Showing papers in "IEEE Transactions on Communications in 1975"

Packet Switching in Radio Channels: Part I--Carrier Sense Multiple-Access Modes and Their Throughput-Delay Characteristics

Abstract: Radio communication is considered as a method for providing remote terminal access to computers. Digital byte streams from each terminal are partitioned into packets (blocks) and transmitted in a burst mode over a shared radio channel. When many terminals operate in this fashion, transmissions may conflict with and destroy each other. A means for controlling this is for the terminal to sense the presence of other transmissions; this leads to a new method for multiplexing in a packet radio environment: carrier sense multiple access (CSMA). Two protocols are described for CSMA and their throughput-delay characteristics are given. These results show the large advantage CSMA provides as compared to the random ALOHA access modes.

Packet Switching in Radio Channels: Part II--The Hidden Terminal Problem in Carrier Sense Multiple-Access and the Busy-Tone Solution

Abstract: We consider a population of terminals communicating with a central station over a packet-switched multiple-access radio channel. The performance of carrier sense multiple access (CSMA) [1] used as a method for multiplexing these terminals is highly dependent on the ability of each terminal to sense the carrier of any other transmission on the channel. Many situations exist in which some terminals are "hidden" from each other (either because they are out-of-sight or out-of-range). In this paper we show that the existence of hidden terminals significantly degrades the performance of CSMA. Furthermore, we introduce and analyze the busy-tone multiple-access (BTMA) mode as a natural extension of CSMA to eliminate the hidden-terminal problem. Numerical results giving the bandwidth utilization and packet delays are shown, illustrating that BTMA with hidden terminals performs almost as well as CSMA without hidden terminals.

A Method of Self-Recovering Equalization for Multilevel Amplitude-Modulation Systems

Abstract: A self-recovering equalization algorithm, which is employed in multilevel amplitude-modulated data transmission, is presented. Such a self-recovering equalizer has been required when time-division multiplexed (TDM) voice or picturephone PCM signals must be transmitted over the existing frequency-division multiplexed (FDM) transmission channel. The present self-recovering equalizer is quite simple, as is a conventional binary equalizer. The convergence processes of the present self-recovering equalizer are shown by computer simulation. Some theoretical considerations on this convergence process are also added.

Packet Switching in a Multiaccess Broadcast Channel: Performance Evaluation

Abstract: In this paper, the rationale and some advantages for multiaccess broadcast packet communication using satellite and ground radio channels are discussed. A mathematical model is formulated for a "slotted ALOHA" random access system. Using this model, a theory is put forth which gives a coherent qualitative interpretation of the system stability behavior which leads to the definition of a stability measure. Quantitative estimates for the relative instability of unstable channels are obtained. Numerical results are shown illustrating the trading relations among channel stability, throughput, and delay. These results provide tools for the performance evaluation and design of an uncontrolled slotted ALOHA system. Adaptive channel control schemes are studied in a companion paper.

Partial-Response Signaling

Abstract: This paper presents a unified study of partial-response signaling (PRS) systems and extends previous work on the comparison of PRS schemes. A PRS system model is introduced which enables the investigation of PRS schemes from the viewpoint of spectral properties such as bandwidth, nulls, and continuity of derivatives. Several desirable properties of PRS systems and their relation to system functions are indicated and a number of useful schemes, some of them not previously analyzed, are presented. These systems are then compared using as figures of merit speed tolerance, minimum eye width, and signal-to-noise ratio (SNR) degradation over ideal binary transmission. A new definition of speed tolerance, which takes into account multilevel outputs and the effect of sampling time, is introduced and used in the calculation of speedtolerance figures. It is shown that eye width, a performance measure that has not been used previously in comparing PRS systems, can be calculated analytically in many cases. Exact values as well as bounds on the SNR degradation for the systems under consideration are presented. The effect of precoding on system performance is also analyzed.

Bistable Behavior of ALOHA-Type Systems

Abstract: Packet switching has found widespread application in computer communications because of its ability to efficiently handle high ratios of peak-to-average data rate. Packet radio is the application of packet switching techniques to radio channels. The resultant multiple-access problem requires novel approaches. Such approaches have been developed by others and have primarily been analyzed in steady-state behavior. This paper demonstrates and analyzes an important aspect of the dynamic characteristics of packet radio, namely, that of bistable behavior. That is, the system possesses two statistically stable equilibrium points, one in a desirable low-delay region, and the other in an undesirable high-delay region. Since the stability is only statistical in nature, the system oscillates between these two points. Even if the resultant steady-state behavior is very poor, this dynamic analysis frequently shows that system performance will be acceptable. This is due to quiet periods (such as at night) which allow the system to recover.

Telecommunications and Organizational Decentralization

Abstract: In recent years, several phenomena have caused significant pressures on the traditional, centralized urban structure. These phenomena include urban sprawl, separation of business and residential areas and concomitant dependence on transportation, the absence of effective or widespread mass transit, and declining oil reserves with rising energy costs. These conditions have made decentralization more attractive to many large organizations currently located in the central business districts (CBD's) of major urban areas. The increasing availability of sophisticated communications and computer technologies may encourage the continued growth and future decentralization of "information industries," thereby producing major urban changes. The telecommunicationsaugmented decentralization of a traditional, centralized organization to a diffused one with an intraorganizational telecommunications network is described. The key factors in this process are discussed: 1) the ability of new telecommunications and computer technologies to maintain or increase productivity for routine clerical and management functions, 2) their availability, and 3) their costs relative to urban transportation systems. Telecommunications-augmented decentralization can have significant impacts on transportation, telecommunications, labor, and land-use policies; specific areas of impact are discussed.

Packet Switching in a Multiaccess Broadcast Channel: Dynamic Control Procedures

Abstract: In a companion paper [1], the rationale for multiaccess broadcast packet communication using satellite and ground radio channels has been discussed. Analytic tools for the performance evaluation and design of uncontrolled slotted ALOHA systems have been presented. In this paper, a Markovian decision model is formulated for the dynamic control of unstable slotted ALOHA systems and optimum decision rules are found. Numerical results on the performance of controlled channels are shown for three specific dynamic channel control procedures. Several practical control schemes are also proposed and their performance compared through simulation. These dynamic control procedures have been found to be not only capable of preventing channel saturation for unstable channels but also capable of achieving a throughput-delay channel performance close to the theoretical optimum.

Correlation Bandwidth and Delay Spread Multipath Propagation Statistics for 910-MHz Urban Mobile Radio Channels

Abstract: Distributions of delay spread and correlation bandwidth at 0.9 and 0.5 correlation for Gaussian wide-sense stationary uncorrelated scattering (GWSSUS) channels associated with 100 small-scale areas at different locations within a 2 × 2.5 km region of New York City are presented. For delay spread the maximum value observed was 3\frac{1}{2};\mu s and l0 percent of the areas exceeded 2\frac{1}{2}\mu s; for correlation bandwidth at 0.9 correlation the minimum was 20 kHz and 10 percent of the areas were less than 30 kHz; for correlation bandwidth at 0.5 correlation the minimum was 55 kHz and 10 percent of the areas were less than 130 kHz. The region is representative of the heavily built-up areas of many large cities in the United States.

A Simple Method for Reliability Evaluation of a Communication System

Abstract: Very few techniques exist for reliability evaluation of communication systems where links as well as nodes have certain probability of failure. This correspondence describes a technique by which the reliability expression for such a system can be conveniently derived. It is also shown that using the concept of this correspondence, it is possible to extend all the existing reliability-evaluation algorithms to communication systems with little effort.

Improving Automatic Repeat-Request (ARQ) Performance on Satellite Channels Under High Error Rate Conditions

Abstract: Large round trip delay associated with satellite channels reduces the throughput for automatic repeat-request (ARQ) system of error control rather drastically under high error rate conditians. Ground segments that usually accompany satellite circuits at both ends introduce bursts of errors, during which block error rates tend to be quite high, bringing down the throughput to very low values. In this paper, some schemes that modify the currently known variations of ARQ (Stop-and-Wait and continuous systems) are suggested with a view to obtaining higher throughput under high block error rate conditions. Specifically, the modified Go-Back- N system appears to be quite attractive, as it gives substantial improvement with little additional complexity in system implementation.

Image Coding Via a Nearest Neighbors Image Model

Abstract: A two-dimensional interpolative model for coding of images is presented. It is shown that this model leads to three different coding algorithms, each algorithm defining a specific transmitter-receiver architecture. Simulation examples on a 255 × 255 image are given. Computational aspects and possible extensions are discussed.

A Robust Adaptive Quantizer

Abstract: A popular method of time-varying quantization involves a multlplicative change in quantizer overload point after each sample time. Although this technique produces an average loading factor that is invariant with rms input, it is vulnerable to transmission errors. Here, we introduce a related quantizer which has the advantage of disipating the effects of transmission errors at any desired rate. On the other hand, adaptation is not perfect. The loading factor now depends on rms input and in designing the quantizer one must compromise between the goal of constant loading factor and the goal of rapid error dissipation. We provide a design method and apply it to the modification of a quantizer currently used for speech. The new quantizer dissipates transmission errors with a 5 ms time constant while maintaining accurate adaptation over a 40 dB input range.

Odd-Bit Quadrature Amplitude-Shift Keying

Abstract: The selection of a particular signal set design for a bandwidth-constrained multiple amplitude-and-phase-amplitude-and-phase-shift-keyed (MAPSK) communication system with a linear additive Gaussian noise channel is influenced by a number of factors, such as average and/or peak signal-to-noise ratio for a given error probability, dynamic range of signal amplitudes, simplicity of generation and detection, and number of bit errors per adjacent symbol error (Gray code properties). This paper compares two possible quadrature amplitudes-shift-keyed (QASK) signal sets when the number of bits per symbol is odd (for the even-bit case, the square array is the only viable QASK choice). The "symmetric" QASK version outperforms the "rectangular" QASK set at a very modest implmentation penalty.

Measuring the Speed of Moving Objects from Television Signals

Abstract: Very simple techniques for estimating the speed of a moving object from a television signal are described. They assume that frame storage is available. One technique requires little more than two threshold circuits and two counters. The accuracy of one technique is better than ± 10 percent, quite sufficient to enable the measure to be used as a control signal for the efficient coding of television type signals.

An Optimum Linear Receiver for Multiple Channel Digital Transmission Systems

Abstract: An optimum linear receiver for multiple channel digital transmission systems is developed for the minimum P e and for the zero-forcing criterion. A multidimensional Nyquist criterion is defined together with a theorem on the optimality of a finite lenght multiple tapped delay line. Furthermore an algorithm is given to calculate the tap settings of this multiple tapped delay line. This algorithm simplifies in those cases where the noise is so small that it can be neglected. Finally as an example the transmission of binary data over a cable, consisting of four identical wires, symmetrically situated inside a cylindrical shield, is considered.

The Residual-Excited Linear Prediction Vocoder with Transmission Rate Below 9.6 kbits/s

Abstract: In this paper we present a new vocoder called the residual-excited linear prediction (RELP) vocoder. The concept of the RELP vocoder combines the advantages of linear predictive coding (LPC) and voice-excited vocoding. In the RELP system, vocal tract modeling is done by the LPC technique, and the LPC residual signal is used as the excitation signal. After low-pass filtering the residual signal is coded by adaptive delta modulation and is spectrally flattened before being fed in the LPC synthesizer. The range of the transmission rate is typically between 6 and 9.6 kbits/s; the synthetic speech in this range is quite good. As the transmission rate is lowered, the synthetic speech quality degrades very gradually. Since no pitch extraction is required, the vocoder is robust in any operating environment.

Congestion Control in Store and Forward Tandem Links

Abstract: This paper discusses the problem of congestion in message-switched data communication networks. This condition occurs when more traffic enters a network than can reasonably be served. Two types of techniques employed to control congestion in already existing national networks are identified. A queueing model which can be used to analyze and compare these control schemes for a simple tandem link network is developed. Finally, analytic results are presented which describe the behavior of the simple network using the two control schemes, and which should be useful in the design of more general networks as well.

On the Control, Stability, and Waiting Time in a Slotted ALOHA Random-Access System

Abstract: This concise paper explores some of the boundaries in performance of slotted ALOHA systems by analyzing a simple and almost optimal centrally supervised control. Our control results in a very simple Markov chain model and allows an examination of stability, conditional waiting time distribution of transmitting terminals, and many other system measures. The key to the simplicity is to have a probability of successful packet transmission that is independent of the number of transmitting terminals. Regarding stability, recent papers have shown that a slotted ALOHA system with an infinite population of terminals producing packets at a Poisson rate λ/slot becomes saturated with retransmissions and breaks down. Here we define a stable ALOHA system as one that clears out the blocked packets in a finite time, and has only a finite number of blocked packets in the system, all with arbitrarily low probability. This is a necessary condition for previous definitions of stability to be meaningful. In considering waiting time, we calculate the mean and other moments of the waiting time of a terminal when it enters the system to find ( n - 1 ) other terminals already there competing for the channel. Under this control, the average time is proportional to n . Two things should be pointed out. The first is that the control requires exact knowledge of the number of terminals contending for the channel, and hence is not implementable, except as an approximation. The second is that the analysis takes into account the dynamic comings and goings of terminals.

Component Signal Separation and Recombination for Linear Amplification with Nonlinear Components

Abstract: LINC is a technique that uses signal processing to produce linear amplification of bandpass signals with grossly nonlinear circuit components. Two important signal-processing functions of LINC are 1) forming two constant envelope phase-modulated signal components from the bandpass input signal and 2) recombining the amplified components to produce an amplified replica of the input signal. Two-tone laboratory tests of a LINC component separator and combiner, not including a limiter and envelope detector, show that, at full output, spurious levels 40 dB below tone level are achievable over a 1-MHz band. Because the laboratory model operated at relatively low frequencies (hundreds of megahertz), scaling up in frequency should result in a LINC with <40-dB spurious over a 10-MHz band. Spurious 30 dB below tone level should be achievable over a bandwidth of 50 to 100 MHz using the same technique of component signal separation. Lower spurious levels or greater bandwidths will require a sin-1phase modulator that is less sensitive to delay in a feedback loop.

Use of "Activity" Classes in Adaptive Transform Image Coding

Abstract: The weighted sum of the absolute values of the transform coefficients, defined herein as the activity index, is proposed as an objective measure of scene busyness (i.e., the density of significant scene detail). For an image divided into subpictures, it is possible to classify each subpicture into a finite number (say four) of categories according to its computed activity index. A different coding scheme, involving different truncation and quantization rules and hence a different number of bits, is used for each activity category. Data compression is efficiently achieved by assigning more bits to code those portions of the image showing the most detail.

On the Capacity of Slotted ALOHA Networks and Some Design Problems

Abstract: Packet switching over broadcast channels with random access schemes is of current interest for local distribution system and for satellite channels. This mode of operation is useful when the communicating devices are mobile and when the ratio of the peak to average data rate requirement of each device is high. Such systems have been analyzed for the case in which all communicating devices are within an effective transmission range of each other; either directly or through the satellite. In this paper, we address broadcast networks in which originating devices cannot directly reach the destination receiver. Thus, devices are introduced which receive these packets and repeat them to the destination. The capacity (maximum throughput) of such systems is determined, and design problems related to the number of repeating devices and the usefulness of directional antennas are resolved.

The Residual Encoder--An Improved ADPCM System for Speech Digitization

Abstract: An improved system for speech digitization using adaptive differential pulse-code modulation (ADPCM) is described. The system uses an adaptive predictor, an adaptive quantizer, and a variable length source coding scheme to achieve a 4-5 dB increase in signal-to-noise ratio over previous ADPCM. The increase can be used to improve speech quality at moderate data rates on the order of 16 kbits/s or to retain the same quality and reduce the data rate to 9.6 kbits/s. The latter alternative permits the use of narrow-band channels. The implementation complexity is on the same order as other ADPCM systems.

Quantization Error in Predictive Coders

Abstract: Predictive coders have been suggested for use as analog data compression devices. Exact expressions for reconstructed signal error have been rare in the literature. In fact most results reported in the literature are based on the assumption of Gaussian statistics for prediction error. Predictive coding of first-order Gaussian Markov sequences are considered in this paper. A numerical iteration technique is used to solve for the prediction error statistics expressed as an infinite series in terms of Hermite polynomials. Several interesting properties of predictive coding are thereby demonstrated. First, prediction error is in fact close to Gaussian, even for the binary quantizer. Sencond, quantizer levels may be optimized at each iteration according to the calculated density. Finally, the existence of correlation between successive quantizer outputs is shown. Using the series solutions described above, performance in terms of meansquare reconstruction error versus bit rate can be shown to parallel the theoretical rate distortion function for the first-order Markov process by about 0.6 bits/sample at low bit rates.

A New Type of Digital Filter for Data Transmission

Abstract: A digital filter is introduced consisting of a transversal part and a simple recursive network. The coefficients of the transversal part are equal to integer powers of two or zero; thus complicated multipliers are avoided and instead a simple routing circuit is used. Use of several types of the recursive network makes the filter applicable in different frequency ranges. The coefficients of the transversal part can be interpreted as differences of successive values of the impulse response of the filter. It is shown that this difference routing digital filter (DRDF) is especially suited for application in data transmission.

The Performance of Staggered Quadrature Amplitude Modulation in the Presence of Phase Jitter

Abstract: We report on the performance of band-limited staggered quadrature amplitude modulation (SQAM) in the presence of phase jitter and additive Gaussion noise. It is demonstrated that offsetting, or staggering, the in-phase and quadrature data streams by a fraction of a symbol interval improves the phase-jitter immunity of a conventional QAM data transmission system. For example, with a raised cosine pulse having unity rolloff, staggering can reduce the effective jitter variance by a factor of two. Under the constraint of no intersymbol interference, the optimum staggering epoch is shown to be half a symbol interval, and since the resulting system is equivalent to a form of vestigial sideband modulation (VSB)VSB is superior to QAM with respect to phase-jitter immunity. Using both optimum pulse design and data staggering, it is shown that the improvement over conventional QAM is proportional to the excess bandwidth. Consequently, SQAM may be warranted whenever a high-quality phase-locked loop is not used to track phase jitter. While the SQAM technique is not new, it has heretofore not been recognized as possessing the above-mentioned advantages.

Nonlinear Analysis of Correlative Tracking Systems Using Renewal Process Theory

Abstract: A new method is presented which describes the behavior of an (N + 1) th-order tacking system in which the nonlinearity is either periodic [phase-locked loop (PLL) type] or a nonperiodic [delay-locked loop (DLL) type]. The cycle slipping of such systems is modeled by means of renewal Markov processes. A fundamental relation between the probability density function (pdf) of the single process and the renewal process is derived which holds in the transient as well as in the stationary state. Based on this relation it is shown that the stationary pdf, the mean time between two cycle slips, and the average number of cycles to the right (left) can be obtained by solving a single Fokker-Planck equation of the renewal process. The method is applied to the special case of a PLL and compared with the so-called periodic-extension (PE) approach. It is shown that the pdf obtained via the renewal-process approach can be reduced to agree with the PE solution for the first-order loop in the steady state only. The reasoning and its implications are discussed. In fact, it is shown that the approach based upon renewal-process theory yields more information about the system's behavior than does the PE solution.

Performance of Hybrid Error Control Schemes of Satellite Channels

Abstract: The effectiveness of hybrid error control schemes involving forward error correction (FEC) and automatic repeat request (ARQ) is examined for satellite channels. The principal features of the channel are: large round-trip transmission delay due to the satellite link, and burst errors introduced by the terrestrial links that connect the users to the satellite link. The performance is estimated for two channels described by Fritchman's simple partitioned finite-state Markov model, and is compared to that obtainable if the channel is considered as a binary symmetric channel of the same bit error probability. Results show that the hybrid schemes offer substantial improvement over ARQ and FEC, and that an optimum exists for the number of errors corrected to obtain maximum throughput efficiency.

Syndrome Decoding of Convolutional Codes

Abstract: The classical Viterbi decoder recursively finds the trellis path (code word) closest to the received data. Given the received data, the syndrome decoder first forms a syndrome, instead. A recursive algorithm like Viterbi's is used to determine the noise sequence of minimum Hamming weight that can be a possible cause of this syndrome. Given the estimate of the noise sequence, one derives an estimate of the original data sequence. While the bit error probability of the syndrome decoder is no different from that of the classical Viterbi decoder, the syndrome decoder can be implemented using a read only memory (ROM), thus obtaining a considerable saving in hardware.

Noniterative Automatic Equalization

Abstract: An equalizer based on the minimum mean-square error (MSE) criterion, but with direct solution of the resulting equations, is described. The direct solution is based on the algorithms devised by Levinson and Trench. With the availability of large-scale integration (LSI) bipolar computing elements, these algorithms are competitive with iterative procedures. A method is considered for the estimation of the required parameters, and for automatic adaption to a changing channel. It is shown that the delay can be adjusted to achieve minimum MSE with respect to that parameter. Simulations undertaken show the robust performance of the algorithm, and that the equalizer performance is not adversely affected by operation in decision-directed mode.