A novel distributed scheme that combines cooperative ARQ with the spectrum leasing paradigm is proposed and analyzed. The strategy harnesses the opportunistic gains of cooperative communications, while inherently providing a spectrum-rewarding incentive for the otherwise non-cooperative relays to assist the source's transmission. As in cooperative ARQ, the source might decide to hand over the possible retransmission slots to nearby stations that were able to decode the original transmission. In the proposed scheme, however, in exchange for the cooperation, the relaying station is also awarded an opportunity to exploit the retransmission slot for its own traffic. Arbitration of relays' retransmissions is performed via an auction mechanism, with the source, the competing relays and the transmission slot acting as the auctioneer, the bidders and the bidding article, respectively. Auction theory (more generally, the theory of Bayesian games) is applied to analyze the scheme performance. It is noted that the setting here can be alternatively seen as a practical framework for implementation of property-rights cognitive radio networks. Numerical results and analysis show that the proposed scheme enables an efficient dynamic resource allocation that provides relevant gains (e.g., transmission reliability) for both the original source (primary) and the cooperating nodes (secondary users).

We propose and analyze a novel decentralized mechanism that motivates otherwise non-cooperative stations to participate as relays in cooperative ARQ protocol. Cooperation incentive is provided by the possibility for the source to lease a portion of retransmission slot for the traffic of relaying terminals. To further leverage the opportunistic nature of cooperative ARQ and obtain a fully decentralized solution, the (motivated) relaying nodes compete for access to the retransmission slot by trying to make the best retransmission offer. Effective arbitration of cooperative retransmissions is performed using auction theory (bidding), with the source in the role of the auctioneer, the relaying nodes acting as the bidders and the (use of the) retransmission slot as the bidding article. It is noted that the proposed solution can be seen as a practical framework for the implementation of cognitive radio networks running according to the property-rights model (spectrum leasing). Numerical results and analysis confirm the efficient dynamic resource allocation property of the proposed scheme, revealing the relevant gains in terms of expected number of (re)transmissions required for successful data delivery for both the source (primary) and the cooperating (secondary) nodes.

The concept of cognitive radio (or secondary spectrum access) is currently under investigation as a promising paradigm to achieve efficient use of the frequency resource by allowing the coexistence of licensed (primary) and unlicensed (secondary) users in the same bandwidth. According to the property-rights model of cognitive radio, the primary terminals own a given bandwidth and may decide to lease it for a fraction of time to secondary nodes in exchange for appropriate remuneration. In this paper, we propose and analyze an implementation of this framework, whereby a primary link has the possibility to lease the owned spectrum to an ad hoc network of secondary nodes in exchange for cooperation in the form of distributed space-time coding. On one hand, the primary link attempts to maximize its quality of service in terms of either rate or probability of outage, accounting for the possible contribution from cooperation. On the other hand, nodes in the secondary ad hoc network compete among themselves for transmission within the leased time-slot following a distributed power control mechanism. The investigated model is conveniently cast in the framework of Stackelberg games. We consider both a baseline scenario with full channel state information and information-theoretic transmission strategies, and a more practical model with long-term channel state information and randomized distributed space-time coding. Analysis and numerical results show that spectrum leasing based on trading secondary spectrum access for cooperation is a promising framework for cognitive radio.

This paper investigates the benefit of adaptive modulation based on channel state information (CSI) in direct-sequence/ code-division multiple-access (DS/CDMA) multihop packet radio networks. By exploiting varying channel conditions, adaptive modulation can be used in ad hoc networks to provide upper layers with higher capacity links over which to relay traffic. Using the /spl alpha/-stable interference model, the distribution of the signal-to-interference ratio (SIR) is obtained for a slotted system of randomly, uniformly distributed nodes using multilevel coherent modulation schemes. Performance is evaluated in terms of the information efficiency, which is a new progress-related measure for multihop networks. Three types of adaptivity are analyzed, differing in the level of CSI available: 1) full knowledge of the SIR at the receiver; 2) knowledge of only the signal attenuation due to fading; and 3) knowledge of only the slow fading component of the signal attenuation. The effect of imperfect channel information is also investigated. Sample results are given for interference-limited networks experiencing fourth-power path loss with distance, Ricean fading, and lognormal shadowing.

This paper analyzes the benefit of adaptive routing based on knowledge of the channel state information in multihop, ad hoc wireless networks that use direct-sequence code-division multiple access. Cross-layer, channel-adaptive routing exploits the inherent spatial diversity of multihop wireless networks to select links with favorable channel conditions. The information efficiency, an extension of a previously used measure called expected progress, is used to evaluate performance. Results show that, combined with adaptive modulation, adaptive routing can improve performance in ad hoc networks by a factor of four to five in channels with Rayleigh fading and lognormal shadowing. The lack of position information in the routing decision would reduce performance by 25%. New approaches to channel-adaptive routing that enable rapid adaptivity to channel conditions are discussed.

In this paper, as part of the adaptive resource allocation and management (ARAM) system (Alagoz, 2001), we propose an adaptive admission control strategy, which is aimed at combating link congestion and compromised channel conditions inherent in multimedia satellite networks. We present the performance comparisons of a traditional (fixed) admission control strategy versus the new adaptive admission control strategy for a direct broadcast satellite (DBS) network with return channel system (DBS-RCS). Performance comparisons are done using the ARAM simulator. The traffic mix in the simulator includes both available bit rate (ABR) traffic and variable bit rate (VBR) traffic. The dynamic channel conditions in the simulator reflect time variant error rates due to external effects such as rain. In order to maximize the resource utilization, both for fixed and adaptive approaches, assignment of the VBR services are determined based on the estimated statistical multiplexing and other system attributes, namely, video source, data transmission, and channel coding rates. In this paper, we focus on the admission control algorithms and assess their impact on quality-of-service (QoS) and forward link utilization of DBS-RCS. We show that the proposed adaptive admission control strategy is profoundly superior to the traditional admission control strategy with only a marginal decrease in QoS. Since the ARAM system has several parameters and strategies that play key roles in terms of the performance measures, their sensitivity analysis are also studied to verify the above foundations.

This paper investigates the benefit of adaptive modulation based on channel state information (CSI) in DS/CDMA multihop packet radio networks. By exploiting varying channel conditions on different links, adaptive modulation can be used in ad hoc networks to provide upper layers with higher capacity links over which to route traffic. Performance is evaluated in terms of the information efficiency, a new progress-related measure for multihop networks. Three types of adaptivity are analyzed, differing in the level of CSI available: (i) full knowledge of the SIR at the receiver, (ii) knowledge of only the signal attenuation due to fading, and (iii) knowledge of only the slow fading component of the signal attenuation. The effect on performance of imperfect channel information is also investigated. Sample results are given for interference-limited networks experiencing fourth-power path loss with distance, Rayleigh fading, and lognormal shadowing.

This paper concerns with the minimum mean-squared error (MMSE) detection of synchronous multicarrier CDMA systems. Maximal-ratio combining (MRC) MMSE detection schemes are proposed for the multicarrier CDMA systems. In the proposed MRC MMSE receiver, the MMSE detection is first performed separately for each carrier, and then the outputs of each MMSE detector are combined in such a way that the resulting signal-to-noise ratio (SNR) is maximized. The performance of the considered MRC MMSE schemes is compared with the conventional MMSE receivers of both jointly and separately designed MMSE filters. It is found that when the system is lightly loaded, the MRC MMSE detection scheme performs comparably to the jointly optimized MMSE detection in the presence of slow frequency-nonselective independent Rayleigh fading for each carrier.

Adaptive modulation in OFDM is a means by which the OFDM transmitter adapts the subchannel bit and power allocation to the amplitude response of the frequency selective channel for improved performance in high data rate wireless communications. Previous studies demonstrating the performance gains of adaptive modulation have assumed the availability of perfect channel state information. This paper examines the impact on performance of an adaptive OFDM system due to imperfect channel information arising from realistic channel estimators and time-varying channels. The latter is of particular relevance to mobile wireless applications, for which OFDM has received much attention. The performance of a typical channel estimator is found to be sufficient to justify the use of adaptive modulation in wireless OFDM systems, but in mobile situations the effect of outdated channel information in the presence of quickly time-varying channels can be significant. The use of channel prediction is shown to mitigate the impact of outdated channel information and improve performance when the Doppler spread, or mobile velocity, is high.

Transmitter optimization, in addition to receiver optimization, contributes significantly to efficient interference suppression in multiple access and multipath channels. The system design is based on the joint optimization of the transmitter and the receiver in a synchronous multiuser channel characterized by multipath propagation. Joint optimization is represented by a linear transformation of the transmitted signals at the transmitter and a linear transformation of received signals at each receiving site that minimize the effect of multiple access and multipath interference. The minimum mean squared error between the true bit value and its estimate at the output of the receiver is taken as the cost function, subject to average and peak transmit power constraints. It is shown that joint transmitter-receiver optimization outperforms significantly either transmitter optimization or receiver-based techniques. The crucial assumption, in the case of multipath channels, is that the transmitter knows the multipath characteristics of all channels and that the channel dynamics are sufficiently slow so that multipath profiles remain essentially constant over a block of preceded bits. The practical applications can be found in indoor and cellular communications, satellite communications, or military communications where nonorthogonal signature waveforms are employed.

We are interested in determining the feasibility of having multiple service providers share the same spectrum in transponder-type. Satellite-based, code-division multiple-access mobile communications. To accomplish this, we compare the total capacity when n systems are operational to that when only a single system uses the frequency band. This comparison is made for both the uplink and the downlink under one of two possible constraints, either a maximum power flux density (PFD) limit per system, or an aggregate PFD limitation. While there are circumstances when the total capacity of multiple systems can achieve a greater value than that of a single service provider, conditions depend on the degree of shadowing, and whether or not the PFD limit is on a per system basis or on an aggregate basis from all systems.>

A DS CDMA satellite uplink with power control error is considered. CDMA system performance is normally interference limited, and individual users typically experience the average performance over all users. This is particularly true in a Gaussian, additive white noise channel, with many users, sufficient processing gain, and effective power control. In this case, the other users appear as Gaussian noise and each user experiences the same environment. Under fading channel conditions however, the average behavior notion is flawed, since the fade conditions for each user are different, even if the fading statistics are identical. An alternative measure of performance under these conditions is the probability of outage. For coded signals, the fading must be averaged over all the symbols in the codeword. For properly interleaved signals and i.i.d. Rayleigh fade statistics, the average must be performed over a chi-squared variate. The probability of outage, defined as the probability that the decoded bit error rate exceeds a specified threshold, is monotonic with the probability that the chi-squared variate is less than a specified threshold. The authors derive upper and lower bounds on the outage probability for soft decision decoding.

The analysis of both the performance and capacity of direct sequence CDMA in terrestrial cellular systems has been addressed in the technical literature. It has been suggested that CDMA be used as a multiple access method for satellite systems as well, in particular for multispot beam low Earth orbit satellites (LEOS). One is tempted to argue that since CDMA works well on terrestrial links, it will nominally work as well on satellite links. However, because there are fundamental differences in the characteristics of the two channels, such as larger time delays from the mobile to the base station and smaller multipath delay spreads on the satellite channels, the performance of CDMA on satellite links cannot always be accurately predicted from its performance on terrestrial channels. In the paper, the authors analytically derive the performance of a CDMA system which operates over a low Earth orbiting satellite channel. They incorporate such effects as imperfect power control and dual-order diversity to obtain the average probability of error of a single user.>

The paper presents an up-to-date overview as well as new results on mobile satellite communications. Specifically, the following subjects are addressed: worldwide activities; satellite orbits; multiple access techniques; Novstar global positioning system; economic and regulatory issues; future trends. Some of these technical and nontechnical subjects appeal disjoint, but they are unique and significant among the many facets, which effect the development and operation of mobile satellite communication systems.>

Traffic models and queueing performance of variable-bit-rate (VBR) video sources in an asynchronous transfer mode (ATM) network are studied. A discrete-time discrete-state Markov chain is used to model the aggregate video traffic with each VBR-coded source being modeled by a renewal process, which has been successfully applied in the analysis of packet voice traffic. Three different methods including the stationary-interval (SI) method, the asymptotic method (ASM), and the hybrid method for queueing network analyzer (QNA) are used to approximate the average queue size. Results for different traffic conditions and different number of VBR sources are compared with the simulation results. It can be observed that as the number of sources increases the aggregate traffic becomes more predictable and the congestion at the common queue becomes smaller. This result verifies the fact that multiplexing a large number of identical video sources on a single high speed link statistically yields significant bandwidth saving. It is also interesting to note that the SI method provides an upper bound and the ASM method yields a lower bound for the average queue size for the type of traffic used in the study. When the number of VBR sources increases, the result deviates from the SI method and approaches the ASM method. In general, the QNA method provides a close match to the simulation result.>

Some results of a study of the command security issues associated with the INTELSAT VI and VII satellites are reported. The configuration and protocols of the INTELSAT command system are briefly described. Three possible configurations for connecting the INTELSAT headquarters and the telemetry, tracking, and command (TTC) stations are distinguished, and a layered architecture is introduced to illustrate the command protocols corresponding to the ISO layers. The impact of introducing command security is then studied for the INTELSAT command network. In order to analyze the effect of errors on the operation and performance of the command network, a finite-state Markov chain is proposed for modeling the command and telemetry channels with their different error and delay characteristics. Possible risks that can threaten the INTELSAT secure command network operations, including single-event upsets, Vcc corruption, and manipulation of transmitted messages, are analyzed. The end-to-end performance of the INTELSAT command network is examined for three possible configurations-terrestrial link, single-hop satellite, and two-hop satellite.>

A combined-transform coding (CTC) scheme to reduce the blocking effect of conventional block transform coding and hence to improve the subjective performance is presented. The scheme is described, and its information-theoretic properties are discussed. Computer simulation results for a chest X-ray image are presented. The CTC scheme, the JPEG baseline scheme, and the conventional discrete Walsh-Hadamard transform (DWHT) are compared to demonstrate the performance improvement for the CTC scheme. The advantages of the CTC scheme include no ringing effect as there is no error propagation across the boundary, no additional computation, and distortion always held within a certain level.>

The rate-distortion performance of a class of non-Gaussian source with memory is studied. The source model is generated from a correlated Gaussian source through a memoryless transform, which actually constitutes a special class of the frequently-used Gibbs model in speech and image processing applications. Its entropy and a rate-distortion bound are evaluated to obtain further insights into this class of source models. Several examples are also given to illustrate the usefulness and tightness of the rate-distortion bound.>

Because of the continually increasing demand for mobile communications, it has been suggested that personal communication networks (PCNs) be established in the 1850-1990 MHz range. However, that band of frequencies is currently occupied by various microwave signals transmitted by users ranging from utility companies to state and local agencies. In order to allow both sets of users to occupy these frequencies as well as improve the spectral efficiency of this band, a spread-spectrum overlay is proposed, whereby a code-division multiple-access (CDMA) PCN would share the spectral band with the existing narrowband microwave traffic. The results of several field tests which have been designed to demonstrate the feasibility of an overlay of this type are discussed.>

A combined-transform coding (CTC) scheme is proposed to reduce the blocking artifact of conventional block transform coding and hence to improve the subjective performance. The proposed CTC scheme is described and its information-theoretic properties are investigated. Computer simulation results for a class of chest X-ray images are presented. A comparison between the CTC scheme and the conventional discrete cosine transform (DCT) and discrete Walsh-Hadamard transform (DWHT) demonstrates the performance improvement of the proposed scheme. In addition, combined coding can also be used in noiseless coding, yielding a slight improvement in the compression performance if it is used properly.

Code Division Multiple Access (CDMA) is a digital multiple-access technique, whereby each signal has its own unique binary sequence, and all signals share the same spectrum. In the United States, personal communication networks using CDMA, will provide to each user, a separate code, which will change from cell-to-cell. We call this code reuse, just like the frequency reuse employed in the FDMA or TDMA systems. However, there are many more codes, than frequency bands.

Many compensative measures have been proposed recently the make the cell loss in ATM (asynchronous transfer mode) networks subjectively imperceptible. These schemes include the simple automatic repeat quest (ARQ) scheme, error concealment by command refreshment, appropriate queueing disciple and priority switching design and layered source coding schemes. The authors summarize these efforts and, in particular, elaborate on different layered source coding schemes. Four types of signal priority classification schemes are identified: bit-plane separation (FDS) combined with bit-plane-frequency separation (CBFS), and feature plane separation (FPS). Different layered coding techniques are discussed and compared. Some open questions and recommendations are also presented for further research and study.>

The probabilistic distribution properties of a set of medical images are studied. It is shown that the generalized Gaussian function provides a good approximation to the distribution of antero-posterior chest radiographs. Based on this result and a goodness-of-fit test. a generalized Gaussian autoregressive model (GGAR) is proposed. Its properties and limitations are discussed. It is expected that the GGAR model will be useful in describing the stochastic characteristics of some classes of medical images and can be used in image data compression and other applications.>

A scheme which allows the existence of two priority classes on a channel, each using a standard carrier-sense multiple-access with collision detection (CSMA/CD) protocol, is introduced. The two priority classes considered are labeled the high-priority class (HP) and the low-priority class (LP) of users. The scheme is based upon the principle that the exclusive access right to the channel is given to the HP users after each successful transmission from all users on the channel. This is accomplished by reserving the channel for a time window during which only HP users can contend for accessing the channel using the CSMA/CD protocol. If the channel is not used during that window or if it is used by more than one HP user (which results in a collision), it is released once again for use by both types of users. The equilibrium throughput and delay equations for both types of users are derived. The effect of HP users' percentage on the channel, the HP packet's length, the width of the reservation window, and the channel traffic on the throughput and delay for both types of users is investigated. The priority scheme introduced lends itself well to the introduction of multiple classes of priority users on the same channel.>

The simple automatic repeat request (ARQ) scheme, error concealment by command refreshment, appropriate queuing discipline and priority switching design, layered source coding schemes, and other signal processing techniques are summarized with emphasis on different layered source coding schemes. Four types of signal priority classification schemes are identified: bit-plane separation (BPS), frequency-domain separation (FDS), combined bit-plane-frequency separation (CBFS) and feature plane separation (FPS). CBFS is shown to offer a good alternative to the FDS for possible cell dropping compensations in the ATM network. Different layered source coding schemes are described and compared, and a CBFS-based combined-transform coding (CTC) scheme is highlighted.>

Consideration is given to the performance of a binary phase-shift-keyed (BPSK) direct-sequence spread-spectrum (DS/SS) communication system in a fading dispersive channel with the direct or specular component of the signal present. Two types of multipath intensity profiles are analyzed. Large values of the multipath spread (larger than a symbol duration) are considered, and a useful approximation to the symbol error probability, based on random spreading sequences, is derived. The effect of pulsed jamming on the performance is analyzed. Both the uncoded and the coded case are considered. In the coded case, hard- and soft-decision decoding metrics, with and without jammer-state information (JSI), are analyzed and discussed.>

The performance of a binary phase-shift-keyed direct-sequence spread-spectrum communication system in a fading dispersive channel with jamming is examined. The channel is modeled as a wide-sense-stationary uncorrelated scattering channel, dispersive only in time. The channel output consists of a strong stable component of the signal plus a continuum of dispersive components and is referred to as a frequency-selective Rician fading channel. To obtain specific results, the multipath intensity profile is modeled by triangular function. Both the uncoded and coded performances are analyzed. The multipath spread is chosen to limit intersymbol interference to K adjacent coded symbols in the coded case and to only adjacent data bits in the uncoded case. No specific interleaver design is considered. A closed-form expression for the probability of error is obtained, assuming random spreading sequences. Numerical results are presented for values of the system and channel parameters. It is shown that a jammer can optimize its strategy if the probability of error is not too high.>

The data queuing performance is analyzed for a fixed-frame-size, integrated voice-data multiplexer (IVDM), where voice has preemptive priority over data, a maximum capacity is available for voice (above which voice freeze-outs occur), and data is queued when there is insufficient capacity. A discrete-state-continuous-time Markov model is used to model a discrete-state-discrete-time IVDM when the average source burst duration is significantly larger than the multiplexer frame time. The correlation of the voice and data source activity is shown to have a significant effect on IVDM data queuing performance. An approximate, simple analytical model for the mean data queue length is derived. The approximate analytical model for mean data queue length is shown to provide a lower bound for exact calculations and simulation results. The concept of maximizing queuing power (throughput/delay) is applied to the data performance.>

In this paper, we briefly describe the physical characteristics of a meteor-burst (MB) communication channel, and then present the results of an analysis showing the expected throughput and the probability of completing a given message under two different scenarios. In the first scenario, the transmitter is presumed to have knowledge (obtained by appropriate probe signals) of the entire duration of a given MB. In the second, only the starting time of each MB is known. Numerical results based on the analysis are provided, and it is shown what types of delays must be tolerated in order to achieve a given degree of reliability of having the transmitted message correctly received at its destination.

A new average equation for the adaptive weights which takes into account the effects of statistical weight fluctuations has been developed. Past researchers have derived results by using an approximate average equation for the weights obtained under the assumption that the weights vary much more slowly than the input process. The weights are then treated as constants for averaging. The new method employs a two-variable perturbation analysis which uses as a small parameter the ratio of the loop bandwidth to the input process bandwidth. The analysis yields a significant correction term for the directivity pattern in interference nulls. This correction term is the result of stochastic weight fluctuations which are usually ignored in most analyses. Comparison of these results to those found by past researchers is given and the results of a computer simulation substantiate the analysis.

-A model of a moving-boundary, fixed frame length, integrated multiplexor is proposed and analyzed. The assignment of slots within a frame to the voice and data sources is made by an allocation function. The joint distributions of queue length and expected waiting time are derived.

This paper presents a simple phasor model for analyzing coherent direct-sequence, pseudonoise (PN), spread-spectrum carriers accessing limiting repeaters in the presence of strong interference. Limiting repeaters are found in several applications, one being satellite relays. The model used provides a rigorous upper bound on system bit error rate depending only on an equivalentEb/N0. Presented for the first time is an application of the model to RF limiters having an AM-to-PM conversion effect. For quadriphase PN, receiverEb/N0is given by the heuristic approach of replacing the limiter with an average power-limited linear device plus an additional lossLdue to limiting. As long as there is no AM-to-PM conversion in the limiter,Lis the same as that obtained by passing multitone CW plus noise through the limiter. For quadriphase PN carriers passing through a limiter with AM-to-PM conversion,Lcan depend on the type of coherent phase tracker used.Eb/N0for biphase PN, in contrast to quadriphase PN, need not have the usual heuristic form given by quadriphase PN.

The performance of four digital modulation techniques are compared to one another when each is received in the presence of adjacent channel interference. The interfering waveforms that each modulation format sees are identical to the modulation format under consideration. The four types of modulation are QPSK, 8-PSK, 16QASK, and QPR. Upper bounds to the probability of error are derived and used to evaluate the relative merits of the different schemes.

Spread-spectrum communications, with its inherent interference attenuation capability, has over the years become an increasingly popular technique for use in many different systems. Applications range from antijam systems, to code division multiple access systems, to systems designed to combat multipath. It is the intention of this paper to provide a tutorial treatment of the theory of spread-spectrum communications, including a discussion on the applications referred to above, on the properties of common spreading sequences, and on techniques that can be used for acquisition and tracking.

The probability of error of both a 16-ary QASK system and an MSK system, each operating over a Rician fading channel, is considered. The channel model has two key features: short term variations are modeled by making the channel frequency selective, and long term variations are accounted for by allowing the specular component of the received signal to fade according to a Rayleigh density. The improvement gained by using maximal-ratio diversity combining is derived, as well as the degrading effects of an adjacent channel. Finally, a comparison between MSK and 16-ary QASK operating over the same channel is presented, and it is shown that MSK results in a smaller probability of error.

In a packet network using medium speed terrestrial connectivity, average end-to-end delay can be reduced by using highspeed satellite overflow channels despite their propagation delay. To investigate networks of this type analytically, we derive a new basic queueing model for overflow systems with buffered traffic. The simple, closed-form expressions facilitate analysis. Approximations are developed for multiple primary systems with a single overflow channel; these agree with simulation. Using the overflow model analysis indicated the following. The reduced queueing delay to the high capacity overflow channel more than compensated for propagation delay. Inherent, broadcast transmission capabilities of satellite channels reduced overall network delay. The high capacity overflow channels permit networks to withstand substantial nodal imbalance and overloads.

The gueast editors attempt to make the arguments: 1) that whereas there are many similarities between military and civil systems, there are also many differences; 2) that thes ources of the differences are iargely summed up with the terms survivability and interoperability; and 3) that broad systems approaches are necessary to achieve the desired goals. The systems approach is rarely described in technical journals. Because of this, the papers invited for this Special Issue feature systems approaches and overviews designed to give the reader some perspective as to what the main issues are in military communications. There is also a sprinkling of techniques-oriented papers, some invited and some submitted, to complete the picture. In keeping with this systems orientation, the Special Issue is divided,i nto four major sections, oriented to the principal categories into which the Defense Department divides its systems for management purposes. The first section contains two papers, both providing architectural overviews over system evolution. The second section is devoted to the category of Long-Haul Communications, with papers addressing various aspects of this category: systems, subsystems, and technology. The third section contains ten papers devoted to Tactical Communications, the name given to the category supporting command and control from the major field comnianders down to the lower echelons of command. The final section containing eight papers treats a variety of technical topics which are neither specifically long-haul nor tactical, but which, although primarily techniques-oriented, address specific military issues.

It is desired to determine the performance of a noncoherent FSK frequency-hopped spread spectrum communication system using MARK and SPACE tones which are not necessarily orthogonal, and in addition using an error-correction code to reduce the susceptibility of the system to jammer-induced errors. The overall bandwidth of the system (i.e., the spread bandwidth) will be held constant so that the processing gain of the system will always be adjusted to maintain this constant bandwidth as other factors (e.g., the bandwidth expansion due to the error correction coding) change the underlying baseband bandwidth.

In most realistic communication systems, be they spread spectrum or conventional systems, some overall bandwidth constraint is usually placed upon the system. In the case of spread spectrum communications, this constraint limits the amount of processing gain that the system can employ. In addition to processing gain, factors such as alphabet size affect the total bandwidth. It is the purpose of this paper to combine these two effects to determine how one should trade off the amount of processing gain one uses relative to the underlying modulation bandwidth when one is operating under the constraint of fixed total system bandwidth.

Summary form only given, as follows. On page 1398 of the above paper (ibid., vol. COM-28,Part II,pp. 1389-1398, Aig. 1980), the Acknowledgment was published incorrectly. The correct Acknowledgment is as follows: "ACKNOWLEDGMENT The authors would like to thank Prof. C. Ziegler for generating the numerical results, and J. Bartow and S. Hall for their helpful suggestions."

A new reservation-based time-division multiple access scheme is proposed. This scheme is analyzed as a cyclic-multiqueue attended by a single server, with attention being focused on an imbedded process obtained by examining the system at special epochs located at the beginnings of each cycle. The resulting functional equation is solved by simplification using an independence assumption. The mean and standard deviation of queue sizes (at the epochs) and cycle times thus obtained are verified, for the special case of Poisson arrivals, using a GPSS-V simulation model.

This paper addresses itself to some comparisons of adaptive routing algorithms in store-and-forward Communication nets. The intent here is to demonstrate how some relatively simple add-ons to already existing adaptive algorithms can decrease the average message delay and increase message throughput in the network. The overview of many existing routing algorithms is intended to point out that, although an algorithm may be adaptive, it is not necessarily good in terms of the performance measures addressed in this paper. The basic objective of this study is to assess the effects of endogenous priority assignment to messages that have reached some specified aging threshold while in the network and to note the effects of Such priority assignment on network performance. The performance measures are average message delay, throughput, and number of messages undelivered. The routing techniques are demonstrated via simulation on an 8node highly connected network and a 19-node Advanced Research Projects Agency (ARPA) network. The routing algorithms are applied with and without network element destruction on the selected networks.

The successive overrelaxation iterative technique is used as the algorithm for the iterative adjustment of transversal equalizer tap gains during a training period using isolated pulses for the minimization of the mean-square error. For certain channel models, substantial improvement is demonstrated in the rate of convergence over the commonly used gradient techniques. Numerical simulations indicate that the relaxation algorithm consistently converged much faster than the gradient techniques. The behavior of the algorithm in noise is also investigated and is shown to be no worse than any other equalizer algorithm.

The classical problem of detecting known signals in additive nonstationary nonwhite noise is treated using state-variable methods. The noise is assumed to be generated as the solution of a linear time-varying differential equation driven by white noise. The Markov property of the noise is used to derive a difference equation for the likelihood ratio. The terms in the difference equation are given directly in terms of the coefficients of the differential equation defining the noise. These results, being in recursive form, reduce considerably the computational effort when discrete samples are used. The signal-to-noise ratio is also derived. Explicit expressions for the likelihood ratio are also obtained for continuous processing. These results are given directly in terms of the signals and the noise-generating equation. Thus, there are no integral equations to solve. The signal-to-noise ratio is calculated explicitly for both stationary and nonstationary noise.