Abstract This paper proposes a new cross‐layer optimization algorithm for wireless mesh networks (WMNs). CDMA/TDD (code division multiple access/time division duplex) is utilized and a couple of TDD timeslot scheduling schemes are proposed for the mesh network backbone. Cross‐layer optimization involves simultaneous consideration of the signal to interference‐plus‐noise ratio (SINR) at the physical layer, traffic load estimation and allocation at medium access control (MAC) layer, and routing decision at the network layer. Adaptive antennas are utilized by the wireless mesh routers to take advantage of directional beamforming. The optimization formulation is subject to routing constraints and can be solved by general nonlinear optimization techniques. Comparisons are made with respect to the classic shortest‐path routing algorithm in the network layer. The results reveal that the average end‐to‐end successful packet rate (SPR) can be significantly improved by the cross‐layer approach. The corresponding optimized routing decisions are able to reduce the traffic congestion. Copyright © 2010 John Wiley & Sons, Ltd.

In this paper we investigate performance of resource allocation schemes in WLANs of metropolitan area using HiperLAN type 2 standard. Inside the WLANs user moves from one place to another and user rates are dynamically adjusted based on its distance from the access point. To manage the wireless resources of the network we propose resource allocation schemes and evaluate their performance. We develop a generic simulation software for the network and use it for three resource allocation policies namely minimum overhead round robin (MORR), which does not depend on user's buffer condition, weighted minimum overhead round robin (WMORR) which is a function of user buffer as well as the waiting time for transmission opportunity and weighted round robin (WRR) which is a function of user buffer only. For performance comparison we evaluate average and variance of number of packets in the user buffer, user buffer overflow and overhead in the uplink phase. Our results show that the second adaptive resource allocation technique i.e., WMORR outperforms the other two.

In this paper, we investigate how Multiple Description Coding (MDC) can improve the user's satisfaction for a group of heterogeneous destinations for video multicast over wireless ad hoc networks. Specifically, we explore the independent-description property of MDC with multiple paths to improve the user's satisfaction. In our approach, the assignment of multiple description (MD) video is randomly done by a multicast source node. Based on the assignment of MD video, the multicast source constructs multiple multicast trees accordingly. Simulation results demonstrate that our approach can achieve higher user's satisfaction regardless of the multicast group size.

Abstract Efficient utilization of bandwidth and high data rates have a great impact on the performance of wireless networks. The cdma2000 1xEV‐DO standard provides high‐speed wireless data services to mobile subscribers based on CDMA technology. In this paper, we study the bandwidth utilization for the 1xEV‐DO packet mode standard. In particular, we develop an analytical model for lowest‐rate‐first, highest‐rate‐first priority scheduling techniques, and two round‐robin fair scheduling techniques over the reverse data channel in cdma2000 1xEV‐DO. For these four scheduling techniques, the distribution of the mobile stations (MSs) among the possible data rates is modeled as a Markov process. An analytical expression for the steady state system throughput is derived from the steady state distribution of the above Markov process. The developed model is validated through simulations. Copyright © 2008 John Wiley & Sons, Ltd.

Wide band mesh or star oriented networks have recently become a subject of greater interest. Providing wideband access for a variety of applications has led to the inception of WIMAX. Classic access techniques such as FDMA and TDMA have been norm for such networks. Their maximum transmitter power is much less than their TDMA and FDMA counter parts, which are very important assets for mobile operation. In this paper we introduce a code division multiple access/Time division duplex technique CDMA/TDD for such networks. The CDMA approach is an almost play and plug technology for wireless access, making it amenable for implementation by WIMAX service station. In addition we adopt a virtual circuit (label) to improve packet overhead efficiency. We outline briefly the new transmitter and receiver structures then evaluate efficiency, delay and delay jitter. By analysis we show the advantages over classic counter parts with respect to total network efficiency achievable especially for larger number of hops.

In this paper we present an optimized resource allocation algorithm for the uplink of space frequency block coded code division multiple access (SFBC-CDMA) systems. The algorithm allocates appropriate base transceiver stations (BTS), antenna array sub system and frequency subcarrier blocks to users such that the pair wise cross correlation between the users is minimized while maximizing the channel coherence time. The intermediate effects such as reduction of the channel estimation error and higher signal powers culminate into more cost effective multiple input multiple output (MIMO) operation. The proposed algorithm shows a noticeable improvement in the bit error rate of the users and the bandwidth efficiency of the system compared to random resource assignment.

This paper presents a novel approach for the construction of Multiple Shortest Path Trees (MSPT) using multiple description (MD) video coding to satisfy the destination requirements (i.e., each destination has a preference number, which depends on its capacity, of video layers). Specifically, the independent-descriptions property of multiple description (MD) video coding along with multiple trees are exploited to increase user satisfaction. This approach is suitable for heterogeneous wireless ad hoc networks. Simulation results demonstrate that our proposed scheme (MSPT-MDC) outperforms MSPT-LC (multiple trees with layered coding) in terms of user satisfaction. In addition, simulation results show that user satisfaction, of the proposed scheme, is insensitive to the change of the number of destinations.

In this paper we propose resource allocation schemes for TDMA/TDD based high performance link adaptable wireless LANs and examine their interrelation with user buffer. User rates are dynamically adjusted based on their distance from the access points and changes as user moves. We simulate three resource allocation policies namely Minimum Overhead Round Robin (MORR), which does not depend on userpsilas buffer condition, Weighted Minimum Overhead Round Robin (WMORR) which is a function of user buffer as well as the waiting time for transmission opportunity and Weighted Round Robin (WRR) which is a function of user buffer only. We evaluate average buffer packet drop probability due to buffer overflow and average frame overhead in uplink phase of the TDMA/TDD frame. Our results show that the second adaptive resource allocation technique i.e. WMORR outperforms the other two.

In this paper, we address the issue of multilayered multicast routing in wireless ad hoc networks (WAHNs). Existing multilayered multicast protocols assume homogeneous ad hoc wireless networks; in which all nodes are of the same type (they have the same processing, data rate and communication capabilities and characteristics). A more realistic assumption is a heterogeneous network; in which nodes have different processing, communication capabilities and other characteristics. In this paper, we assume heterogeneous network; in which nodes have different capabilities. Two multilayered multicast routing algorithms are proposed, namely, multiple trees based on shortest path tree (MSPT) and multiple trees based on Steiner minimum tree (MSMT). We assume that each destination has a preference number of video layers; which is equal to its capacity. Moreover, we do not consider only the capacities of nodes in the network but also the bandwidth of each link. Simulations show that the proposed schemes greatly improve the QoS requirements (improve user satisfaction ratio (USR)) for a set of destinations. In addition, simulations show that multiple trees schemes achieve substantially higher satisfaction ratio than the single tree scheme.

The capacity demands required to handle new wireless services such as e-mail, web surfing, and video streaming led to a new era of wireless technologies. cdma2000 1xEV-DV is considered to be the solution for the ever-growing demand of high-speed wireless packet data transmission. We investigate several aspects related to autonomous rate assignment schemes over the cdma2000 1xEV-DV reverse packet data channel. In particular, based on the developed rise over thermal model, we provide an upper bound for the reverse packet data channel throughput as a function of the number of mobile stations that are allowed to transmit instantaneously on each time slot. We also propose several autonomous rate assignment schemes that provide a significant throughput improvement relative to other published schemes.

Abstract Radio resource management (RRM) is vital for the next generation wireless networks. RRM comprises many functionalities and this paper focuses on the investigation of the performance of several adaptive call admission/congestion control policies based on a window‐measurement estimation of the status of the buffer at the base station under the hybrid TDMA/CDMA access scheme. In our study, we interrelate the physical limitations of the base stations (i.e. the number of transmission and reception modems), call and burst level traffic, instantaneous buffer conditions and end‐to‐end bit error performance in one queuing problem. Subsequently, a window‐measurement estimator is implemented to estimate the likelihood of buffer congestion at the base station. Accordingly, the traffic loads shall be controlled. We use event‐driven simulation to simulate the multimedia integrated CDMA networks where heterogeneous traffic users are multiplexed into a simple TDMA frames. The simulation results show outstanding performance of the proposed call admission/congestion control policies in guaranteeing QoS requirements. Copyright © 2005 John Wiley & Sons, Ltd.

The rapid growth of multimedia group-based applications motivates research into QoS guaranteed multicast network. In the multicast network carrying traffics of different QoS requirements, bandwidth fragmentation is considered as a bottleneck for traffic admission fairness and efficient usage of bandwidth resource. When the bandwidth fragmentation happens, the chance of accepting high-bandwidth flows is greatly decreased. To prevent bandwidth fragmentation in multicast network, we propose a new approach of bandwidth fragmentation avoided QoS multicast routing in this paper. In our new approach, we integrate active admission control into traditional QoS multicast routing algorithms to gain bandwidth fragmentation immunity. Moreover, an algorithm named dynamic bandwidth allocation with adaptive constraint is proposed and investigated as an example of active admission control algorithm in this paper. Because this algorithm is nonlinear and unsolvable by analytical approach, we employ OPNET simulation to study its performance.

The next generation communication system will provide high quality multimedia service in a more flexible and intelligent manner. In this paper, we propose a new SIP over MPLS network architecture to achieve this goal. To integrate SIP protocol with the traffic engineering (TE) function of MPLS network seamlessly and facilitate SIP call setup, the SIP-MPLS traffic aggregation server (TA server) is highlighted in this new architecture. Furthermore, the call admission and bandwidth renegotiation algorithms running on the TA server are investigated carefully. We rely on OPNET simulation to assess the performance of our network architecture and the capability of different call admission and bandwidth renegotiation algorithms. The simulation results show that the algorithm of TBCA-BP is a suitable candidate for accomplishing the call admission and bandwidth renegotiation task due to its satisfying performance, and our network architecture is able to shorten the SIP call setup delay greatly by using the TA server.

To make the best use of network resources across a QoS guaranteed multicast network, it is urgent to add a traffic engineering mechanism to the existing QoS multicast routing protocols. We propose a new approach of integrating QoS multicast routing with TE-oriented admission control to offer a traffic engineering function. Moreover, a TE-oriented admission control algorithm, namely dynamic bandwidth allocation with adaptive constraint (DBA-AC), is suggested to make the best use of resources across the entire multicast network by distributing traffic over different paths. Because this admission control algorithm is nonlinear and unsolvable by an analytical approach, we employ OPNET simulation to study its performance.

Next generation wireless networks are characterized as heterogeneous networks, particularly in terms of its underlying technology. One of the challenges of these heterogeneous networks is to manage handoff. Mobile IP is chosen for managing the handoff to accommodate the all-IP vision of the future interconnected networks. However, the handoff management of the mobile IP is mainly for data services where delay is not of a major concern. Therefore, it would be considerable challenge to achieve low latency handoff for real-time services. In this paper, we propose a multicasting scheme for delay-sensitive applications. The proposed scheme is shown to reduce the latency introduced during the process of mobile IP-based handoff between heterogeneous networks. For low data rate applications, in which real-time voice services operate, the proposed scheme implements an adaptive packet size technique to reduce both the probability of packet loss and the handoff latency. We present simulation results to verify the improvements achieved using the proposed multicasting scheme as opposed to the standard mobile IP

In recent years, the use of directional antennas in wireless networks has been widely studied. Since the medium access control (MAC) protocol of the IEEE 802.11 standard is designed for the use of omnidirectional antennas, it cannot perform efficiently when directional antennas are used. In this paper, we study the performance of an efficient two-channel MAC protocol for ad hoc networks when equipped with directional antennas. The proposed protocol utilizes the large throughput offered by directional antennas using two frequency division multiplexed channels. The first channel is used for control information and the second for user data transmission. Based on this, the proposed MAC protocol operates in two main modes, the omnidirectional mode where one antenna is used for the transmission of users' control frames, and the directional mode where antenna arrays are used for the transmission of data frames. The proposed protocol is assessed using computer simulations based on randomly generated network topologies reflecting the random movement of nodes in the network. Based on these random topologies, we present performance comparisons with the existing MAC protocols using different system parameters. In all cases, the proposed MAC protocol is shown to offer a significant throughput improvement relative to the existing protocols

By most predictions, fourth-generation wireless systems will be composed of heterogeneous networks. Two important constituent networks will be wireless LAN (WLAN) and cellular mobile radio systems (e.g., UMTS). WLAN is designed for low-range, high/medium data rate access and can be used as a complement to larger cellular mobile radio systems. In this paper, we consider a heterogeneous system in which WLANs are built on the boundaries of the UMTS cells with hot spots to provide additional bandwidth and to support handover between cellular cells. The cellular bandwidth reservation for handover and the new call queueing are considered. We show that such an integrated system can be modeled by a discrete 3-D Markov chain. Using this analytical model, the handover and new call blocking probabilities, queueing probability and mean queueing delay are found under various system parameters. These results then allow us not only to quantify the performance gain of integrated systems but also to optimize system parameters.

A call admission control algorithm must try to admit as many calls as possible provided that their quality of service (QoS) requirements can be met without violating those of previously admitted calls. In this paper, we propose a simple and effective call admission control algorithm and its associated resource allocation mechanism ,referred to as the flexible call admission control, for recently proposed multi-pattern (MP) wireless local area networks (WLANs). The proposed flexible call admission control effectively takes the advantages of flexible pattern assignment in MP WLANs and the rate-adaptive feature of multimedia services to support multiple classes of traffic with diverse QoS requirements and priority levels. With the use of an innovative performance estimation mechanism, the proposed flexible call admission control and resource allocation algorithm has considerably lower complexity than that of the existing schemes. Simulation results have demonstrated that the use of MP FAC provides much higher system throughput and lower call blocking probability. It should be emphasized that the proposed FAC, although designed for MP WLANs, also works well with existing standard WLANs.

The expected number of retransmissions is a very important parameter to evaluate the multicast performance. It is often used to estimate other performance parameters such as bandwidth consumption and delay. A simple and efficient evaluation to these parameters is helpful for design of multicast protocols. In this work, we will derive a good analytical evaluation to the number of retransmissions. We also obtain the total bandwidth consumption for hierarchical reliable multicast where some special receivers or routers are assigned for retransmission handling of each domain. The optimal partitioning of the tree also depends on topology of networks for each subgroup. The effects of topology on optimal placements are analyzed using bandwidth consumption as criteria for hierarchical reliable multicast.

Compelling features of wireless local area networks (WLANs), put a variety of wireless service demands in place. In order to adjust system parameters to fulfill specific needs of different applications, a mathematical description of the system turns to be helpful. The inherent complexity of the wireless access, makes this description very challenging. We propose a new performance model for the IEEE 802.11 WLAN in ad hoc mode. The ad hoc mode has been chosen since we eventually aim at interconnected WLAN clusters where no base station exists. The model is based on the presentation of the system with a pair of one-dimensional state diagrams which can easily accommodate variations of many input parameters. The corresponding state variables are contention window size and buffer occupancy of each user in the system. The input parameters considered are: packet fragmentation factor, buffer size, and maximum allowable number of retransmissions. However, the approach taken is capable of ingesting many other probable parameters of interest. System performance criteria under study are: throughput, delay, and probability of fail to deliver. The last two are crucial for real-time applications.

In this paper the bit error probability of the SFH/MFSK system in presence of wideband and tone interference in a fading environment is evaluated. Simultaneous license-free operation of TDMA, and direct sequence (DS) based LAN clusters is assumed to be the prime interference facing the subject FH wireless LAN (of the IEEE 802.11 standard). The wideband DS interference is considered to have a sinc/sup 2/() shape psd centered at the middle of FH signal band and tone interferers are assumed to be spread over the whole FH band of the intended signal. Assumptions of slow non-selective fading along with slow hopping justify a quasi-static treatment. Results shows that the FH system is more sensitive to tone interference and fading and less sensitive to wideband interference variations. The paper sheds light on the performance of wireless LAN operation in the unlicensed free band where many other LANs may exist.

This paper investigates the performance of a new adaptive call admission policy based on a window-measurement estimation of the status of the buffer at the base station where two classes of customers (voice and data) are multiplexed into L servers. In this work, we interrelate physical limitations of the base stations (i.e., the number of transmission and reception modems), call and burst level traffic, instantaneous ATM buffer conditions, and end-to end bit error performance. Considering these parameters, we combine them all in one queueing problem and devise a new hybrid congestion based call admission policy. This policy is investigated in conjunction with CDMA integrated networks utilizing the IS-95 transmission/reception platforms. The results show that the proposed policy does maintain the QoS required for the networks under expected traffic load.

In ATM networks, fixed-length cells are transmitted. A cell may be discarded during the transmission due to buffer overflow or detection of errors. Cell discarding seriously degrades transmission quality. This paper analyzes a hybrid automatic repeat request/forward error control (ARQ/FEC) cell-loss recovery scheme that is applied to virtual circuits (VCs) of ATM networks. FEC is performed based on a simple single-parity code, while a Go-Back-N ARQ is employed on top of that. Both throughput efficiency and reliability analysis of the hybrid scheme are presented. In the process we investigate the interactive effects of the network parameters (number of transit nodes, traffic intensity, ARQ packet length, ...) on the performance. The analysis provides a method for optimizing the FEC code size for a given network specification.

A new multilayer error control technique is presented and analyzed. As the cell travels the inter-network enroute to the destination ATM user node interface (UNI), the CRC parity is checked and the cell is dropped or passed to the next ATM hop accordingly. A standard go-back-N ARQ technique is applied end to end while a lost-cell concealment technique cooperates below as a forward error correction mechanism that also operates end to end. The error detection and correction techniques above are applied to connectionless oriented traffic. The hierarchy of the error control technique above is presented and the end-to-end user performance in terms of net throughput and reliability is analyzed. We investigate the interactive effects of the channel and ATM signalling parameters on the system performance, with particular emphasis on the design of the error control scheme.

The selective repeat (SR) transport user end-to-end throughput and delay characteristics are analyzed in a hybrid asynchronous transfer mode/time division multiple access (ATM/TDMA) network. The SR transport user is assumed with an underlying cell-level global congestion control in the ATM multiplexer node. The analysis is based on the interaction of packet level control with the queue management at the ATM cell level. The end-to-end transport users are assumed over an M-node virtual circuit (VC) to derive the throughput and delay using Norton equivalent queueing model. The transport service characteristic of the model is obtained from the end-to-end protocol efficiency. The ATM layer is assumed with a leaky bucket (LB), virtual leaky bucket (VLB), modified LB (mLB), or modified VLB (mVLB) congestion control. A global congestion control scheme prioritizes transit traffic over local traffic and ensures the quality of service (QOS) to several classes of traffic. Based on global congestion status, the transport users modulate their end-to-end flow control parameters, i.e., packet size for video and voice users and window size for data users. The probability of cell-loss at the ATM layer is reflected at the transport layer to derive the effective throughput and delay characteristics. The mVLB scheme consistently provided better end-to-end throughput and delay performance for SR transport user.

We pursue a performance analysis for computing the various performance criteria in a hybrid time division/asynchronous transfer mode/code division multiple access network, i.e. TDMA/ATM/CDMA network. Users accessing this TDMA/ATM/CDMA uplink frame are assumed to belong to one of 4 service classes, namely video, voice, file and interactive data. Each user accesses only a portion of the subframe slots assigned to its class. A variable frame boundary strategy is used to adjust the subframe boundaries depending on the call load. To alleviate congestion in the assumed hubless signalling free satellite network, the satellite measures the uplink traffic of each class and issues pilot congestion control indicators to on-going calls of each class. These will be subsequently used by ground users to control their activities and police their calls using modified versions of leaky bucket and virtual leaky bucket congestion control techniques. The new techniques alleviate many of difficulties of specific slot assignment, onboard call management, superframe counting and management involved in the state of the art TDMA based systems, and yield a call establishment free yet dynamic and very well controlled access technique.

The new concepts of adaptive time hopping and variable frame code division (CDMA) multiple access are introduced. By a unified analysis, the probabilities of bit and packet errors in multipath fading environment for five time division (TDMA), code division, and time hopping (TH) related multiaccess networks are obtained; namely, TDMA, CDMA, CDMA/TDMA, Adaptive CDMA/TH, and variable frame CDMA/TDMA networks. The delay and useful throughputs of the five systems are also evaluated for data and voice traffic. All systems compared have the same channel power and bandwidth and support the same traffic. Though implementation issues are not covered, CDMA systems are put at a disadvantage (compared to cellular-type FDMA networks, for example) by ignoring such inherent advantages as voice silence utilizations and automatic frequency reuse. Nontheless, two CDMA systems outperform TDMA systems at low and medium input traffics.>

This special issue covers the topics: power control, multiuser receiver and joint detection, network aspects, adaptive equalisation, coding aspects, broadband CDMA systems, and optical systems.>

In Elhakeem and Rahman (1988) a new spread spectrum (SS) system was proposed that does not have a separate acquisition mode of operation. The system uses random frequency hopping, Gold codes, two receiver banks, and a combined acquisition/coding technique so as to eliminate the acquisition stage of operation. In the present paper, an algorithmic detection scheme is proposed using two sets of receivers that combines acquisition and demodulation simultaneously for the said system. The authors show by analysis the merits of the new technique and compare it against a conventional frequency hopping multi-access system that needs non-overlapping code acquisition mode for its operation. The throughput efficiencies of the conventional and the new acquisitionless system are compared in the case where Reed Solomon codes are used.>

This paper presents a concatenated coded modulation scheme for error control as applied to a spread spectrum multiple access system. A system model is proposed that allows one to apply modulation code (block) and a pseudo-noise (PN) spreading sequence to the data symbols to be transmitted. Gold sequences are employed for the purpose of PN spreading. Some good inner codes were chosen from Kasami (1990). Performance of those codes in an additive Gaussian noise channel is investigated with 5-30 users transmitting simultaneously. The performance of the system for coded 8-PSK were compared with uncoded QPSK under similar environment. The coded modulation scheme showed significant performance improvement with little or no bandwidth increase. The short modulation code is used as the inner code and a relatively powerful Reed-Solomon (RS) code is used as the outer code. Soft decision Viterbi decoding can be used for inner code. The proposed scheme is attractive for high speed satellite communications for file transfer applications where great reliability is desired.>

A performance analysis to compute the packet loss, call blocking, and packet delays of a typical user in an integrated voice-data-video satellite internetworking environment is discussed. The uplink technique used is a hybrid packet/circuit switched approach of the demand assignment type, while the downlink is a time-division-multiplexing (TDM) technique. Onboard the satellite, a baseband nonblocking switch is used to route the packets from input to output ports. Various amounts of input and output buffering as well as priority rules and blocking resolution algorithms are used. The authors conduct a performance analysis for the problems at hand and identify the best ranges for the different parameters involved.>

The authors prove by analysis the possibility of gaining a minimum of 2 dB in signal-to-noise ratio by just, splitting the users to three or more groups and identifying each one by an orthogonal waveform (on top of his short Gold code) in a PSK/DS spread-spectrum network. User signals may arrive in a code-asynchronous fashion at the receiver, however, it is shown that by using the scheme the average code cross-correlation is minimal compared to the classic code division multiple access (CDMA) system. Both the chip-synchronous and asynchronous cases are investigated and the uniform and optimal cases of dividing the users into orthogonal groups are analyzed. The superior bit error and network data throughput results in the different fading and forward error correction environments make the system a strong candidate for competitive domestic applications.>

A new analog speech scrambling technique that can be looked upon as processing the analog signal in two dimensions of time and frequency is introduced. A parallel bank of time-variant filters and a dither random pulse amplitude signal, both generated from multinonlinear direct-sequence spread-spectrum codes, operate on the analog signal to destroy its intelligibility. At the receiver, a reversed sequence of time and frequency processing takes place; with appropriate synchronization through a preamplifier, the analog received signal is recovered, with a minimum amount of distortion. The building blocks are introduced, and a statistical analysis is conducted to evaluate the security of the scrambling system, in terms of the autocorrelation and cross covariance. The security of the hybrid system is compared to that of other one-dimensional scrambling schemes, and derivations thereof.>

A spread-spectrum system that does not have a separate state for initial code acquisition is presented. A uniform random variate selects one of several Gold codes for transmission, thus removing the notion of pseudorandom codes from spread-spectrum systems, making the effective code length infinite, and leading to acquisitionless systems. Because coding is critical to the proposed multiaccess frequency-hopping (FH) system, more powerful codes are needed. The performance of the proposed multiaccess system using the combined modulation and coding technique (trellis) concatenated with Reed-Solomon codes in partial-band jamming is investigated. The FH multiaccess performance of noncoherent soft detection of MFSK in association with trellis coding is introduced and the performance compared to that using RS outer/RS inner concatenated codes.>

Two forms of fast circuit switching are modeled and studied through mathematical analysis and Monte Carlo simulation. The two examples described use the switching technique in an optical-fiber-based local area network with a star topology. The technique is compatible with time-division multiplexing techniques that are used for a range of traffic classes. Further, the technique trades transmission capacity for processing power, which is the critical limitation of the system. By means of a certain independence assumption, the first form of fast circuit switching is modeled as an M/G/1 queue. The results of the analysis show excellent agreement with simulation. The general result is that there is good system throughput, despite simplicity of processing. The second form, in which rather than first-come first-served the discipline is to search for a message whose destination queue is free, is studied by means of simulation alone. The results indicate an improved performance with a modest increase in processing power.>

The probability of bit error is evaluated for a communication system using frequency hopping for jamming rejection and quadrature partial-response signaling for baseband modulation. The environment in which the system is operating is modeled as severe partial-band tone jamming under frequency-selective fading affecting both the signal and the jammer. Error-correction coding is used to improve the performance of the system.>

A technique for data transmission over a CATV network, structured in a two-hop configuration, is presented. The technique combines the use of random access over the outer hop (distribution branches) and spread spectrum over the inner hop (main trunk) to maximize the CATV network throughput. A model is constructed for the spread spectrum access strategy. The throughput-delay relationship is obtained as a function of several design parameters. Numerical results are presented and discussed based on the assumptions and limitations of the model.

The various issues relating to the use of direct-sequence (DS) spread-spectrum (SS) modulation as access protocols for a local computer network are investigated. Voice and data messages having different traffic characteristics are packetized and the information bits DPSK modulated, after which spreading by the DS gives the waveform to be transmitted. Each node within the network will have a SS modem, to which all kinds of voice and data sources of different rates can be connected. Each node will be characterized by a certain DS code, thus getting security as a fringe benefit. Except for the code circuit, all SS modems will look the same; moreover, all the DS signals transmitted will have the same spectral properties, thus achieving uniformity and modularity of signals and equipments throughout the network. Among the performance criteria, we choose the signal-to-noise ratio (SNR) at the demodulator, the packet delay, the probability of correct packet detection, and the useful throughput. A new definition is suggested for the network efficiency and all performance criteria for the proposed network are compared to those of a classic FDMA/ALOHA-type network under the same circumstances.

New techniques for acquiring the codes used in frequency hopping (FH), hybrid frequency hopping/time hopping (FH/TH), and frequency hopping/direct sequence (FH/DS) spread-spectrum systems are presented. Autoregressive spectral estimation is employed to recover the FH code, quadrature processing is used for the DS code, while the TH gating code is recovered by a simple threshold test used in conjunction with the adaptive filter used for the spectral estimation. A wide margin of spectral estimation errors can be tolerated since an acquisition logic is used to estimate the FH code phase. The proposed techniques require acquisition times only of the order of the shift-register generator lengths, whereas other schemes of serial and parallel search require acquisition times of the order of the code lengths. Detailed simulation results will identify the working carrier-to-noise ratio (CNR) range for the different techniques presented as well as the various thresholds involved.