In this article, we analyze the performance of the long-term evolution random access procedure with the Third Generation Partnership Project's access class barring (ACB) mechanism in an energy harvesting (EH) machine-to-machine (M2M) scenario. To circumvent the state-space explosion in the conventional Markov-chain-based analysis due to time-dependent traffic pattern and data and energy buffer status, we develop an analytical model that combines mean-value analysis with the Markov-based analysis. Based on the analytical model, the random access success probability, the access delay of the network, and the average time duration between two successive successful transmissions are derived. Our analysis suggests that in the EH scenario, despite the lower number of the contending nodes in comparison with the non-EH scenario, the ACB parameters must be chosen in a more conservative way to avoid excessive collisions. The ACB parameters include access barring rate and mean barring duration. We also study an energy threshold-based activation policy and investigate the joint effects of this policy and the ACB mechanism on the random access success probability. The extensive simulations were conducted to evaluate the accuracy of the analytical model.

We consider uplink transmission using a single carrier frequency division multiple access (SC-FDMA) scheme in a cellular network with human-to-human (H2H) and machine-to-machine (M2M) communications. The M2M traffic is relayed through energy harvesting (EH) gateways. An optimization framework is developed to minimize the data dropping in the EH gateways caused by delay constraint violation of the M2M traffic while taking into account the rate requirements of the H2H communication, the data and energy causality constraints as well as the SC-FDMA transmission constraints. By introducing two transforms, the original problem is expressed in a linearly separable form in terms of its discrete and continuous variables with the convexified continuous part. Then, Generalized Benders Decomposition is applied to solve the problem by decomposing it into primal and master problems. Due to the NP-hardness of the optimal solution, a low-complexity method is proposed by combining the solution of the primal problem with a heuristic resource block allocation algorithm. Simulation results show that the proposed heuristic method performs better than two alternative heuristic methods when applied to small scale and large scale networks.

Single-radio mesh routers operating on a single channel suffer from low throughput due to collisions. Equipping mesh routers with multiple radios operating on non-overlapping channels can significantly improve the throughput. However, the assignment of channels to radios in a multi-radio mesh network is a challenging task. In this paper, we propose a channel assignment algorithm, TICA (Topology-controlled Interference-aware Channel-assignment Algorithm), which significantly improves network throughput by minimizing interference within the mesh network using a novel approach of controlling the network topology based on power control before intelligently assigning the channels to the multi-radio mesh routers, as well as guaranteeing network connectivity.

The design of a fully-integrated multiband tunable linearized CMOS active analog phase shifter with active loss compensation that has broadband signal transmission and provides multiband phase shifting and that allows for broadband distortion reduction is presented. The S21differential power gain peaks at 7 dB and then rolls off to a unity gain bandwidth of 11.5 GHz. The simulated phase S21shift tuning performance is better than 50 degrees at multiple bands of 2.4 GHz and 5.8 GHz. The simulated linearized phase shifter IIP3 show a 10 dB improvement. Simulation results show that the co-design of the active analog phase shifter with on-chip loop antenna in both transmit and receive path has phase S21shift tuning performance which is better than 50 degrees and is compared to a stand alone on-chip loop antenna S21phase performance. The CMOS active analog phase shifter was fabricated using the 0.13 mum CMOS technology and has a total silicon chip area of 1.887times0.795 mm2.

This paper presents a novel methodology for finding the network connectivity in wireless mesh networks while taking into account dependencies between links of geometrically co- located nodes, as well as the reduced connectivity of boundary nodes. A closed form expression to find the upper bound for the network connectivity of a triangular lattice topology was derived. A comparison study between models that assume independent nodes' connectivities and our model shows that the commonly used assumption of link independence almost always underestimate the network connectivity. This work also shows that the effect of boundary nodes on the network connectivity can not be eliminated by assuming an infinitely large network since the assumption overestimates the network connectivity by a non negligible amount. The analysis also shows that the error due to either assumption depends on the link connectivity as well as the network size, and can be very significant.

This paper presents a digitally controlled oscillator (DCO) with a novel architecture. The proposed DCO achieves low noise operation without the need for an LC resonator or specialized radio frequency (RF) process layers. Frequency tuning is achieved by controlling capacitive coupled charge injection. The design is easily integrated with digital CMOS circuits and has the flexibility to be scaled to other processes or tuned for specific applications. A proof of concept circuit has been implemented on a 0.13 mum CMOS process. The tuning range is from 820 MHz to 1.09 GHz. The phase noise at 1GHz with an offset frequency of 100 KHz is -92.8 dBc/Hz. The worst case cycle-to-cycle jitter is less than 1.28% with an RMS jitter of 1.7423 ps. Total power consumption is 6.12 mW.

In this work, the optimal network plan in terms of node class selection and number of nodes required to achieve a given network reliability in a wireless mesh network (WMN) was investigated. The network is said to be optimal if it minimizes the network implementation cost while achieving the application matrix requirements. In this work, these requirements are described in terms of communication range, minimum network reliability, and achievable throughput. A node class is generally characterized by its capabilities (such as transmitted power and sensitivity) and associated cost. In order to tackle this problem, we developed an analytical model and methodology for WMN optimization. The proposed model differs from previously used models in that it takes into account linkspsila dependencies of geometrically co-located nodes, the effect of boundary nodespsila connectivity, in addition to fading and shadowing effects. This methodology was used to find the structure of the optimal policy for two important topologies: the one-dimensional linear network topology, and the two-dimensional triangular lattice topology. The proposed methodology resulted in a qualitative and quantitative description of the optimal WMN planning policy.

This paper presents a novel methodology for finding the network connectivity in wireless mesh networks. This methodology takes into account link dependencies in neighboring nodes. The methodology was used to determine the upper bound for the connectivity in a one-dimensional chain topology as well as in the two-dimensional triangular lattice topology. We show that the assumption of link independence almost always underestimates the network connectivity. Our derived closed form expressions hence serve as more accurate upper bounds for network connectivity. The conducted comparison studies show that the error due to the assumption of node independence depends on the link connectivity, and can be very significant. The devised methodology can also be used to quantify the error and define the range of link connectivities within which the assumption of link independence can be used.

An 8-bit successive approximation analog-to-digital converter (ADC) with offset correction circuitry is presented for implantable sensor applications. The ADC is designed in a 0.13μm CMOS process technology and operates with voltage supplies down to 0.35V using MOSFETs operating in their sub-threshold region of operation. Sample rates of 60kS/s are achieved with an INL and DNL of approximately 0.26LSB and 0.35LSB respectively. The SAR ADC achieves 10.7pJ/cycle operating at 20kS/s with a 0.4V supply. An offset correction circuit is included to dynamically minimize the offset voltage on the comparator.

An 8-bit successive approximation analog-to-digital converter (ADC) for ultra low power applications is presented. It is designed in a standard 0.13mum CMOS process technology. The design can operate with low voltage supplies down to 0.45 V. It makes use of sub-threshold transistor operation to achieve nanowatts of power consumption at sample rates exceeding 60kS/s. A specially designed switch allows large input swings. Post layout simulations show an INL and DNL of approximately 0.3LSB and 0.45LSB respectively

Service Discovery Protocols (SDP) allow users and applications to automatically locate services without prior configuration. Since Mobile Ad Hoc Networks (MANETs) constitute an integral part of the ubiquitous computing scenario, increasing efforts are geared toward providing service discovery in such environments. In this work, we adapt the widely employed Service Discovery Protocols (SDP) Version 2 to MANETs. We implement and analyze the adapted protocol, SLPManet, using Network Simulator Version 2. Consequently, we propose a caching modification that increases the discovery success by 38% for poorly performing scenarios, as well as reduces the aggregate bandwidth consumption and average lookup latency by up to 98%.

In the world of wireless ad-hoc networking, the problem of service discovery is essential and important in order to detect and access services between mobile computers. This has hence led to the advent of numerous service discovery protocols (SDPs). We recognize the need for a legitimate framework for the qualitative and quantitative evaluation of SDPs in mobile networking contexts such mobile ad hoc networks (MANETs). This paper presents benchmark for MANET (BENCHManet), an initiative development of a comprehensive and sophisticated benchmark composed of reference tests, each of which reflects the configuration found in various realistic MANET applications. The presented benchmark allows for a more practical and comprehensive evaluation, as well as a fair comparison of SDPs

Abstract This paper proposes a transmission‐scheduling algorithm for interference management in broadband wireless access networks. The algorithm aims to minimize the cochannel interference using basestation coordination while still maintaining the other quality of service (QoS) requirements such as packet delay, throughput and packet loss. The interference reduction is achieved by avoiding (or minimizing) concurrent transmission of potential dominant interferers. Dynamic slot allocation based on traffic information in other cells/sectors is employed. In order to implement the algorithm in a distributed manner, basestations (BSs) have to exchange traffic information. Both real‐time and non‐real‐time services are considered in this work. Results show that significant reduction in the packet error rate can be achieved without increasing the packet delay at low to medium loading values and with a higher but acceptable packet delay at high loading values. Since ARQ schemes can also be used for packet error rate reduction, we compare the performance of the proposed scheme with that of ARQ. Results indicate that although ARQ is more effective in reducing packet error rate, the proposed algorithm incurs much less packet delay particularly at medium to high loading. Copyright © 2006 John Wiley & Sons, Ltd.

In this paper, we present a novel transmission scheduling scheme for fixed broadband wireless access (FBWA) networks. In order to mitigate high co-channel interference resulting from dense reuse of frequency, scheduling schemes often consider interference management issues as an essential part. To that end, a series of literature has been published recently, where a group of base stations form an interference group, and scheduling scheme deployed in the group allows only one base station to transmit at a time. As a result of time orthogonality in transmissions, the co-channel interference and hence the packet error rate can be reduced. However, prohibiting concurrent transmissions in these orthogonal schemes introduces higher end-to-end packet delay which might not be desirable for real-time traffic such as voice and video. Our proposed scheme aims to improve delay performance as well as to ensure better resource utilization in such schemes in several ways, most notably by introducing opportunistic non-orthogonality in transmissions, incorporating channel state based scheduling decisions, and employing adaptive modulation and coding.

This work analyzes the impact of having more than one frequency channel on outage probability and node throughput in TDMA-based broadband fixed cellular multihop networks. The scenario considered is a low node density, start-up network with inadequate single-hop coverage. In this scenario the user terminals themselves act as relays, therefore, there are no additional relays deployed by the service provider: besides, no separate relaying channels are used (this scenario ensures a fair comparison between the multihop cellular network considered here and the traditional single-hop cellular network). In a situation where there is outage with a single-hop due to poor coverage, a suitable multihop route with minimum number of hops is sought. It is observed that as the number of available channels increases the coverage is enhanced remarkably; in addition, the average node throughput is increased as well although no explicit attempt for throughput maximization is made during the search for the multihop routes.

This paper proposes a joint adaptive rate selection and power control algorithm for broadband TDMA/TDM wireless networks. The proposed algorithm is a modified, enhanced, and more robust version of the selective power control with active link protection (SPC-ALP) algorithm proposed for adaptive transmission rate and power control in CDMA networks [R.Jantti and S. Kim, 2000]. Unlike SPC-ALP, the proposed algorithm uses adaptive coding and modulation (instead of variable spreading gain) for transmission rate control. In addition, the proposed algorithm is different in two aspects; the first one is the inclusion of the cochannel interferer assistance mode, while the second one is the use of the signal quality as a criterion for user removal. Results show that the proposed algorithm outperforms SPC-ALP in terms of the net throughput and signal quality measures such as the outage probability and frame error rate. Channel reallocation is also studied and found to be very effective in enhancing the system performance particularly at low to medium loading.

In this paper, we attempt to enhance the performance of W-CDMA downlink for third generation (3G) land mobile satellite communication systems for Web traffic. It has been shown that Web traffic requires very low bit error rate (BER) in order to avoid severe degradation in its throughput over satellite links. In Hijres et al. (1999), it has been shown that power control errors can seriously degrade the performance of the downlink in W-CDMA mobile satellite systems. Hence, we focus on the combined effect of using the radio link protocol (RLP) and macrodiversity (space diversity) to improve the performance of turbo coded downlink under power control errors for Web traffic. The macrodiversity is employed at the mobile Web terminal rather than at the satellite, due to satellite space limitation. Simulation results demonstrate that the combined effect of using RLP and macrodiversity is essential in reducing the frame error rate to a very low level in order to support Web traffic over W-CDMA mobile satellite links. This very low frame error rate has been achieved at very low Eb/No, e.g. FER /spl ap/10/sup -5/ has been achieved at Eb/No=1 dB for downlink which suffers from large power control error with standard deviation (/spl sigma//sub s/) of 4 dB. It is extremely important to operate downlinks of mobile satellites at very low Eb/No in order to preserve satellite capacity, as mobile satellites are power limited.

The transmission of MPEG-4 video over explicit rate networks provides an opportunity for complete interaction between the application and the network where the application can request the required bandwidth from the network and then adapts its rate based on the network offered rate. One method of rate adaptation is to encode video in multi layers, were the basic layer is always transmitted and the enhanced layer is only transmitted when enough resources are available. In this paper, we propose a novel intelligent VOP discard algorithm (IVDA) that controls the transmission of enhanced layer VOP The proposed algorithm depends on the knowledge of the current network state, the encoding sequence, nature of the various VOP types, and the buffer occupancy in order to control the transmission of the enhanced layer VOP The proposed algorithm was evaluated for various MPEG-4 test sequences with different traffic characteristics under various network conditions.

This paper presents a robust algorithm for voice activity detection (VAD) based on newly established properties of the higher order statistics (HOS) of speech. Analytical expressions for the third and fourth-order cumulants of the LPC residual of short-term speech are derived assuming a sinusoidal model. The flat spectral feature of this residual results in distinct characteristics for these cumulants in terms of phase, periodicity and harmonic content and yields closed-form expressions for the skewness and kurtosis. Important properties about these cumulants and their similarity with the autocorrelation function are revealed from this exploratory part. They show that the HOS of speech are sufficiently distinct from those of Gaussian noise and can be used as a basis for speech detection. Their immunity to Gaussian noise makes them particularly useful in algorithms designed for low SNR environments. The proposed VAD algorithm combines HOS metrics with second-order measures, such as SNR and LPC prediction error, to classify speech and noise frames. The variance of the HOS estimators is quantified and used to yield a likelihood measure for noise frames. Moreover, a voicing condition for speech frames is derived based on the relation between the skewness and kurtosis of voiced speech. The performance of the algorithm is compared to the ITU-T G.729B VAD in various noise conditions, and quantified using the probability of correct and false classifications. The results show that the proposed algorithm has an overall better performance than G.729B, with noticeable improvement in Gaussian-like noises, such as street and parking garage, and moderate to low SNR.

The system capacity of TDMA systems can be enhanced by utilizing the slow frequency hopping feature. More capacity gain can be achieved if interference reduction techniques such as smart antennas and discontinuous transmission are used. This paper addresses the capacity gain in the downlink due to the use of smart antennas in an FH-TDMA cellular system. We estimate the system capacity analytically and by simulation. Multiple switched beam smart antennas are utilized in the study. Perfect power control and discontinuous transmission (using voice activity) are assumed in the simulation and in theoretical analysis. The TDMA system is assumed to be GSM-like. However the analysis is valid for any TDMA system.

The transmission of real-time MPEG-4 video presents network designers with various challenges primarily because of the lack of advance knowledge of its real-time traffic characteristics and in the face of stringent QoS requirements. In order to carry out real-time MPEG-4 video, we propose a novel source architecture that combines traffic shaping, rate shaping, early VOP dropping, and utilizing the feedback information as conveyed by the network through the B-RM cells of the ABR service. The proposed scheme is evaluated using an ATM/ABR network simulator and a real video trace encoded into two layers according to the MPEG-4 standard. The simulation results presented in this paper demonstrate that the proposed architecture is capable of protecting the basic layer and satisfying the delay bound as specified by the application. Furthermore, the proposed scheme is capable of protecting the video quality from any sudden drops in the network bandwidth, yet still providing a graceful degradation if the congestion condition persists.

A new method for speech enhancement based on optimal filtering, subbands, and higher-order cumulants is proposed in this paper. The key idea is to use the 4th cumulant to estimate the parameters required for the enhancement fil-ters: It is shown that the kurtosis of noisy speech may be used to estimate the SNR and the probability of speech presence when speech is divided in narrow bands and mod-eled as a sinusoidal signal. The resulting algorithm is tested in typical mobile noise conditions and proves effective under such types as street, office and fan noises. Compared to the TIA IS-127 standard for noise reduction, the proposed algorithm is better at preserving the harmonic structure of the speech and results in overall more noise reduction in Gaussian-like conditions. However, this comes at the cost of slightly more noise artifact, mostly at very low SNR and non-Gaussian conditions. 1.

This article addresses the problem of instantaneous signal-to-noise ratio (SNR) estimation during speech activity for the purpose of improving the performance of speech enhancement algorithms. It is shown that the kurtosis of noisy speech may be used to individually estimate speech and noise energies when speech is divided into narrow bands. Based on this concept, a novel method is proposed to continuously estimate the SNR across the frequency bands without the need for a speech detector. The derivations are based on a sinusoidal model for speech and a Gaussian assumption about the noise. Experimental results using recorded speech and noise show that the model and the derivations are valid, though not entirely accurate across the whole spectrum; it is also found that many noise types encountered in mobile telephony are not far from Gaussianity as far as higher statistics are concerned, making this scheme quite effective.

Given a quasi-cyclic code with minimum Hamming distance d, a set of convolutional codes is derived with free distance equal to d. It is shown that an increase in the rate of these codes results in a decrease in the memory length. The connection between these codes is illustrated with several well-known quasi-cyclic codes. The free distance of some partial unit memory convolutional codes can be determined using the results in this correspondence.

A new class of protocols like RTP, RTSP and RSVP are tailoring today's packet networks for streaming real time multimedia data that once carried only non real time traffic. Multimedia networking is now possible with the existing network infrastructure. We describe a real time multimedia presentation system for media on demand over the Internet. It uses RTP and is capable of real time video/audio transmission. Our system consists of servers that maintain video/audio clips and clients that retrieve and present them to the users. It provides VCR like control features for user interactivity during playback. The paper describes the system architecture, and the communication protocols, along with media synchronization and dynamic QoS control mechanisms.

The design and performance of a high data rate modem that transmits asynchronous transfer mode (ATM) packets over indoor radio channels is discussed. A discrete Fourier transform (DFT)-based multicarrier modulation technique is used to mitigate intersymbol interface (ISI) caused by a multipath spread of up to 250 ns. A rate one-half channel code is used to combat fading. Computer simulation is performed to investigate the system performance for five different multipath intensity profiles. The system performance is compared using a differentially coherent scheme and a coherent scheme based on channel estimation. The effects of carrier frequency offset and some of the hardware nonlinearities are discussed. Using various channel codes, an signal-to-noise ratio (SNR) of 14-21 dB is required at an average bit error rate (BER) of 10/sup -5/ to transmit a total of 155 Mb/s data over a bandwidth of 280 MHz.

An adaptive polling scheme for multiple access in a broadband wireless local area networks (LANs) is proposed. It is designed to accommodate asynchronous transfer mode (ATM) traffic with a large and variable range of bit rates. The radio architecture is microcellular, with switched sectored-beam antennas and a small frequency reuse factor. Features of the multiple access scheme include polling rate depending on a terminal's average bit rate, fixed-length frames divided into segments corresponding to base antenna sectors, and automatic-repeat-request (ARQ) for error control, combined with randomized sector segment ordering. The steady-state performance (cell loss, call blocking and dropping probabilities, waiting time, and buffer occupancy) are evaluated by simulation for an open indoor environment, and for handoff, ARQ, and simple power control enhancements; focusing mainly on the effects of the propagation and interference conditions.

Presents predetection and postdetection combining schemes for selection diversity reception with multiple antennas for MM-wave indoor radio channels. For those combining schemes, a reduction in complexity is achieved by limiting the number of combined signals to small values and by increasing the number of received signals. Bit error rate (BER) performance of binary phase shift keying (BPSK) with predetection combining of selected signals (CSS) and BER performance of differential BPSK with postdetection CSS are analyzed for slow fading and Rayleigh-distributed envelope statistics. Predetection maximal ratio combining of signals that comes from a single group or several groups of diversity channels as well as postdetection combining of received signals for groups of channels are considered. In comparing predetection combining with groups (PCG) and predetection combining of the best signals (PCB), we observe that the required SNR for achieving a certain BER is approximatively the same (with PCG having a slight advantage of 0.5 dB) for a given number, N, of diversity channels and L combined signals. Furthermore. PCG is equivalent to PCB for L=N since both techniques then correspond to conventional predetection maximal ratio combining (MRC), PCG and PCB are also equivalent when L=1 as both schemes then correspond to conventional selection combining. A small degradation of approximately 2 dB in the required SNR is observed when postdetection diversity reception with groups (PDG) is used instead of PCG. For L=N, PDG reduces to post detection MRC. The PDG technique is considered more suitable than PCB or PCG for MM-indoor wireless systems.

An adaptive polling scheme for multiple access in a broadband wireless local area networks (LANs) is proposed. It is designed to accommodate asynchronous transfer mode (ATM) traffic with a large and variable range of bit rates. The radio architecture is microcellular, with switched sectored-beam antennas and a small frequency reuse factor. Features of the multiple access scheme include polling rate depending on a terminal's average bit rate, fixed-length frames divided into segments corresponding to base antenna sectors, and automatic-repeat-request (ARQ) for error control, combined with randomized sector segment ordering. The steady-state performance (cell loss, call blocking and dropping probabilities, waiting time, and buffer occupancy) are evaluated by simulation for an open indoor environment, and for handoff, ARQ, and simple power control enhancements; focusing mainly on the effects of the propagation and interference conditions.

Indoor wireless communications will become important in providing access to networks which will increasingly use ATM technology to allow multimedia service applications. This paper presents two possible architectures for wireless access to an ATM LAN: the centralized and the distributed wireless controller cases. The detailed signaling procedures for both the cases are discussed, and the two architectures are compared in terms of their signaling and complexity implications.

The paper presents a low complexity, low delay variable rate speech coding method suitable for low power (CDMA or TDMA) indoor wireless communications. The indoor wireless communications channel is characterized by relatively deep fades resulting in bursty errors. A robust speech codec is required to combat this bursty nature. Low power operation is essential, which negates the applicability of LD-CELP (G.728) for indoor wireless applications, Historically, ADPCM (G.721) has been proposed, which lacks robustness to random and bursty errors and has a relatively high bit rate (32 kb/s). To circumvent the need for relatively complex echo-cancellation, low end to end delays are required which implies that interleaving is not applicable. In the paper, a low complexity, low delay, variable rate CELP algorithm is proposed and techniques to improve the robustness to bursty errors are investigated. The new technique is compared to ADPCM on the basis of complexity and performance (MOS) over measured indoor wireless channels at 1.7 GHz with various combinations of space, frequency and code diversity.>

32 kb/s adaptive differential pulse coded modulation (ADPCM) is a widely accepted high quality speech coding algorithm. Single chip devices exist for 32 kb/s ADPCM. This algorithm requires a relatively large bandwidth and does not perform exceptionally well in noisy wireless environments. The 16 kb/s low-delay code excited linear prediction (LD-CELP) coding algorithm does perform well in noisy environments and requires a reduced bandwidth. Through algorithm optimization and simulated verification of mixed analog and digital VLSI partitioning a single chip implementing the 16 kb/s LD-CELP voice coding algorithm will be possible. The performance of this coder chip will be comparable to the quality of 32 kb/s ADPCM. In this paper we propose the use of a link between Comdisco's SPW and Synopsis to determine the hardware related performance degradation and the technology/complexity related power dissipation and area parameters. It will both prove the concept of design and determine the VLSI implementation feasibility.>

The six-state model for on-off characteristics of conversational speech established by P.T. Brady (1969) does not consider pauses in speech shorter than 200 msec. This paper proposes modifications to the Brady model to represent the effects of the short pauses while preserving the effects of the longer silences and the dynamics of the interaction between speaking parties. The modified Brady model provides a tool for more accurately assessing the performance of new-generation wireless communication systems, which will utilize the short silence pauses as well as the longer periods of silence.>

In order to make the analytical evaluation of portable radio link performance tractable while avoiding lengthy simulations, intersymbol interference and cochannel interference are often modeled as non-fading, Gaussian noise. This paper investigates the viability of such an approach for wideband channels where equalization and space diversity are used to combat interference and signal fading. Comparisons of results obtained using such an assumption of interference with those of a simulation-based approach indicate that as long as the number of cochannel interferers is smaller than the space diversity order, the variance of the Gaussian interfering symbols can be adjusted to provide a reasonable estimate of performance over error rates of interest.>

The direct provision of connectionless service in BISDN calls for servers that are connected to or are part of an ATM network to provide the routing function at input speeds up to 622 Mb/s. Routing is achieved in such a server by changing the VCI/VPI headers in the ATM cells; actual switching is done by existing switches in the ATM network. The paper presents an architecture capable of executing all the functions of a server at input speeds up to 622 Mb/s, scalable to multiple inputs at that speed, making use of processors and special hardware that are available today. To avoid storing large quantities of data, the architecture routes data packets by examining routing information in the initial cell of the packet and routing subsequent cells as they arrive rather than waiting until the complete packet has arrived. It is capable of handling packets that have been multiplexed at the SAR sublayer using AAL Type 3/4 and, with minor modifications, could also handle Type 5 traffic. Arguments are also presented for the use of AAL Type 5 for the direct connectionless service.>

A tree-searched multistage vector quantization (VQ) scheme for linear prediction coding (LPC) parameters which achieves spectral distortion lower than 1 dB with low complexity and good robustness using rates as low as 22 b/frame is presented. The M-L search is used, and it is shown that it achieves performance close to that of the optimal search for a relatively small M. A joint codebook design strategy for multistage VQ which improves convergence speed and the VQ performance measures is presented. The best performance/complexity tradeoffs are obtained with relatively small size codebooks cascaded in a 3-6 stage configuration. It is shown experimentally that as the number of stages is increased above the optimal performance/complexity tradeoff, the quantizer robustness and outlier performance can be improved at the expense of a slight increase in rate. Results for log area ratio (LAR) and line spectral pairs (LSPs) parameters are presented. A training technique that reduces outliers at the expense of a slight average performance degradation is introduced. The method significantly outperforms the split codebook approach.>

The authors present a tree searched multi-stage vector quantization (TS-MSVQ) scheme which achieves spectral distortion lower than 1 dB with low complexity and good robustness using 24 b/frame. The M-L search is used and it is shown that it achieves performance close to that of the optimal search for a relatively small M. The best performance/complexity trade-offs are obtained with relatively small size codebooks cascaded in a three-four stage configuration. Results for log-area ratio (LAR) and line spectral pain (LSP) parameters are presented. A training technique which reduces outliers at the expense of a slight average performance degradation is introduced. The robustness across different languages and input spectral shapings is studied. Finally, it is shown that TS-MSVQ significantly outperforms the split-codebook approach.>

Compound strategies of equalization and space diversity in the form of an optimum baseband combiner are attractive for wide-band time division multiple access (TDMA) portable communication radio links in order to combat dispersive fading and cochannel interference. This paper investigates the performance of such a scheme in conjunction with convolutional coding and soft-decision Viterbi decoding via a semi-analytical technique based on the method of moments. Such an approach avoids a Gaussian characterization of interference and yields results for both ideal interleaving and no interleaving. With dual space diversity, three taps per forward filter and a data rate of 10 Mb/s, it is shown that, although a third space diversity branch remains preferable in terms of performance, channel coding can be a viable alternative, particularly in terms of outage rate, to increasing the space diversity order even in the absence of interleaving provided the signal-to-interference ratio is sufficiently high

A system model is proposed that allows one to apply both trellis coding and PN spreading sequence to the data symbols to be transmitted. Rate n/n+1, trellis codes using 2/sup n+1/-point MPSK signal constellations are investigated when Gold sequences are used for PN spreading. Performance in an additive white Gaussian noise (AWGN) channel is investigated, with 5-20 users transmitting simultaneously. Using the criteria of equal complexity and throughput, the performance of the trellis codes in a SSMA (spread spectrum multiple access) environment is compared to that of medium-rate to low-rate convolutional codes through the use of a generalized transfer function bound. The average degradation due to the interuser interference is determined by the method of moments. The validity of approximating the interuser interference as a Gaussian random variable is also investigated. The numerical results illustrate that for a given complexity, chip rate and throughput, low-rate convolutional codes provide the best performance in an SSMA system. As lower-rate convolutional codes are used, there is an increase in the effective interuser interference due to the greater cross-correlation effects from using shorter PN sequences, or alternatively from the effects of partial cross-correlation. However, this increased degradation is more than overcome by the increased distance properties of the low-rate codes.>

The results of temporally and spatially distributed wideband (impulse response) propagation experiments in the 900 MHz and 1.7 GHz radio frequency bands in two different buildings on fixed indoor radio links are reported. Results from the temporal experiments show that, for a specific location in either of the two buildings, the dynamics of indoor channels are slightly less random at 910 MHz than at 1.7 GHz. It is believed that this would result in marginally better performance on a given transmit/receive link in the 900 MHz band. The spatially distributed measurements showed that the structures of average impulse-response envelopes differed for channels in the two buildings. In one building, RMS delay spreads were slightly greater in the 1.7 GHz band for over 90% of transmit/receive link configurations. In the other building, RMS delay spreads were marginally greater in the 900 MHz band for 70% of the configurations.>

An examination is made of the performance of a squaring loop and a fourth-power loop used for tracking the received carrier phase of binary continuous-phase modulated (CPM) signals with index 1/2. Noise-free analysis is performed first and a simple relationship between the phase error and the control voltage is obtained for each case, with the premodulation filter response as a parameter. Linearized analysis of the phase error variance is then conducted in the presence of additive white Gaussian noise. It is shown that the phase variance in both cases contains a self-noise component that sets a lower bound on the synchronizer's performance at very high signal-to-noise ratios. Numerical results are presented for a number of popular CPM signals. It is found that the squaring loop performs best with MSK (minimum shift keying), while the fourth-power loop performs best with duobinary MSK.>

Detection of continuous phase modulation (CPM) signals can be accomplished using quadrature coherent detectors which include a pair of linear and time-invariant postdetection filters. Performance of the quadrature detectors is highly sensitive to the postdetection filters response. This paper presents a rigorous derivation of an optimum postdetection filter response. The derivation is based on minimizing the mean-square interference subject, to the constraint that the filter noise bandwidth is held constant. The amount of computations involved is fairly small and increases linearly with the receiver observation interval. Performance analysis results for several modulation techniques are presented.

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.

This paper presents the basic architecture and performance of a mobile radio multiaccess voice/data system. Natural pauses in conversational speech allow bandwidth saving through interleaving of data packets and talkspurts from different voice sources. A speech detector designed specifically for the mobile environment is presented. Blocking and delay performance of the multiaccess uplink is analyzed for voice traffic, assuming no traffic effects from the low priority data packets. Performance results from simulation are then presented for two downlink strategies in a two-hop virtual circuit in which a base station acts as a relay. The results verify also that the uplink analysis is valid for low voice traffic. For the data traffic, simulation results are presented in terms of data packet transmission delay and probability of collision with talkspurts. The results indicate that data flow may be limited by the collision factor. This work concludes that relative to conventional radio telephoning in which two channels are dedicated to each transmitter/receiver pair, a bandwidth reduction of 30-35 percent can be achieved.

A mathematical model is developed to determine the probability of successful transmission of a packet over a fading land mobile data channel.

A novel approach to modeling the slotted-ALOHA random access channel is presented. The objectives of the model are threefold: (1) to identify the channel's stability boundaries analytically as a function of the system's independent input parameters; (2) to investigate the impact of buffer-control on the channel's behavior; and (3) to facilitate the structuring of individual users' buffer-control as the focus of a dynamic control scheme. The significance of maintaining channel stability via buffer-control lies in its conceptual simplicity and implementability. Simulation runs of a finite-population model have clearly revealed the variation in individual users' buffer sizes and their sensitivity to the retransmission procedure.

Access control to shared files in a distributed computing environment requires an efficient method of allocating file resources with local and remote user processes. While software controlled access methods are convenient from the user's point of view, they give rise to serious operational problems such as job interferences (deadlock situations) and critical race conditions. Two software controlled access schemes, one centralized and one distributed are described in this paper. A basic set of protocols is derived to illustrate the implementation aspects of each scheme in a given network environment. The two approaches are then evaluated in the light of several performance criteria.

The problems of file allocation and capacity assignment in a fixed topology distributed computer network are examined. These two aspects of the design are tightly coupled by means of an average message delay constraint. The objective is to allocate copies of information files to network nodes and capacities to network links so that a minimum cost is achieved subject to network delay and file availability constraints. A model for solving the problem is formulated and the resulting optimization problem is shown to fall into a class of nonlinear integer programming problems. Deterministic techniques for solving this class of problems are computationally cumbersome, even for small size problems. A new heuristic algorithm is developed, which is based on a decomposition technique that greatly reduces the computational complexity of the problem. Numerical results for a variety of network configurations indicate that the heuristic algorithm, while not theoretically convergent, yields practicable low cost solutions with substantial savings in computer processing time and storage requirements. Moreover, it is shown that this algorithm is capable of solving realistic network problems whose solutions using deterministic techniques are computationally intractable.