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This paper presents indoor and field experimental results on extremely high-speed packet transmissions of 4.92 Gbps in a 100 -MHz channel bandwidth, i.e., the frequency efficiency of approximately 50 bit/second/Hz, in the downlink OFDM radio access. The required received signal-to-noise power ratio (SNR) is less than 30dB, which is almost the upper limit in cellular environments near a cell site even with a light channel load, with the aim of application to future universal broadband packet radio access. We apply 12-by-12 MIMO multiplexing using the maximum likelihood detection employing QR decomposition and the M-algorithm (QRM-MLD) with adaptive selection of the surviving symbol replica candidates (ASESS),64QAM data modulation, and Turbo coding with the coding rate of R = 8/9 to achieve an extremely high frequency efficiency level. In the field experiments conducted in the Yokosuka Research Park (YRP) district of Yokosuka city, we show that the measured throughput of 4.92 Gbps is achieved at the average received SNR per receiver antenna of approximately 28.0 and 28.5 dB when the respective receiver antenna separation is 40 cm (6.2lambda) and 10 cm (1.5lambda) and the transmitter antenna separation is 70 cm (10.9lambda) at the average speed of 10 km/h under non-line-of-sight (NLOS) conditions.

This paper presents experimental results on real-time packet transmission of greater than 1 Gb/s using 4-by-4 multiple-input-multiple-output (MIMO) multiplexing and maximum-likelihood detection (MLD)-based signal detection with a decreased level of computational complexity in orthogonal frequency-division multiplexing (OFDM) radio access. We apply our previous algorithm called adaptive selection of surviving symbol replica candidates (ASESS) based on the maximum reliability in MLD employing QR decomposition and the M-algorithm (QRM-MLD) to reduce the extremely high level of computational complexity in the conventional MLD. The experimental results using multipath fading simulators are in good agreement with the computer simulation results. The loss in the required received signal energy per bit-to-background noise power spectrum density ratio (Eb/N0) is suppressed to within approximately 1-2 dB. Therefore, through experiments, we demonstrate that the QRM-MLD employing ASESS is very beneficial in reducing the influence of hardware implementation loss, as well as in decreasing the required received Eb/N0. We further show that the extremely high-speed real-time packet transmission of greater than 1 Gb/s in a 100-MHz channel bandwidth (i.e., 10 bit/s/Hz) is achieved at the average received Eb/N0 per receiver antenna of approximately 12 dB using 16QAM modulation and turbo coding with the coding rate of 8/9 in 4-by-4 MIMO multiplexing

This paper presents two new methods that use local information alone to predict the resource demands of and determine resource reservation levels for future handoff calls in multimedia wireless IP networks. The proposed methods model the instantaneous resource demand directly. This differs from most existing methods that derive the demands from modeling the factors that impact the demands. As a result, the proposed methods allow new and handoff calls to: (1) follow non-Poisson and/or nonstationary arrival processes; (2) have arbitrary per-call resource demands; and (3) have arbitrarily distributed call and channel holding times. The first method is based on the Wiener prediction theory and the second method is based on time series analysis. Our simulations show that they perform well even for non-Poisson and nonstationary handoff call arrivals, arbitrary per-call bandwidth demands, and nonexponentially distributed call and channel holding times. They generate closely comparable performance with an existing local method and an existing collaborative method that uses information about mobiles in neighboring cells, under assumptions for which these other methods are optimized. The proposed methods are much simpler to implement than most other existing methods with fewer capabilities.

Using the MPEG-4 advanced audio coder (AAC) music as an example of streaming applications, we investigate the improvement of error performance for the streaming service by link adaptation and power control techniques in an enhanced general packet radio services (EGPRS) cellular network. A low packet error rate and variability are essential in providing a short error-burst length so that error concealment techniques can be effectively applied to music packets. We study the effects of a combined link adaptation and power control scheme (referred to as the error-based scheme) for achieving a target error rate and reducing error variability. By simulation, we compare the error performance of the error-based scheme at both the EGPRS block and AAC frame level with another adaptation algorithm (referred to as the throughput-based scheme) with a goal of maximizing overall network throughput. It is found that when offered with a similar traffic load, the former scheme can provide noticeable improvement of music quality over the throughput-based scheme. To achieve a similar AAC frame error rate, our results also show that the error-based scheme can increase the link throughput over the throughput-based scheme by 66.7% in one of our examples. These results reveal that by aiming at required error targets and thus reducing error variability, the error-based scheme for link adaptation and power control are helpful in improving quality and capacity for streaming applications.

A new method is described for routing multimedia traffic in a frequency-hop (FH) store-and-forward packet radio network. The method is illustrated for traffic of two types, each type having its own throughput, delay, and error-rate requirements. A typical application is the routing of voice and data packets in a distributed multiple-hop network. In such an application, voice packets cannot tolerate much delay, but they are allowed to contain a small number of frame erasures while data packets must be delivered error-free even if a moderate delay is required to do so. The fully distributed routing protocol presented in the paper takes into account the type of service required for each type of traffic, and it adapts to the interference as seen by the FH radio receivers in the network. Our approach to multimedia routing is based on least-resistance routing with different link and path resistance metrics for different message types. Each of the resistance metrics for a link reflects the ability of the link to provide the service required by the one of the message types. This includes, but is not limited to, a measure of the likelihood of successful reception by the FH radio receiver for that link. The route selection for a particular type of packet depends on the resistances of the links along the routes from that packet's source to its destination. In general, different routes may be selected for different types of packets. The primary conclusion of this paper is that the quality of service increases for each of the two types of multimedia traffic if the routing protocol accounts for the type of message that is being relayed.

We present an efficient broadcast scheduling algorithm based on mean field annealing (MFA) neural networks. Packet radio (PR) is a technology that applies the packet switching technique to the broadcast radio environment. In a PR network, a single high-speed wideband channel is shared by all PR stations. When a time-division multi-access protocol is used, the access to the channel by the stations' transmissions must be properly scheduled in both the time and space domains in order to avoid collisions or interferences. It is proven that such a scheduling problem is NP-complete. Therefore, an efficient polynomial algorithm rarely exists, and a mean field annealing-based algorithm is proposed to schedule the stations' transmissions in a frame consisting of certain number of time slots. Numerical examples and comparisons with some existing scheduling algorithms have shown that the proposed scheme can find near-optimal solutions with reasonable computational complexity. Both time delay and channel utilization are calculated based on the found schedules.

A frequency-hopped spread-spectrum (FH/SS) packet radio network is considered. Instead of discarding the received packets which were unsuccessfully decoded, the receiver keeps and combines the different received copies of the same packet to improve the throughput and reduce the delay necessary to transmit a packet successfully. It is shown that the throughput is substantially increased over a similar system where the uncorrectable packets are discarded. This system is equivalent to a fast frequency-hopped system where the diversity level is adapted according to the channel conditions. Reed-Solomon (RS) codes are used for error correction and detection.

A set of computer-based modeling tools to help understand the performance and behavior of spread spectrum networks is presented. These tools allow for numerical evaluation of important network performance measures and experimentation with network parameters. A connectivity model for a direct sequence spread spectrum packet radio network which allows the computation of network throughput and can be used to study routing techniques is discussed. This model places direct sequence spread spectrum networks on an equal footing with narrowband networks which have been modeled using the collision channel and transmission range concepts. The mathematical models and algorithms used to evaluate network performance are described, and several examples are presented to demonstrate the impact of various parameters. >

The authors obtain the optimum transmission ranges to maximize throughput for a direct-sequence spread-spectrum multihop packet radio network. In the analysis, they model the network self-interference as a random variable which is equal to the sum of the interference power of all other terminals plus background noise. The model is applicable to other spread-spectrum schemes where the interference of one user appears as a noise source with constant power spectral density to the other users. The network terminals are modeled as a random Poisson field of interference power emitters. The statistics of the interference power at a receiving terminal are obtained and shown to be the stable distributions of a parameter that is dependent on the propagation power loss law. The optimum transmission range in such a network is of the form CK/sup alpha / where C is a constant, K is a function of the processing gain, the background noise power spectral density, and the degree of error-correction coding used, and alpha is related to the power loss law. The results obtained can be used in heuristics to determine optimum routing strategies in multihop networks. >

A link throughput analysis is presented for a slotted frequency-hop multiple-access (FHMA) packet radio network (PRN) operating in the presence of background noise, partial-band noise jamming, and partial-band tone jamming. The PRN consists of an arbitrary number of transceivers arranged in a paired-off topology. Forward error-correction coding is used for packet protection. M-ary FSK modulation is used with hard-decision decoding. Expressions are derived for the link throughput in terms of the channel cutoff rate and capacity. The dependency of the optimal processing gain, code rate, and jamming fraction on the population size, traffic intensity, bit energy to background noise ratio, and bit energy to jammer noise ratio is examined in detail. It is shown that a properly designed (optimized) PRN using random-access FHMA offers a significantly larger heavy-load throughput than a random-access frequency-division multiple-access PRN. >

An examination is made of the performance of type-I hybrid ARQ (automatic repeat request) protocols in a slotted direct-sequence CDMA (code-division multiple access) network operating in a hostile jamming environment. The network consists of an arbitrary number of transceivers arranged in a paired-off topology. The traffic arrival process is derived by means of a Markov model. Throughput-delay expressions are derived in terms of the channel cutoff rate and capacity. The effects of jammer state information are discussed. Network design parameters are identified and their dependency on system parameters is examined in detail. It is shown that, for a given population size, traffic intensity, and bit energy/jammer noise ratio, there is an optimal probability of retransmission, code rate, and processing gain that maximizes network performance in the presence of worst-case pulse jamming. >

The performance analysis of unslotted spread-spectrum packet radio network is very difficult because the received signal-to-noise ratio (SNR) fluctuates during the packet transmission. Past analyses are either based on complete enumeration or restricted to performance bounds. Here, a technique based on ballot theory is developed to analyze the performance of unslotted ALOHA spread-spectrum packet radio networks. This technique is also used to analyze the channel load sensing access protocol. An L-channel model is assumed. >

During the past several years there has been much activity in the development of networking hardware and software for personal computers. At the same time the amateur radio community has been equally active in the development of networking equipment and procedures especially suited for radio circuits. Although the initial efforts have been with terrestrial circuits, experimentation with and development of satellite-based systems is well underway. While we are aware of the many commercial and government organizations active in specialized packet radio systems, our goal is to establish radio networks with inexpensive equipment that is within reach of a large number of individuals with limited budgets. This article will provide a brief history of hardware, software, and protocol development activities as well as a description of amateur packet radio operations. Also discussed are satellite-based operations and development efforts. These include AMSAT-OSCAR-10 and UoSAT-OSCAR-11, currently operating in orbit, and PACSAT, JAS-1, and Phase 3-C, in the design and construction phases.

The problem of analyzing the thoughput of packet radio networks with realistic topologies is considered. We present an algorithm for the solution of this problem and show that both the memory requirements and running time of this algorithm in practice grow polynomially with the size of the problem. Although in theory both can grow exponentially in the worst case, we offer computational experience with the procedure and show that for realistic topologies where connectivity is related to distance, the rate of growth is quadratic in the number of links. Even for regular grids, which are pathological in their symmetry, the rate of growth is only cubic in the number of links. We thus conclude that the procedure is effective for realistic topologies with up to several hundred nodes.

The purpose of this paper is to present an asymptotic performance analysis of a class of multihop radio networks using the calculus of residues. The analysis provides a closed-form approximation of global performance measures, whose precision increases exponentially with the size of the system. The complexity of computing these approximations is independent of the problem size. The method is easily applicable to a variety of topologies and access protocols.

The packet experimental communications system (packet XCS) is a new experimental voice and data switch. It uses a local-area network (LAN) for digital voice transmission, with local intelligence for switching. The packet XCS also has highly distributed control. The individual sites cooperate to provide user services as well as internal data management. We have learned that several local networks, including CSMA/CD networks, can be made to work well for voice transmission and that highly distributed control is practical in such a system. A system has been constructed which is used as a testbed for distributed voice and data communications experiments. This system is purely for experimentation and does not indicate a direction for future Bell System product offerings.

Packet radio networks that employ several parallel multiple-access channels are considered. An architecture for such a network dictates the selection of channels for packet transmission. We propose and analyze two multichannel architectures. In the first, each node employs a single radio and is assigned a channel on which it listens when it does not transmit. To transmit a packet, the node tunes its radio to the channel of the intended receiver for that transmission only. The second architecture requires each radio to use a single channel for both transmission and reception, but provides some of the nodes with more than one radio each, allowing them to serve as bridges between channels. Within these architectures, one can further select the amount of routing information held by each node and the channel-access protocol, both of which greatly affect the network performance. To ascertain the effects of the various parameters, we calculate the throughput in both architectures. The channel-access protocols we consider are slotted ALOHA and CSMA with and without capture. We also evaluate the effect of increasing the amount of routing information held by the nodes.

Fading channels may improve the capacity of slottedALOHA networks. Rayleigh fading and spatially distributed packet transmitters diminish mutual packet interference and thus reduce the average blocking probability of a shared receiver. A transmission model, combining Poisson statistics of the offered data traffic with Rayleigh statistics for the fading channel, is developed. Analytical and numerical results are presented, indicating significant improvements to signaling in cellular mobile networks and in packet radio systems with fading radio channels.

We consider a double-tier communication network architecture for metropolitan area and packet radio networks for controlling the access of network terminals into a single shared broadcast channel. Terminals are organized into local groups in accordance with their geographical proximity or other criteria. Each local group can be associated with a primary station. The latter can serve as an unbuffered repeater for terminal transmissions unto the multiple-access channel. A polling policy is used by the primary stations to gain access into the shared communications backbone. Once a primary station is provided access into the channel, it initiates a local access control procedure. Message delay performance results and bounds are obtained for general reservation local-access procedures. In particular, we analyze and present performance results for a polling/tree-random-access procedure which can be effectively used for many packet radio and cellular digital radio networks, as well as for a polling/positionalpriority scheme for CATV and fiber-optic based networks.

Carrier sense multiple-access (CSMA) protocols are widely used in local area networks (LAN's) to control access to a shared communications channel such as a coaxial cable or radio frequency band. CSMA protocols are designed to exploit the property that the signal propagation time across the LAN is much smaller than the packet transmission time. Consequently, their performance depends on the exact timing of a sequence of asynchronous events originating at different points in the network. Previous models of CSMA protocols have not captured this essential space-time characteristic. Instead, the system has been reduced to a 1-dimensional model (i.e., a sequence of events on a single time line) by assuming either that events are synchronous (i.e., "slotted" operation of the protocol) or that all stations are mutually equidistant (so it can be assumed that each station's time line is identical). The novelty in our work is to describe the system state in terms of an ( N + 1 )-dimensional "ribbon" of space-time, allowing us to faithfully model the exact timings of events for LAN's where the stations are distributed over a fully connected region in N -space. For example, an Ethernet-like "bus" network can be viewed as a 1-dimensional LAN, a terrestrial packet radio network can be viewed as a 2dimensional LAN, and so on. First we highlight some of the key properties exhibited by such a model in the general case, including the notion of an embedded Markovian sequence of "idle points," and the decomposition of the space-time ribbon into cycles (each of which can be further decomposed into a number of "regions"). Then we present extensive results for the interesting special case of a 1-dimensional LAN, i.e., an Ethernet-like "bus" network. Our results show that previous models of bus LAN's can significantly underestimate their performance.

In this paper, it is proposed to employ random multiple signal levels for channel access in packet broadcast networks. We present priority-free random access protocols that possess the advantage of capture effect. The presented schemes are applied to the slotted ALOHA, and the performance is analyzed based on a conservative capture model. Closed-form expressions for the system throughput are derived for a general two-signal level system and a general m -signallevel system. It is shown that the maximum throughput for the twolevel system increases from 0.47 to 0.52 as the separation between the two levels increases. For the m -level system, the maximum throughput increases from 0.52 to 0.66 as m increases from three to infinity. Then a rotary-priority sure-capture random access scheme is presented, which can achieve perfect channel utilization. The time-delay characteristic and the throughput-delay tradeoff are analyzed for the simplest two-level system for which the higher level is double the lower level. The results compare favorably to those of the conventional slotted ALOHA system which employs a single signal level for packet transmission. A number of open problems are addressed.

A review of the satellite-based packet radio activities taking place in the Amateur Radio Service is presented. Five projects are discussed. The Digital Communications Experiment (DCE) on UoSAT-OSCAR-11 is a store-and-forward system that is accessed by a limited number of stations around the world designated as DCE gateways. These gateway stations retrieve and store messages forwarded to them via terrestrial packet radio networks. The Japanese FO-12 satellite is a store-and-forward system, and its user interface is much like computer bulletin boards accessed via land lines. It is available for use by all radio amateurs with the required equipment. Both UO-11 and FO-12 are in low earth orbits. The RUDAK experiment and the PACSAT project are also discussed. >

A continuous-time Markov-chain model for an asynchronous communication spread-spectrum code-division-multiple-access (CDMA) packet radio network is developed. The network is composed of mutually independent users. The receiver-based code is considered; a terminal with a packet to send looks up the destination's code and transmits on that code. Each user senses the channel load and refrains from transmission if the channel load exceeds the channel threshold. The model makes it possible to study the threshold effect of channel load on the performance of the CDMA packet radio network. Improvements in performance of spread-spectrum packet radio networks due to channel-load sensing are shown. Steady-state results for throughput are obtained. >

The integration of digital voice with data in a common packet-switched network system offers a number of potential benefits, including reduced systems cost through sharing of switching and transmission resources, flexible internetworking among systems utilizing different transmission media, and enhanced services for users requiring access to both voice and data communications. Issues which it has been necessary to address in order to realize these benefits include reconstitution of speech from packets arriving at nonuniform intervals, maximization of packet speech multiplexing efficiency, and determination of the implementation requirements for terminals and switching in a large-scale packet voice/data system. A series of packet speech systems experiments to address these issues has been conducted under the sponsorship of the Defense Advanced Research Projects Agency (DARPA). In the initial experiments on the ARPANET, the basic feasibility of speech communication on a store-and-forward packet network was demonstrated. Techniques were developed for reconstitution of speech from packets, and protocols were developed for call setup and for speech transport. Later speech experiments utilizing the Atlantic packet satellite network (SATNET) led to the development of techniques for efficient voice conferencing in a broadcast environment, and for internetting speech between a store-and-forward net (ARPANEI) and a broadcast net (SATNET). Large-scale packet speech multiplexing experiments could not be carried out on ARPANET or SATNET where the network link capacities severely restrict the number of speech users that can be accommodated. However, experiments are currently being carried out using a wide-band satellite-based packet system designed to accommodate a sufficient number of simultaneous users to support realistic experiments in efficient statistical multiplexing. Key developments to date associated with the wide-band experiments have been 1) techniques for internetting via voice/data gateways from a variety of local access networks (packet cable, packet radio, and circuit-switched) to a long-haul broadcast satellite network and 2) compact implementations of packet voice terminals with full protocol and voice capabilities. Basic concepts and issues associated with packet speech systems are described. Requirements and techniques for speech processing, voice protocols, packetization and reconstitution, conferencing, and multiplexing are discussed in the context of a generic packet speech system configuration. Specific experimental configurations and key packet speech results on the ARPANET, SATNET, and wide-band system are reviewed.

The effects of capture on the average system throughput and delay performance of slotted ALOHA were analyzed for slow and fast Rayleigh fading radio channels. A short-range multipoint-to-base station packet radio network is considered. It is shown that larger capture effects and thus improved network performance can be achieved with proper choice of modulation. It is also shown that the use of simple error-correcting codes improves capture. The use of selection diversity also improves the capture effect both for fast and slow fading. It is concluded that the inverse distance variability of the received signal is the main reason for the capture effect. The Rayleigh fading alone yields a very small contribution in terms of throughput; nonetheless, it helps to stabilize the system. Numerical results are presented for a slotted ALOHA system with 50 users. It is found that the maximum average throughput can be increased from about 36% to almost 60% by using channel coding and space diversity. >

A solution for the connection of personal computers or workstations to a central station (represented by a mainframe or a server attached to a local area network) by means of a shared radio channel is presented. Issues regarding the multiple-access protocol are discussed, and an access scheme that is based on the independent stations algorithm (ISA) is presented. The necessary modifications of the basic ISA are discussed, and the overall algorithm is analyzed and tested by simulation. An implementation that is currently being developed is briefly described, focusing on the choices made for the logical link control protocol, the transmission techniques, and the interface software. >

A class of discrete-time priority queueing systems with partial interference is considered. In these systems, N nodes share a common channel to transmit their packets. One node uses a random access scheme, while other nodes access the channel according to preassigned priorities. Packet arrivals are modeled as discrete-time batch processes, and packets are forwarded through the network according to fixed prescribed probabilities. Steady-state analysis of the class of systems under consideration is provided. In particular, we present a recursive method for the derivation of the joint generating function of the queue lengths distribution at the nodes in steady state. The condition for steady state is also derived. A simple example' demonstrates the general analysis and provides some insights into the behavior of systems with partial interference such as multihop packet radio systems.