This paper presents an analysis of loss of decentralized environmental notification messages (DENMs) in a vehicle-to-vehicle scenario in which DENMs are transferred over subchannels of the sidelink specifically allocated to DENM transmission. A simple but meaningful Markov chain model of DENM generation that captures a major requirement of DENM generation and simultaneously recognizes the potentially bursty nature and correlation in the DENM generation process is presented. DENM loss is determined according to a capture model, and closed-form formulae are presented for state dependent probability mass function of the number of DENMs lost as a function of the system parameters. To the best of our knowledge, mass functions of loss of DENMs have not appeared in the literature to date. It is shown that the stochastic equilibrium loss distribution at a given offered load and number of vehicles is independent of the burstiness of the DENM generation process. Numerical examples show how the correlation coefficient of the number of transmitting vehicles is affected by the expected length of the burst period and the number of vehicles generating DENMs at a fixed level of offered load. It is concluded that stochastic equilibrium loss information is inadequate for traffic engineering for DENM transmission.

In this paper, we present a detailed analysis of Blue-tooth Low Energy-based scanning systems wherein the objective is to successfully scan all items of a group of a prescribed size within a prescribed scanning period at a prescribed probability of success, for example, less than one failure in one million. We show that a design based upon independence among collision events fails to achieve the target reliability objectives by roughly an order of magnitude. Our analysis, which is verified through extensive simulation, shows that correlation among the collision events has a major impact upon the scanning time required to successfully scan all the members of a group.

We quantify and compare the complexities of two computational algorithms, the Floyd-Warshall algorithm and a widest-remaining- path-first algorithm (WRaPFt), which we introduce here, that compute abstract node representations for autonomous systems in hierarchical networks. Both algorithms compute the traversing characteristics under bandwidth dependent link costs and give identical results. The techniques discussed here can be applied to any cost metrics that can be defined as elements of a semi-ring. We compare the execution time and the number of steps required to compute the efficient frontiers for the case of flow-controlled cost functions. For all node pairs, we found WRaPFt to be faster at lower connectivities and Floyd- Warshall to be faster at higher connectivities, the crossover point being roughly 0.5. Execution time for the WRaPFt algorithm is never more than 1.7 times the execution time of the Floyd-Warshall algorithm. Thus, WRaPFt can provide a speed-up factor of 1.7/ V in routing table update computation. Finally, we found that the execution time increases approximately proportional to the 4th power of V for both algorithms for fully connected random test networks.

Space communications feature large round-trip time (RTT) delays and highly variable data error rates. In this paper, we propose a new content transfer protocol based on RaptorQ codes and turbo codes, incorporated with a real-time channel prediction model, to maximize file transfer from space vehicles to Earth stations. While turbo codes are used to correct channel errors, RaptorQ codes are applied to eliminate the need for negative-acknowledgment (NAK) of the loss of any specific packet. To reduce the effect of channel variation, a practical signal-to-noise ratio (SNR) prediction model that is used to periodically adjust the turbo encoder in distant source spacecraft is developed. This new protocol, termed as the dynamic code selection method (DCSM), is compared with two other methods: a turbo codes based genie method (upper bound of DCSM performance) assuming that the channel condition is perfectly known in advance and a static method in which a fixed turbo encoder is used throughout each communication pass. Simulation results demonstrate that the genie and the new DCSM methods can increase telemetry channel throughput by about 47% and 46%, respectively, compared to the static method. In addition, throughput performance using the DCSM is about 99% of the upper bound achieved with the genie method.

This research vision and challenges paper focuses on microprocessor design and activity-recognition data processing for medical devices in student-athlete health care. Sports are the second leading cause of mild traumatic brain injury (mTBI) for people aged between 15 and 24 years. Significant work has been done in the use of sensors to determine the linear and rotational acceleration of head impacts, and their potential correlation to concussions. However, recent studies have shown that subconcussive trauma may be a better indicator of future risks, such as impaired cognitive, motor, and sensory functions, as well as loss of control of emotion and behavior. Therefore, the current reactive and hospital-centered approach where waiting until a concussion has been diagnosed before providing quality care to student athletes is insufficient. Our project will enable a transformation in student-athlete healthcare through the design of sensors towards a preventive, proactive, and evidence-based approach for assessing and mitigating the risk of long-term brain injury. Research and develop improvements towards a next-generation form-factor xPatch device with radio transceivers for secure and efficient processing of head impact data, while considering transmission range vs. power consumption, receiver sensitivity, data rate, start-up time, latency, capacity, and cost. We will enhance X2 Biosystems's currently existing Injury Management System data platform through development and integration of on-chip data processing towards real-time recording and secure wireless transmission of impact data. Integration of 6 degree-of-freedom support vector data will enable improved machine learning-based principal component analysis of head impact data. In order to provide real-time integration of on-chip machine learning and cloud-based impact classification algorithms, we will also develop an encrypted 915MHz ISM band radio solution for secure and reliable transmission of impact data for real-time interpretation by athletic trainers or certified medical professionals on the sidelines within athletic environments.

In this paper, we investigate the quality of sensor localization that can be achieved by analyzing the strengths of signals transmitted by the sensors and measured at anchors placed at the corners of a rectangular area within which the sensors are deployed. The propagation environment is slow Rayleigh fading for which the parameters are unknown. Three localizations approaches are investigated-direct computation, least squares, log-likelihood-all taking advantage of the multipath resolution capability of the receiver. The numerical results demonstrate that localization accuracy improves as the number of resolved multipath components increases. The results also show that the log-likelihood approach provides the best localization accuracy by a very small margin over the least squares approach, but that the least squares approach has significantly smaller run times. The direct computation approach produces less accurate localization than the other two approaches, but produces accurate trends in behavior at very low computational costs.

Presented herein is a new protocol, RaptorQP2P, for reliable peer-to-peer sharing of large files-currently on the order of hundreds of MB but expected to grow to the terabyte range-over networks. The new protocol features two levels of RaptorQ encoding. At the top layer, the entire file is RaptorQ encoded to yield a collection of source blocks and repair blocks. At the lower layer, each source/repair block is RaptorQ encoded independently to yield a collection of source symbols and repair symbols for the block. The symbols are independently transferred among the peers and when a sufficient number of distinct symbols for a block have been received, whether source or repair, the block can be reconstructed. Similarly, the file can be reconstructed using a sufficient number of arbitrary distinct blocks. With real world traces measured on the BitTorrent system, we have conducted extensive simulations to evaluate and compare the performance of RaptorQP2P and BitTorrent. The results indicate that the new protocol handles network dynamics and peer churn smoothly, has excellent scalability with respect to both file size and user population, and performs well. As an example, for a 512 MB file distributed to 1000 peers, download completion time using the new protocol was found to be approximately 45.5% of the completion time required using BitTorrent.

We investigate the quality of sensor localization achievable in a slow Rayleigh fading environment using three different localizations approaches.

We introduce a queue-adaptive scheduling mechanism that facilitates scheduling differentiation among nodes having traffic within the same access category in IEEE 802.11-based ad hoc networks, which to the best of our knowledge, has not been previously addressed. We investigate the effectiveness of the proposed mechanism, which is easy to implement and requires minimal overhead, via simulation. More specifically, we investigate the effectiveness of dynamically controlling the values of the interframe space, contention window, and the transmission opportunity in the context of voice over IP in a distributed IEEE 802.11 ad hoc network. Our simulation results show that the proposed mechanism considerably lowers the percentage of dropped packets as the number of active calls increases beyond the network's call carrying capacity, dramatically improves system stability at higher traffic loads, and results in a significant improvement in call carrying capacity.

The objective of this paper is to determine the conditions under which two ordinary users in a communication network having information bound for the same receiver can benefit by relaying portions of each others data. Each user is allocated the same bandwidth, and the objective of each user is to maximize information transfer per joule. The users are free to either use their entire allocated bandwidth to transfer their own messages or to allocate portions of their bandwidth to cooperatively transfer messages with another user. In the non-cooperative mode, the users maximize return by adjusting power whereas in the cooperative mode, the users vary both their transmission powers and the proportion of hared bandwidth to achieve maximal utility. Solution techniques are developed for the case of cooperating users in symmetric user-position and channel condition setups. Within this setup, there are two options, one where each user gets the same return (symmetric), and one in which the two users may achieve different returns (asymmetric). We find that gains are possible with both options and that the asymmetric option yields larger returns but requires scheduling in order to achieve equal throughput.

We designed, implemented, and evaluated a compact signal processing system on a Berkeley mote for vehicle classification via acoustic characteristics. The main components of this system include a sound sampler, FFT converter, an energy-based adaptive CFAR detector, and a classifier. We chose principal component analysis (PCA) as the primary classifier because of its high accuracy, good scalability, and low resource requirements. This system enables a Berkeley mote to detect a passing vehicle and to decide its type. Only the result of the identification analysis is reported to sink nodes. With this proof-of-concept implementation, we explored limits of system design on a mote. Using a testbed based on this system, we compared classification quality, energy consumption, and latency of the PCA classifier with two typical classifiers in both centralized and distributed signal processing scenarios. Briefly, we found that the PCA yields better classification quality but has slightly lower energy efficiency than the others. By extensive simulations, we studied the dependency of PCA on signal quality and efficiency issues of signal processing in WSNs

MPEG encoded video accounts for a considerable share of high-volume traffic in both wireline and wireless networks of the next generation. MPEG video requires high bandwidth, and its QoS is highly sensitive to packet loss, delay and bit error rate. Such unique features raise great challenges for traffic engineering. However, the scarcity of video traces bottlenecks the advances in video traffic study. In this paper, we propose a VBR video traffic model that accurately characterizes VBR MPEG4 video traces and mimics their statistical properties. Two highlights of our model include a novel scene-based Markov modulated process and the feedback control on the frame size distribution. We validate our model by comparing various statistics of the simulated traces with those of the original traces. It is shown that our model provides an effective solution to obtain ample high-quality traces for video traffic analysis and network performance evaluation.

Traffic engineering and admission control schemes for QoS support over the Internet share the hypothesis that offered traffic must be policed. Recently, measurement-based techniques have been proposed as effective QoS-guarantee mechanisms. For them, a key element is a thorough understanding of the characteristics of regulated traffic. In this paper, we present an analysis of the traffic output process for a leaky-bucket-regulated voice source with silence suppression, which would be a prime candidate for VoIP service. Specifically, we present a procedure for calculating the mean and the variance of the amount of regulated traffic as a function of the measurement period. We evaluate the quality of our results via simulation.

In this paper, we quantify the resources-bandwidth and credits-required to guarantee an upper bounded of a prescribed violation probability violates the probability of a prescribed class-based target packet delay. We assume that the upper bound on the traffic arrival process for each class can be described statistically via effective envelopes. Our results are derived using a network calculus-based model of the system. We consider both deterministic and probabilistic delay guarantees. We find that the resources required to guarantee a probabilistic delay upper bound are less than those required to guarantee a deterministic delay upper bound of the same value.

The buffered crossbar packet switch employs the crossbar-based switching fabric architecture with internal buffers and the combined input and output queueing scheme. It is promising to support quality of service (QoS) in buffered crossbar switch without complex implementation. In this paper, we study the problem of supporting traffic classes whose QoS requirements are stated statistically in buffered crossbar switches. Specifically, we investigate the resource allocation issue. We extend the network calculus-based model and techniques developed in Q. Duan and J. N. Daigle (Oct. 2002) to determine the amount of bandwidth and number of credits that must be allocated in a buffered crossbar switch to guarantee each traffic class a packet delay that is upper bounded with a small violation probability. We find that the required amount of resources for probabilistic delay upper bound guarantee is less than what is required for deterministic delay upper bound guarantee. This implies that bandwidth and credits can be saved in buffered crossbar switch if resource allocation is based on probabilistic delay objective.

We examine the traffic-handling capabilities of general packet radio service (GPRS) with respect to supporting IP-based Internet services. We begin with an overview of GPRS in which we discuss its general characteristics, connection set-up procedures, and data transfer procedures, both for the reverse link and the forward link. We then present an analytical model to assess throughput of the reverse link as a function of the number of users connected and the distribution of user message lengths. Next, we investigate the capability of GPRS to support World Wide Web access using the analytical model. Specifically, we present a realistic scenario for user sessions operating under the Hypertext Transfer Protocol (HTTP) and we assess the transaction-handling capabilities as a function of the number of user sessions, taking into account network delays, forward link transmission, random access delay, and other factors. The results obtained by using the analytical model in both cases are verified by simulation. We also consider, via simulation, a scenario where both continuously backlogged users and users operating HTTP sessions are present. We find that, in the case of continuously backlogged users, an increase in the number of contention slots does not always translate to an increase in throughput, while, in the case of users operating HTTP sessions, the downlink serves as the main bottleneck in the system. We conclude with a discussion of some open issues in the design of GPRS-based Internet access.

We develop a model to quantify the performance of message transmission systems in which users must reserve transmission resources via a contention mechanism prior to transmission. Our work is motivated by a desire to understand the performance characteristics of systems such as the general packet radio service (GPRS), where the single forward link of the wireless access system is organized as a sequence of frames, each of which has first a contention period and then a service period. There are a fixed number of fixed-length contention slots in each contention period. Each contending customer chooses at random the slot in which to contend, and success is determined by a capture model. A contender who fails waits for the next contention period, then again chooses at random the slot in which to contend; this process is repeated until the contender is successful. Customers who have contended successfully are served during the service period, which has a prescribed number of fixed-length slots, on a first-come-first-served (FCFS) basis, with the required number of service units being drawn independently from a general discrete distribution having finite support. We model the system as a Markov renewal process embedded at service departure times. We solve the model and then compute the equilibrium distributions of the number of customers in the system at arbitrary points in time and at customer arrival times. Finally, we give a numerical example in which we demonstrate the usefulness of our results in understanding the behavior of GPRS.

We consider the problem of allocation of resources within a buffered crossbar switch with credit-based flow control to support quality of service for traffic flows in the Internet. Specifically, we assume there is a collection of classes of service, each characterized by a strict delay upper bound that applies across all packets of the class. Flows entering the switch are characterized by arrival curves, and the switch must achieve delay objectives subject to the arrival curves. We develop a technique for determining the amount of bandwidth and the number of credits that must be allocated to a flow in order to guarantee its delay requirements. This technique also provides the basis for analyzing the maximum traffic handling capacity of a switch. We provide numerical examples that illustrate the application of our techniques. Finally, we discuss the performance of buffered crossbar switches having finite buffers with respect to that of output buffered switches under the same class of scheduling regimes.

In a hierarchical network, groups of nodes are represented by logical nodes for the purposes of simplifying routing. Each group has a set of ingress-egress nodes, and routing information is conveyed to the outside world in the form of a transition matrix that gives the cost of traversing the network between each ingress-egress node pair. In this paper, we present a minimal logical node representation that has sufficient descriptive power to enable path selection in support of connection admission control for services that have both path (restrictive) and link (additive) constraints. For example, the representation can be used to find a path that maximizes bandwidth subject to a delay constraint or minimizes delay subject to a bandwidth constraint. We present our solution in the form of a matrix whose elements are vectors, each of which specifies the efficient frontier of the solution space, and we specify an efficient procedure for constructing the efficient frontier. We present the least upper bound on the number of elements that must be present in the efficient frontier. We provide numerical examples that illustrate construction of the efficient frontier.

In this paper, we consider a single-server service system with contention periods preceding the service operation. Customer interarrival times are exponentially distributed with a rate that depends on the current system occupancy, and customers must be successful in the contention phase in order to obtain access to the service phase. There are C contention slots in each contention period, which are chosen at random by the contending customers, and success is determined by a capture model. Each contention period is followed by S service periods. Customers who have already contended successfully receive service on a FCFS basis, and service times are drawn independently from a discrete distribution. We model the system as a Markov renewal process embedded at departure service times. We solve the model and then compute the equilibrium distributions of the number of customers in the system at an arbitrary point in time and at customer arrival epochs. Finally, we give numerical examples that show the effects of contention on system performance.

The bottleneck in many telecommunication systems has often been modeled by an M/G/1 queueing system with priorities. While the probability generating function (PGF) for the occupancy distribution of each traffic class can be readily obtained, the occupancy distributions have been obtainable only rarely. However, the occupancy distribution is of great importance, particularly in those cases where the moments are not all finite. We present a method of obtaining the occupancy distribution from the PGF and demonstrate its validity by obtaining the occupancy distributions for a number of cases, including those with regularly varying service time distributions.

In this paper, we discuss the traffic-handling capabilities of the general packet radio service (GPRS) with respect to supporting IP-based Internet services. We begin with an overview of GPRS. We then present an analytical model through which to assess throughput of the reverse link for a scenario in which users are continuously backlogged. Next, we present a realistic scenario for user sessions operating under the hypertext transfer protocol (HTTP), and we assess the transaction-handling capabilities, again using the analytical model. We conclude with a discussion of some open issues in the design of GPRS-based Internet access.

This paper reports the findings of a simulation study of the queueing behavior of "best-effort" traffic in the presence of constant bit-rate and variable bit-rate isochronous traffic. In this study, best-effort traffic refers to ATM cells that support communications between host end systems executing various applications and exchanging information using TCP/IP. The performance measures considered are TCP cell loss, TCP packet loss, mean cell queueing delay, and mean cell queue length. Our simulation results show that, under certain conditions, best-effort TCP traffic may experience as much as 2% cell loss. Our results also show that the probability of cell and packet loss decreases logarithmically with increased buffer size.>

The paper reports the findings of a simulation studying the queueing behavior of "best-effort" traffic in the presence of constant-bit-rate (CBR) and variable bit-rate (VBR) isochronous traffic. In the study, best-effort traffic refers to ATM cells that support communication between host end systems executing various applications and exchanging information using TCP/IP. The performance measures considered are TCP cell loss, TCP packet loss, mean cell queueing delay, and mean cell queue length. The simulation results show that, under certain conditions, best-effort TCP traffic may experience as much as 2% cell loss. The results also show that the probability of cell and packet loss decreases logarithmically with increased buffer size.>

The queueing behavior of many communication systems is well modeled by a queueing system in which time is slotted, and the number of entities that arrive during a slot is dependent upon the state of a discrete time, discrete state Markov chain. Techniques for analyzing such systems have appeared in the literature from time to time, but distributions have been presented in only rare instances In the paper, the authors present the probability generating function (PGF) for joint and marginal buffer occupancy distributions of statistical time division multiplexing systems in this class. They discuss inversion of the PGF using discrete Fourier transforms, and also discuss a simple technique for obtaining moments of the queue length distribution. Numerical results, including queue length distributions for some special cases, are presented.>

The authors consider a service system in which n servers render service to two classes of prioritized traffic, which arrive to the system according to independent Poisson processes. Newly arriving call requests are granted server access on a first-come-first-served (FCFS) basis as long as there are fewer than g servers occupied. If fewer than g servers are free at the arrival time of a lower priority customer, the higher priority call requests are granted immediate service unless all servers are busy, in which case the call is dropped. This model has been used to study the handoff problem in a cellular telephone system, but the analytic approach taken was very complicated. The authors present a simple, novel, alternative approach to solving for the equilibrium probabilities for the number of lower priority calls in the queue and other quantities of interest. They describe two additional alternative approaches based on Neuts' matrix analytic approach for checking results, and point out that while the results of all these approaches agree with each other, they differ from previously published results. Because these three approaches are essentially independent, it is conjectured that there are problems with the earlier numerical results. Further work has revealed that this is the case.>

The queuing behavior of many communication systems is well modeled by a queuing system in which time is slotted and in which the number of entities arriving during a slot is dependent upon the state of a discrete-time, discrete-state Markov chain. The probability generating function is presented for joint and marginal buffer occupancy distributions of statistical time-division multiplexing systems in this class. A simple technique is discussed for obtaining moments of the queue length distribution. In addition, a discussion is presented of inversion of the probability generating function. Numerical results, including queue length distributions for some special cases, are presented.>

The system is modeled as a single-server queueing system having a finite number of each of two types of worker: document user and document entry operator. Worker think times are assumed to be exponentially distributed while job service times are drawn from general distributions; worker think times and job service times are dependent upon worker type. Throughput and waiting times as a function of the number of users of each type are analyzed under a nonpreemptive rule in which document use has priority over document entry.>

A packet-switching node is modeled as a single-server queueing system. Messages in class c have J/sub c/ packets, each of which has a length which may be drawn from a general service time distribution prescribed on the basis of the packet's position in the message. All of the messages' packets arrive at the instant of message arrival; each packet within the message has a preassigned priority. The first packet of the message joins a transmission queue at the message arrival instant, there being one queue for each distinct priority. Upon completion of transmission of a given packet within a message, its successor packet joins the queue of appropriate priority. Packets which are waiting in the queues are serviced by the trunk under a nonpreemptive priority discipline. A method is presented for finding average message system times for the service discipline just described. This class of service disciplines includes several service disciplines which have been reported in the literature including the PR+RR, the HOL, and HOLP, the RR, and others that are discussed herein. Some numerical examples are presented.>

The authors consider worker waiting times and throughput for an optical-disk-based document storage and retrieval system in an office environment. They model the system as a single-server queuing system having a finite number of each of two types of worker: document user and document entry. Worker think times are assumed to be exponentially distributed while job service times are drawn from general distributions; worker think times and job service times are dependent on worker type. Throughput and waiting time as a function of the number of users of each type are analyzed under a nonpreemptive rule in which document use has priority over document entry. The methodology developed to assess average delays in finite-source queuing systems with priority appears to be novel.>

Three disciplines for prioritized transmission of messages in packet switching systems are considered. Messages from a finite number of classes having their lengths drawn independently from general service time distributions are assumed to arrive to the system according to independent Poisson processes. After entering the switch, the messages are divided into packets, overhead is added, and then the packets join a queue to be served according to one of the following disciplines: head-of-the-line (HOL), HOL with message preemption, and prioritized round robin (RR). Preemption of packet transmission is not allowed in any of the disciplines, and the service disciplines of different classes need not be the same. One result of the work presented here is a model to assess delays in the case of the prioritized RR; this work appears to be the first in which messages are not of fixed length and all quanta from a given message are served at the same priority. Numerical results which illustrate effects of choice of packet lengths and service disciplines upon delay of messages from the different classes are presented.

In a packet voice communication system, packets are fed to a common queue by a number of independent voice sources and are removed from this queue on a first-come-first-serve basis for transmission over a communication link of finite capacity. Each voice source alternates between active periods, during which packets are generated at regular intervals, and inactive periods, during which no packets are generated. In this paper, we discuss three models, a semi-Markov process model, a continuous-time Markov chain model, and a uniform arrival and service model, to assess the queueing behavior of such systems. Numerical results obtained from each of the three models are compared to each other, to results obtained from a discrete event simulation program, and to results obtained from anM/D/1analysis. Parameters of the model are the average duration of active and inactive periods, the packet generation rate, the communication link capacity, and the total number of voice sources. Conclusions are drawn regarding which models appear to be most appropriate in the parameter ranges investigated.

Jobs associated with messages arriving to a communication processing system (CPS) can be described as ordered sets of prioritized tasks, where each task represents the execution of some part of the system's software. The service system of the CPS consists of a collection of queues, one for each distinct priority, serviced by a single server under a preemptive resume discipline. The routing of the job through this collection of queues is defined by a job's priority vector which specifies the priority at which each task in the job is to execute. For example, if a two-task job's priority vector is (3, 2), then the job first joins the priority 3 queue, obtains service for the first task, then joins the priority 2 queue. In a previous paper, the author and a colleague described a method through which average delays in such task-oriented processing systems could be computed for the special case in which all jobs have identical structure. This paper presents a substantially simplified version of the analysis methodology presented previously and extends that work to the case of a finite number of job classes, each job class having a finite number of tasks.

This paper presents closed form expressions for average delay in a class of work conserving single server queueing systems suggested by, but not limited to, the behavior of heavily I/O oriented computing systems. Jobs arrive at the system according to a Poisson process, and each job consists of a set of tasks which must be accomplished in a prescribed order. The tasks with indexihave a general service time distribution Giand a priorityPiwhich.is arbitrary but prescribed. Tasks are serviced within composite priority groups in a preemptive resume fashion and in the order in which, the tasks reached a ready-for-service status. Such queue behavior would be characteristic, for example, of a front end processor or a terminal controller in a communications environment. A design example is presented which illustrates the usefulness of the model to compare alternatives in communications systems design and other areas as well.