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Showing papers in "IEEE Journal on Selected Areas in Communications in 1997"


Journal Article•DOI•
TL;DR: This paper describes a self-organizing, multihop, mobile radio network which relies on a code-division access scheme for multimedia support that provides an efficient, stable infrastructure for the integration of different types of traffic in a dynamic radio network.
Abstract: This paper describes a self-organizing, multihop, mobile radio network which relies on a code-division access scheme for multimedia support. In the proposed network architecture, nodes are organized into nonoverlapping clusters. The clusters are independently controlled, and are dynamically reconfigured as the nodes move. This network architecture has three main advantages. First, it provides spatial reuse of the bandwidth due to node clustering. Second, bandwidth can be shared or reserved in a controlled fashion in each cluster. Finally, the cluster algorithm is robust in the face of topological changes caused by node motion, node failure, and node insertion/removal. Simulation shows that this architecture provides an efficient, stable infrastructure for the integration of different types of traffic in a dynamic radio network.

1,695 citations


Journal Article•DOI•
TL;DR: Since lost packets are recovered by local retransmissions as opposed to retransmission from the original sender, end-to-end latency is significantly reduced, and the overall throughput is improved as well.
Abstract: This paper presents the design, implementation, and performance of a reliable multicast transport protocol (RMTP). The RMTP is based on a hierarchical structure in which receivers are grouped into local regions or domains and in each domain there is a special receiver called a designated receiver (DR) which is responsible for sending acknowledgments periodically to the sender, for processing acknowledgment from receivers in its domain, and for retransmitting lost packets to the corresponding receivers. Since lost packets are recovered by local retransmissions as opposed to retransmissions from the original sender, end-to-end latency is significantly reduced, and the overall throughput is improved as well. Also, since only the DRs send their acknowledgments to the sender, instead of all receivers sending their acknowledgments to the sender, a single acknowledgment is generated per local region, and this prevents acknowledgment implosion. Receivers in RMTP send their acknowledgments to the DRs periodically, thereby simplifying error recovery. In addition, lost packets are recovered by selective repeat retransmissions, leading to improved throughput at the cost of minimal additional buffering at the receivers. This paper also describes the implementation of RMTP and its performance on the Internet.

721 citations


Journal Article•DOI•
TL;DR: It is shown that the cell residence time can be described by the generalized gamma distribution and the negative exponential distribution is a good approximation for describing the channel holding time.
Abstract: A mathematical formulation is developed for systematic tracking of the random movement of a mobile station in a cellular environment. It incorporates mobility parameters under the most generalized conditions, so that the model can be tailored to be applicable in most cellular environments. This mobility model is used to characterize different mobility-related traffic parameters in cellular systems. These include the distribution of the cell residence time of both new and handover calls, channel holding time, and the average number of handovers. It is shown that the cell residence time can be described by the generalized gamma distribution. It is also shown that the negative exponential distribution is a good approximation for describing the channel holding time.

652 citations


Journal Article•DOI•
TL;DR: A layered video compression algorithm which, when combined with RLM, provides a comprehensive solution for scalable multicast video transmission in heterogeneous networks.
Abstract: The "Internet Multicast Backbone," or MBone, has risen from a small, research curiosity to a large-scale and widely used communications infrastructure. A driving force behind this growth was the development of multipoint audio, video, and shared whiteboard conferencing applications. Because these real-time media are transmitted at a uniform rate to all of the receivers in the network, a source must either run at the bottleneck rate or overload portions of its multicast distribution tree. We overcome this limitation by moving the burden of rate adaptation from the source to the receivers with a scheme we call receiver-driven layered multicast, or RLM. In RLM, a source distributes a hierarchical signal by striping the different layers across multiple multicast groups, and receivers adjust their reception rate by simply joining and leaving multicast groups. We describe a layered video compression algorithm which, when combined with RLM, provides a comprehensive solution for scalable multicast video transmission in heterogeneous networks. In addition to a layered representation, our coder has low complexity (admitting an efficient software implementation) and high loss resilience (admitting robust operation in loosely controlled environments like the Internet). Even with these constraints, our hybrid DCT/wavelet-based coder exhibits good compression performance. It outperforms all publicly available Internet video codecs while maintaining comparable run-time performance. We have implemented our coder in a "real" application-the UCB/LBL videoconferencing tool vic. Unlike previous work on layered video compression and transmission, we have built a fully operational system that is currently being deployed on a very large scale over the MBone.

576 citations


Journal Article•DOI•
TL;DR: In this article, a survey of protocol functions and mechanisms for data transmission within a group, from multicast routing problems up to end-to-end multipoint transmission control is presented.
Abstract: Group communication supports information transfer between a set of participants. It is becoming more and more relevant in distributed environments. For distributed or replicated data, it provides efficient communication without overloading the network. For some types of multimedia applications, it is the only way to control data transmission to group members. This paper surveys protocol functions and mechanisms for data transmission within a group, from multicast routing problems up to end-to-end multipoint transmission control. We provide a bibliography which is organized by topic.

358 citations


Journal Article•DOI•
TL;DR: In this paper, the authors study the problem of constructing multicast trees to meet the quality of service requirements of real-time interactive applications operating in high-speed packet-switched environments and present a heuristic that demonstrates good average case behavior in terms of the maximum interdestination delay variation.
Abstract: We study the problem or constructing multicast trees to meet the quality of service requirements of real-time interactive applications operating in high-speed packet-switched environments. In particular, we assume that multicast communication depends on: (1) bounded delay along the paths from the source to each destination and (2) bounded variation among the delays along these paths. We first establish that the problem of determining such a constrained tree is NP-complete. We then present a heuristic that demonstrates good average case behavior in terms of the maximum interdestination delay variation. The heuristic achieves its best performance under conditions typical of multicast scenarios in high speed networks. We also show that it is possible to dynamically reorganize the initial tree in response to changes in the destination set, in a way that is minimally disruptive to the multicast session.

343 citations


Journal Article•DOI•
TL;DR: It is shown that a receiver-initiated error control protocol which requires receivers to transmit NAKs point-to-point to the sender provides higher throughput than a sender- initiated counterpart for both classes of applications.
Abstract: Sender-initiated reliable multicast protocols based on the use of positive acknowledgments (ACKs) can suffer performance degradation as the number of receivers increases. This degradation is due to the fact that the sender must bear much of the complexity associated with reliable data transfer (e.g., maintaining state information and timers for each of the receivers and responding to receivers' ACKs). A potential solution to this problem is to shift the burden of providing reliable data transfer to the receivers-thus resulting in receiver-initiated multicast error control protocols based on the use of negative acknowledgments (NAKs). We determine the maximum throughputs for generic sender-initiated and receiver-initiated protocols for two classes of applications: (1) one-many applications where one participant sends data to a set of receivers and (2) many-many applications where all participants simultaneously send and receive data to/from each other. We show that a receiver-initiated error control protocol which requires receivers to transmit NAKs point-to-point to the sender provides higher throughput than a sender-initiated counterpart for both classes of applications. We further demonstrate that, in the case of a one many application, replacing point-to-point transfer of NAKs with multicasting of NAKs coupled with a random backoff procedure provides a substantial additional increase in the throughput of a receiver-initiated error control protocol over a sender-initiated protocol. We also find, however, that such a modification leads to a throughput degradation in the case of many-many applications.

324 citations


Journal Article•DOI•
TL;DR: Simulation results over random networks show that unconstrained algorithms are not capable of fulfilling the QoS requirements of real-time applications in wide-area networks, and semiconstrained and constrained heuristics are capable of successfully constructing MC trees which satisfy the QS requirements ofreal-time traffic.
Abstract: Multicast (MC) routing algorithms capable of satisfying the quality of service (QoS) requirements of real-time applications will be essential for future high-speed networks. We compare the performance of all of the important MC routing algorithms when applied to networks with asymmetric link loads. Each algorithm is judged based on the quality of the MC trees it generates and its efficiency in managing the network resources. Simulation results over random networks show that unconstrained algorithms are not capable of fulfilling the QoS requirements of real-time applications in wide-area networks. Simulations also reveal that one of the unconstrained algorithms, reverse path multicasting (RPM), is quite inefficient when applied to asymmetric networks. We study how combining routing with resource reservation and admission control improves the RPM's efficiency in managing the network resources. The performance of one semiconstrained heuristic, MSC, three constrained Steiner tree (CST) heuristics, Kompella, Pasquale, and Polyzos (1992), constrained adaptive ordering (CAO), and bounded shortest multicast algorithm (BSMA), and one constrained shortest path tree (CSPT) heuristic, the constrained Dijkstra heuristic (CDKS) are also studied. Simulations show that the semiconstrained and constrained heuristics are capable of successfully constructing MC trees which satisfy the QoS requirements of real-time traffic. However, the cost performance of the heuristics varies. The BSMA's MC trees are lower in cost than all other constrained heuristics. Finally, we compare the execution times of all algorithms, unconstrained, semiconstrained, and constrained.

315 citations


Journal Article•DOI•
TL;DR: A comprehensive history and survey of IS methods is presented, and a guide to the strengths and weaknesses of the techniques is offered, to indicate which techniques are suitable for various types of communications systems.
Abstract: Importance sampling (IS) is a simulation technique which aims to reduce the variance (or other cost function) of a given simulation estimator. In communication systems, this usually, but not always, means attempting to reduce the variance of the bit error rate (BER) estimator. By reducing the variance, IS estimators can achieve a given precision from shorter simulation runs; hence the term "quick simulation." The idea behind IS is that certain values of the input random variables in a simulation have more impact on the parameter being estimated than others. If these "important" values are emphasized by sampling more frequently, then the estimator variance can be reduced. Hence, the basic methodology in IS is to choose a distribution which encourages the important values. This use of a "biased" distribution will, of course, result in a biased estimator if applied directly in the simulation. However, there is a simple procedure whereby the simulation outputs are weighted to correct for the use of the biased distribution, and this ensures that the new IS estimator is unbiased. Hence, the "art" of designing quick simulations via IS is entirely dependent on the choice of biased distribution. Over the last 50 years, IS techniques have flourished, but it is only in the last decade that coherent design methods have emerged. The outcome of these developments is that at the expense of increasing technical content, modern techniques can offer substantial run-time saving for a very broad range of problems. We present a comprehensive history and survey of IS methods. In addition, we offer a guide to the strengths and weaknesses of the techniques, and hence indicate which techniques are suitable for various types of communications systems. We stress that simple approaches can still yield useful savings, and so the simulation practitioner as well as the technical researcher should consider IS as a possible simulation tool.

284 citations


Journal Article•DOI•
TL;DR: Comparisons with outdoor experimental data collected in Manhattan and Boston show that the computer-based propagation tool can predict signal strengths in these environments with very good accuracy, showing that simulations, rather than costly field measurements, can lead to accurate determination of the coverage area for a given system design.
Abstract: Engineers designing and installing outdoor and indoor wireless communications systems need effective and practical tools to help them determine base station antenna locations for adequate signal coverage. Computer-based radio propagation prediction tools are now often used in designing these systems. We assess the performance of such a propagation tool based on ray-tracing and advanced computational methods. We have compared its predictions with outdoor experimental data collected in Manhattan and Boston (at 900 MHz and 2 GHz). The comparisons show that the computer-based propagation tool can predict signal strengths in these environments with very good accuracy. The prediction errors are within 6 dB in both mean and standard deviation. This shows that simulations, rather than costly field measurements, can lead to accurate determination of the coverage area for a given system design.

274 citations


Journal Article•DOI•
TL;DR: An ATM-based implementation of DT-DVTR in LEO satellite ISL networks is presented with some emphasis on the optimization alternatives, and the performance in terms of delay jitter is evaluated for an example ISL topology.
Abstract: Satellite systems are going to build a part of the future personal communications infrastructure. The first-generation candidates for satellite personal communication networks (S-PCN) will rely on low Earth orbit (LEO) and medium Earth orbit (MEO) constellations. A noticeable trend in this field is toward broadband services and the use of ATM. For LEO satellite systems employing intersatellite links (ISLs), this paper proposes an overall networking concept that introduces the strengths of ATM to their operation. The core of the paper is the design of a new routing scheme for the periodically time-variant ISL subnetwork, discrete-time dynamic virtual topology routing (DT-DVTR), and its ATM implementation. DT-DVTR works completely off line, i.e., prior to the operational phase of the system. In a first step, a virtual topology is set up for all successive time intervals of the system period, providing instantaneous sets of alternative paths between all source-destination node pairs. In the second step, path sequences over a series of time interval are chosen from that according to certain optimization procedures. An ATM-based implementation of DT-DVTR in LEO satellite ISL networks is presented with some emphasis on the optimization alternatives, and the performance in terms of delay jitter is evaluated for an example ISL topology.

Journal Article•DOI•
TL;DR: Early experiments with the WATMnet prototype have been conducted to validate major protocol and software aspects, including DLC, wireless control, and mobility signaling for handoff, Selected network-based multimedia/video applications requiring moderate bit-rates have been successfully demonstrated on the laptop PC.
Abstract: A prototype microcellular wireless asynchronous transfer mode network (WATMnet) capable of providing integrated multimedia communication services to mobile terminals is described in this paper. The experimental system's hardware consists of laptop computers (NEC Versa-M) with WATMnet interface cards, multiple VME/i960 processor-based WATMnet base stations, and a mobility-enhanced local-area ATM switch. The prototype wireless network interface cards operate at peak bit-rates up to 8 Mb/s, using low-power 2.4 GHz industrial, scientific, and medical (ISM)-band modems. Wireless network protocols at the portable terminal and base station interfaces support available bit rate (ABR), variable bit rate (VBR), and constant bit rate (CBR) transport services compatible with ATM using a dynamic time-division multiple-access/time-division duplex (TDMA/TDD) MAC protocol for channel sharing and data link control (DLC) protocol for error recovery. A custom wireless control protocol is also implemented between the portable and base units for support of radio link related functions such as user registration and handoff. All network entities including the portable, base and switch use a mobility-enhanced version of ATM ("Q.2931+") signaling for switched virtual circuit (SVC) connection control functions, including handoff. In the first stage of the prototype, the application-level API is TCP/UP over ATM ABR service class using AAL5. Early experiments with the WATMnet prototype have been conducted to validate major protocol and software aspects, including DLC, wireless control, and mobility signaling for handoff, Selected network-based multimedia/video applications requiring moderate bit-rates (/spl sim/0.5-1 Mb/s) in the ABR mode have been successfully demonstrated on the laptop PC.

Journal Article•DOI•
TL;DR: A novel dynamic guard channel scheme is proposed which adapts the number of guard channels in each cell according to the current estimate of the handoff call arrival rate derived from the current number of ongoing calls in neighboring cells and the mobility pattern, so as to keep the handoffs call blocking probability close to the targeted objective.
Abstract: In future personal communications networks (PCNs) supporting network-wide handoffs, new and handoff requests will compete for connection resources in both the mobile and backbone networks. Forced call terminations due to handoff call blocking are generally more objectionable than new call blocking. The previously proposed guard channel scheme for radio channel allocation in cellular networks reduces handoff call blocking probability substantially at the expense of slight increases in new call blocking probability by giving resource access priority to handoff calls over new calls in call admission control. While the effectiveness of a fixed number of guard channels has been demonstrated under stationary traffic conditions, with nonstationary call arrival rates in a practical system, the achieved handoff call blocking probability may deviate significantly from the desired objective. We propose a novel dynamic guard channel scheme which adapts the number of guard channels in each cell according to the current estimate of the handoff call arrival rate derived from the current number of ongoing calls in neighboring cells and the mobility pattern, so as to keep the handoff call blocking probability close to the targeted objective while constraining the new call blocking probability to be below a given level. The proposed scheme is applicable to channel allocation over cellular mobile networks, and is extended to bandwidth allocation over the backbone network to enable a unified approach to prioritized call admission control over the ATM-based PCN.

Journal Article•DOI•
TL;DR: An asynchronous transfer mode (ATM)-based concept for the routing of information in a low Earth orbit/medium Earth orbit (LEO/MEO) satellite system including intersatellite links (ISLs) is proposed, with specific emphasis on the design of an ATM-based routing scheme for the ISL part of the system.
Abstract: An asynchronous transfer mode (ATM)-based concept for the routing of information in a low Earth orbit/medium Earth orbit (LEO/MEO) satellite system including intersatellite links (ISLs) is proposed. Specific emphasis is laid on the design of an ATM-based routing scheme for the ISL part of the system. The approach is to prepare a virtual topology by means of virtual path connections (VPCs) connecting all pairs of end nodes in the ISL subnetwork for a complete period in advance, similar to implementing a set of (time dependent) routing tables. The search for available end-to-end routes within the ISL network is based on a modified Dijkstra (1959) shortest path algorithm (M-DSPA) capable of coping with the time-variant topology. With respect to the deterministic time variance of the considered ISL topologies, an analysis of optimization aspects for the selection of a path at call setup time is presented. The performance of the path search in combination with a specific optimization procedure is-by means of extensive simulations-evaluated for example LEO and MEO ISL topologies, respectively.

Journal Article•DOI•
TL;DR: It is found that there is a tradeoff among concentration of residue, strictness of fairness, and implementational simplicity (for the design of high-speed switches) in the general multicast switching problem.
Abstract: We design a scheduler for an M/spl times/N input-queued multicast switch. It is assumed that: 1) each input maintains a single queue for arriving multicast cells and 2) only the cell at the head of line (HOL) can be observed and scheduled at one time. The scheduler needs to be: 1) work-conserving (no output port may be idle as long as there is an input cell destined to it) and 2) fair (which means that no input cell may be held at HOL for more than a fixed number of cell times). The aim is to find a work-conserving, fair policy that delivers maximum throughput and minimizes input queue latency, and yet is simple to implement. When a scheduling policy decides which cells to schedule, contention may require that it leave a residue of cells to be scheduled in the next cell time. The selection of where to place the residue uniquely defines the scheduling policy. Subject to a fairness constraint, we argue that a policy which always concentrates the residue on as few inputs as possible generally outperforms all other policies. We find that there is a tradeoff among concentration of residue (for high throughput), strictness of fairness (to prevent starvation), and implementational simplicity (for the design of high-speed switches). By mapping the general multicast switching problem onto a variation of the popular block-packing game Tetris, we are able to analyze various scheduling policies which possess these attributes in different proportions. We present a novel scheduling policy, called TATRA, which performs extremely well and is strict in fairness. We also present a simple weight-based algorithm, called WBA.

Journal Article•DOI•
TL;DR: A new capacity design method based on theoretical expressions for GoS and QoS as functions of traffic intensity and CAC thresholds is described, and computer simulation results are presented that strongly support the proposed design method.
Abstract: Since code-division multiple-access (CDMA) capacity is interference limited, call admission control (CAC) must guarantee both a grade of service (GoS), i.e., the blocking rate, and a quality of service (QoS), i.e., the loss probability of communication quality. This paper describes the development of a new capacity design method based on these two concepts. Theoretical expressions for GoS and QoS as functions of traffic intensity and CAC thresholds are first derived from the traffic theory viewpoint, and then a design method using these expressions is presented. At that time, two strategies for CAC are assumed. One is based on the number of users, and the other is based on the interference level. Computer simulation results are presented that strongly support the proposed design method. Furthermore, numerical examples and a performance comparison of the two strategies considering various propagation parameters, nonuniform traffic distributions, and various transmission rates are shown.

Journal Article•DOI•
TL;DR: The results indicate that variable-rate transmission can increase the quality of the decoded sequences without increases in the end-to-end delay, and it is shown that for the leaky-bucket channel, the channel constraints can be combined with the buffer constraints, such that the system is identical to CBR transmission with an additional, infrequently imposed constraint.
Abstract: Variable bit-rate (VBR) transmission of video over ATM networks has long been said to provide substantial benefits, both in terms of network utilization and video quality, when compared with conventional constant bit-rate (CBR) approaches. However, realistic VBR transmission environments will certainly impose constraints on the rate that each source can submit to the network. We formalize the problem of optimizing the quality of the transmitted video by jointly selecting the source rate (number of bits used for a given frame) and the channel rate (number of bits transmitted during a given frame interval). This selection is subject to two sets of constraints, namely, (1) the end-to-end delay has to be constant to allow for real-time video display and (2) the transmission rate has to be consistent with the traffic parameters negotiated by user and network. For a general class of constraints, including such popular ones as the leaky bucket, we introduce an algorithm to find the optimal solution to this problem. This algorithm allows us to compare VBR and CBR under the same end-to-end delay constraints. Our results indicate that variable-rate transmission can increase the quality of the decoded sequences without increases in the end-to-end delay. Finally, we show that for the leaky-bucket channel, the channel constraints can be combined with the buffer constraints, such that the system is identical to CBR transmission with an additional, infrequently imposed constraint. Therefore, video quality with a leaky-bucket channel can achieve the same quality of a CBR channel with larger physical buffers, without adding to the physical delay in the system.

Journal Article•DOI•
TL;DR: It is proved that MIP is loop-free at every instant, and that it is deadlock-free and obtains multicast routing trees within a finite time after the occurrence of an arbitrary sequence of topology or unicast changes.
Abstract: In network multimedia applications such as multiparty teleconferencing, users often need to send the same information to several (but not necessarily all) other users. To manage such one-to-many or many-to-many communication efficiently in wide-area internetworks, it is imperative to support and perform multicast routing. Multicast routing sends a single copy of a message from a source to multiple receivers over a communication link that is shared by the paths to the receivers. Loop-freedom is an especially important consideration in multicasting because applications using multicasting tend to be multimedia and bandwidth intensive, and loops in multicast routing duplicate looping packets. We present and verify a new multicast routing protocol, called multicast Internet protocol (MIP), which offers a simple and flexible approach to constructing both group-shared and shortest-paths multicast trees. MIP can be sender-initiated or receiver-initiated or both; therefore, it can be tailored to the particular nature of an application's group dynamics and size. MIP is independent of the underlying unicast routing algorithms used. MIP is robust and adapts under dynamic network conditions (topology or link cost changes) to maintain loop-free multicast routing. Under stable network conditions, MIP has no maintenance or control message overhead. We prove that MIP is loop-free at every instant, and that it is deadlock-free and obtains multicast routing trees within a finite time after the occurrence of an arbitrary sequence of topology or unicast changes.

Journal Article•DOI•
TL;DR: Two mobile location management algorithms for ATM (asynchronous transfer mode) networks based on the PNNI (private network-to-network interface) standard are presented and it is observed that the two schemes show a contrasting behavior in terms of the value to be used for the parameter S to achieve the least average total cost.
Abstract: This paper presents two mobile location management algorithms for ATM (asynchronous transfer mode) networks based on the PNNI (private network-to-network interface) standard. The first solution is called the mobile PNNI scheme because it builds on the PNNI routing protocol. It uses limited-scope (characterized by a parameter S) reachability updates, forwarding pointers, and a route optimization procedure. The second solution is called the LR (location registers) scheme because it introduces location registers (such as the cellular home and visitor location registers) into the PNNI standards-based hierarchical networks. This scheme uses a hierarchical arrangement of location registers with the hierarchy limited to a certain level S. Analytical models are set up to compare the average move, search, and total costs per move of these two schemes for different values of the CMR (call-to-mobility ratio), and to provide guidelines for selecting parameters of the algorithms. Results show that at low CMRs (CMR<0.025), the LR scheme performs better than the mobile PNNI scheme. We also observe that the two schemes show a contrasting behavior in terms of the value to be used for the parameter S to achieve the least average total cost. At low CMRs, the parameter S should be high for the mobile PNNI scheme, but low for the LR scheme, and vice versa for high CMRs.

Journal Article•DOI•
TL;DR: The influence of the repeated attempt effect on the quality of service experienced by the mobile customers is discussed by means of numerical results.
Abstract: In the planning of modern cellular mobile communication systems, the impact of customer behavior has to be carefully taken into account. Two models dealing with the call retrial phenomenon are presented. The first model considers a base station with a finite customer population and repeated attempts. A Markov chain modeling is proposed, and an efficient recursive solution of the state probabilities is presented. The second model focuses on the use of the guard channel concept to prioritize the handover traffic. Again, the retrial phenomenon plays an important role. The influence of the repeated attempt effect on the quality of service experienced by the mobile customers is discussed by means of numerical results.

Journal Article•DOI•
TL;DR: An efficient broadcast scheduling algorithm based on mean field annealing (MFA) neural networks to schedule the stations' transmissions in a frame consisting of certain number of time slots is presented.
Abstract: 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

Journal Article•DOI•
G.P. Pollini1, Chih-Lin I2•
TL;DR: Results indicate that over a wide range of parameters, it may be possible to reduce both the radio bandwidth and fixed network signaling load for a modest increase in call setup delay.
Abstract: Future microcellular personal communications systems (PCSs) will be characterized by high user density and high mobility. It is expected that registrations will incur a large amount of the radio link signaling traffic. A profile-based strategy (PBS) is proposed to reduce the signaling traffic on the radio link by increasing the intelligence within the fixed network. The system maintains a sequential list of the most likely places where each user is located. The list is ranked from the most to the least likely place where a user is found. When a call arrives for a mobile, it is paged sequentially in each location within the list. When a user moves between location areas in this list, no location update is required. The list may be provided by the user or may be based on each user's past calling history. The method for doing this is outside the scope of this work. This work focuses on the potential performance improvements that can result from maintaining such a list. This paper compares the performance of the proposed strategy to the typical geographic-based location-tracking schemes being implemented in evolving digital cellular and cordless standards. Key performance measures for the comparison are radio bandwidth, fixed network SS7 traffic, and call setup delay. We investigate the conditions under which the PBS performs better than the traditional scheme. Results indicate that over a wide range of parameters, it may be possible to reduce both the radio bandwidth and fixed network signaling load for a modest increase in call setup delay.

Journal Article•DOI•
TL;DR: This work presents a uniform call admission control scheme based on a Chernoff bound method that provides an easy and flexible mechanism for supporting multiple VBR service classes with different QoS requirements, and evaluates the efficacy of the call admission Control scheme over a set of MPEG-1 coded video tracts.
Abstract: Variable bit-rate (VBR) compressed video is known to exhibit significant, multiple-time-scale rate variability. A number of researchers have considered transmitting stored video from server to a client using smoothing algorithms to reduce this rate variability. These algorithms exploit client buffering capabilities and determine a "smooth" rate transmission schedule, while ensuring that a client buffer neither overflows nor underflows. We investigate how video smoothing impacts the statistical multiplexing gains available with such traffic, and we show that a significant amount of statistical multiplexing gains can still be achieved. We then examine the implication of these results on network resource management and call admission control when transmitting smoothed stored video using VBR service with statistical quality-of-service (QoS) guarantees. Specifically, we present a uniform call admission control scheme based on a Chernoff bound method that uses a simple, novel traffic model requiring only a few parameters. This scheme provides an easy and flexible mechanism for supporting multiple VBR service classes with different QoS requirements. We evaluate the efficacy of the call admission control scheme over a set of MPEG-1 coded video tracts.

Journal Article•DOI•
TL;DR: Predictive rate-distortion (RD) optimized motion estimation techniques are studied and developed for very low bit-rate video coding and indicate that they yield very good computation-performance tradeoffs.
Abstract: Predictive rate-distortion (RD) optimized motion estimation techniques are studied and developed for very low bit-rate video coding. Four types of predictors are studied: mean, weighted mean, median, and statistical mean. The weighted mean is obtained using conventional linear prediction techniques. The statistical mean is obtained using a finite-state machine modeling method based on dynamic vector quantization. By employing prediction, the motion vector search can then be constrained to a small area. The effective search area is reduced further by varying its size based on the local statistics of the motion field, through using a Lagrangian as the search matching measure and imposing probabilistic models during the search process. The proposed motion estimation techniques are analyzed within a simple DCT-based video coding framework, where an RD criterion is used for alternating among three coding modes for each 8/spl times/8 block: motion only, motion-compensated prediction and DCT, and intra-DCT. Experimental results indicate that our techniques yield very good computation-performance tradeoffs. When such techniques are applied to an RD optimized H.263 framework at very low bit rates, the resulting H.263 compliant video coder is shown to outperform the H.263 TMN5 coder in terms of compression performance and computations simultaneously.

Journal Article•DOI•
TL;DR: An analytical model is developed to evaluate the performance of hierarchical cellular networks with subscribers of varying mobility, and quantify the gain obtained by providing overflow to alternate resources as well as the advantages in resource reassignment according to the speed classification.
Abstract: Hierarchical cellular networks with subscribers of varying mobility are considered. Microcells are used to address the high-intensity traffic of mainly slow mobility areas, and macrocells are overlaid over the microcells to cater mainly to high-mobility lower density traffic. The two tiers of microcells and macrocells provide a secondary resource for new traffic as well as handoffs for mobile subscribers of different mobility classes. Furthermore, resources in alternate layers are monitored to assign the appropriate resource types when they become available. We develop an analytical model to evaluate the performance of such systems, and quantify the gain obtained by providing overflow to alternate resources as well as the advantages in resource reassignment according to the speed classification.

Journal Article•DOI•
TL;DR: A new nonlinear integer programming representation of the static channel assignment (SCA) problem is formulated and a new self-organizing neural network is proposed which is able to solve the SCA problem and many other practical optimization problems due to its generalizing ability.
Abstract: We examine the problem of assigning calls in a cellular mobile network to channels in the frequency domain. Such assignments must be made so that interference between calls is minimized, while demands for channels are satisfied. A new nonlinear integer programming representation of the static channel assignment (SCA) problem is formulated. We then propose two different neural networks for solving this problem. The first is an improved Hopfield (1982) neural network which resolves the issues of infeasibility and poor solution quality which have plagued the reputation of the Hopfield network. The second approach is a new self-organizing neural network which is able to solve the SCA problem and many other practical optimization problems due to its generalizing ability. A variety of test problems are used to compare the performance of the neural techniques against more traditional heuristic approaches. Finally, extensions to the dynamic channel assignment problem are considered.

Journal Article•DOI•
TL;DR: In this paper, the problem of wireless access to asynchronous transfer modes (ATMs) was studied, and the authors proposed a polling scheme with non-preemptive priority for all the CBR and VBR sources.
Abstract: We study the problem of wireless access to asynchronous transfer modes (ATMs). We consider three classes of ATM sources: constant bit rate (CBR), variable bit rate (VBR), and available bit rate (ABR). We propose a polling scheme with nonpreemptive priority. Under such a scheme, we derive sufficient conditions such that all the CBR sources satisfy their jitter constraints and all the VBR sources satisfy their delay constraints. The remaining bandwidth is used by the ABR sources, for which we adapt a random access scheme proposed by Chen and Lee (1994). For this random access scheme, we derive the throughput-offer load characteristic, and thus the capacity. Based on this, we propose adaptive random access schemes that track the offer load to its optimal value. Our simulations show that our adaptive schemes maintain a high throughput with respect to the whole range of system load.

Journal Article•DOI•
Hong-Yi Tzeng1, Kai-Yeung Sin•
TL;DR: This work establishes a unified framework to derive a multicast congestion control protocol for an ABR service from a given rate-based unicast protocol, and generalizes a known necessary and sufficient condition on the max-min fairness of unicast rate allocation for a multicasts service.
Abstract: In the ATM Forum activities, considerable efforts have focused on the congestion control of point-to-point available bit rate (ABR) service. We present a novel approach that extends existing point-to-point (unicast) congestion control protocols to a point-to-multipoint (multicast) environment. In particular, we establish a unified framework to derive a multicast congestion control protocol for an ABR service from a given rate-based unicast protocol. We generalize a known necessary and sufficient condition on the max-min fairness of unicast rate allocation for a multicast service. Using this condition, we show that the resulting multicast protocol derived using our framework preserves the fairness characteristics of the underlying unicast protocol. The practical significance of our approach is illustrated by extending a standard congestion control mechanism for an ABR service to a multicast environment. The performance of the resulting multicast protocol is examined using benchmark network configurations suggested by the traffic management subworking group at the ATM Forum, and simulation results are presented to substantiate our claims.

Journal Article•DOI•
TL;DR: It is shown that when call-holding times are Erlang distributed, easy-to-compute formulas for the probability of a call completion and the expected effective call- holding times for both a complete call and an incomplete call can be derived.
Abstract: It is well known that, due to the mobility of a portable and limited channel availability, calls of portables may not be completed due to being blocked or terminated during the call initiation or the handover process. The characteristics of the call-completion and call-holding times for both a complete call and an incomplete call are of critical importance for establishing the actual billing process in the PCS network. We derive the call-completion probability (hence, call-dropping probability) and the effective call-holding time distributions for complete/incomplete calls with a general cell-residence time and a general call-holding time are analyzed, and general computable formulas are obtained. We show that when call-holding times are Erlang distributed, easy-to-compute formulas for the probability of a call completion and the expected effective call-holding times for both a complete call and an incomplete call can be derived.

Journal Article•DOI•
TL;DR: A new scheme to partition and track mobile users and its implementation based on cellular architecture are proposed and the cost analysis model is used to demonstrate the feasibility of updating and searching costs to reduce the amount of resources when using wireless channels.
Abstract: The partition of location areas is designed to minimize the total costs of finding users' location and tracking their movement in personal communication networks (PCNs). A new scheme to partition and track mobile users and its implementation based on cellular architecture are proposed. According to the tracking strategy, the mobile station (MS) transmits only update messages at specific reporting cells, while the search for a mobile user is done at the vicinity of the cell to which the user has just reported. We use the cost analysis model to demonstrate the feasibility of updating and searching costs to reduce the amount of resources when using wireless channels. Simulations are performed to compare the performance of three schemes: always update, always search, and our new scheme.