We address the problem of optimizing the delivery of multimedia services with different quality-of-service (QoS) requirements to mobile users. We assume that the network provides two distinct classes of service to which users may subscribe: Premium, or Economy. Subscribers to the Premium service pay more for their connections but receive a higher level of quality measured by a set of parameters such as call blocking probability, coding rate, and format of the multimedia services. By optimizing the delivery of the multimedia services, we mean that the network guarantees that all users receive their agreed-upon contractual level of quality, while maximizing the links' throughput, avoiding congestion, and maintaining the QoS requirements for each type of media (e.g., video, voice and data). Our proposed solution is based upon utilizing genetic algorithms to solve a multiobjective optimization function that adaptively selects the downloading bit-rate for each type of traffic flow subject to the constraints of the optimization function. A traffic flow is an abstract of aggregate traffic of the same type of media (e.g., voice, video, or data) that is downloaded to a group of users who share some common attribute such as the same class of service. The optimization function is adaptive in the sense that the selected downloading bit-rates are time-dependant according to the dynamics of the links' traffic-loads and users' requests. It is implemented on every output port of each node in the network. The function is used to control a filter that changes the coding rate of each media-type and, if necessary, performs transcoding of one, or more, media-types (i.e., video, voice, or data). Simulation results show significant improvement in terms of increasing the number of admitted users, while maintaining the QoS requirements, as well as target call blocking rates. An interesting result to report is that the performance improvement of the system (measured by the gain in the number of admitted users at a certain utilization factor) is not simply bounded by the maximum available link throughput. It is, rather, limited by the additional revenue gained by admitting more users. The increase in the revenue saturates at a certain offered traffic-load. Hence, it is not worth it, from a service provider perspective, to admit additional users above this traffic load despite the fact that the filtering algorithm results indicate otherwise.

We propose a novel wireless radio resource manager to control both the transmission power and the bit rate of mobile devices cooperatively, whereas previous work has focused on handling them separately. The proposed scheme is called Genetic Algorithm for Mobiles Equilibrium (GAME). Based on an evolutionary computational model, GAME assigns optimally both the transmitting power and bit rate values to every mobile device in a given cell. Optimal allocation is in the sense that every mobile unit gets only enough resources necessary for meeting or exceeding its quality of service (QoS) requirements. GAME solves an optimization function that strives to maintain the QoS requirements of the different multimedia streams subject to the physical channel characteristics. Having done that, we gain further benefits as well. In addition to maintaining the QoS requirements, GAME tends to prefer assigning lower levels of transmission power, thus extending the mobile units' battery life, minimizing interference seen by other users, and significantly decreasing the call blocking (or dropping) rates. Furthermore, GAME radio resource allocations reduce infrastructure costs by requiring fewer base stations per square kilometer. In our experiments, we have seen a 70% average expansion in the base station coverage area with 40% decrease in mobile call outage probability among other benefits.

In this paper, we present an adaptive filtering strategy for improving the delivery of multimedia services with guaranteed quality of service (QoS) on an end-to-end basis. Multimedia content is downloaded along an integrated wired/wireless network from media servers to mobile end-users. The proposed strategy assures that the network resources (e.g. capacities, buffer) are efficiently utilized and the QoS requirements (e.g. delay bandwidth) of two different classes of end-users are always respected as traffic flows in downstream. This objective is achieved by filtering, in each node of the network, the amount as well as type of content downloaded such that possible congestion episodes are avoided and the downloading speed of content is maintained. According to the proposed strategy, filters are able to select the bit-rate values required to download traffic by adaptively scaling the coding rate and, if necessary, transcoding media streams. The action performed by each filter is controlled by an optimization function that is implemented in our proposed resource manager. Simulation results demonstrate that the filtering strategy has significantly decreased the call-blocking rate while maintaining users QoS requirements and maximizing network capabilities.

We consider the combined control of mobile stations' transmitter power and bit rate in a CDMA microcellular environment. The objective is maximizing the number of QoS-satisfied mobile users in conjunction with minimizing network infrastructure cost. We first review the challenges associated with microcells and user mobility in a Manhattan-like street grid. Next, we discuss the additional aid offered by the combined scheme over the power-only method in utilizing the scarce radio resources efficiently. We demonstrate through link-level and system-level simulations the significant enhancements over the established power-only control scheme. Both coverage and spectrum efficiencies are improved for voice/data services having diverse QoS requirements.

Abstract Wireless research activities span a wide range of disciplines: communications engineering, software engineering and computer sciences covering all aspects of communication and services including client platforms and server deployment, network protocols and infrastructure, service development and deployment and so on. Our research group at the IBM T. J. Watson Research Center has initiated a number of research activities covering several aspects of the wireless research. In this paper, we highlight our work on the Bluetooth™ wireless technology and the BlueDrekar™ software stack, the WebSplitter middleware for content distribution, and we will share our vision for interconnecting trillion devices. Copyright © 2002 John Wiley & Sons, Ltd.

This paper introduces PARO, a power-aware routing optimization that helps to minimize the transmission power needed to forward packets between wireless devices in ad hoc networks. Using PARO, one or more intermediate nodes called "redirectors" elects to forward packets on behalf of source-destination pairs thus reducing the aggregate transmission power consumed by wireless devices. PARO is applicable to a number of networking environments including sensor networks, home networks and mobile ad hoc networks. In this paper, we present the detailed design of PARO and evaluate the protocol using simulation and experimentation. We show through simulation that PARO is capable of outperforming traditional broadcast-based routing protocols (e.g., MANET routing protocols) due to its power conserving point-to-point on-demand design. We discuss some initial experiences from an early implementation of the protocol in an experimental wireless testbed using off-the-shelf radio technology.

We propose an algorithm for controlling mobile users transmitter power and information bit rate cooperatively in CDMA networks. The active component of this scheme is called genetic algorithm for mobiles equilibrium (GAME). Based an an evolutionary computational model, the base station tries to achieve an adequate equilibrium between its users. Thereof each mobile can send its traffic with a suitable power to support it over the different path losses and interference. In the mean time, its battery life is being preserved while limiting the interference seen by neighbors. This optimization is based on each user traffic, required signal quality, the channel characteristics and users spatial distribution. A significant enhancement in signal quality and power level has been noticed through several experiments.

We consider the combined control of mobile stations' transmitter power and bit rate in CDMA microcells. The objective is to maximize the number of QoS-satisfied mobile users. We discuss the additional aid offered by the combined scheme over the power-only method in utilizing the scarce radio resources efficiently. Next, we review the problems associated with the microcellular environment and users' mobility in a Manhattan-like street grid. We demonstrate through system level simulations the significant enhancements over the established power-only control scheme. Both spectrum efficiency and system capacity are improved for multimedia services having diverse QoS requirements.

WebSplitter symbolizes the union of pervasive multi-device computing and collaborative multi-user computing. WebSplitter provides a unified XML framework that enables multi-device and multi-user Web browsing. WebSplitter splits a requested Web page and delivers the appropriate partial view of each page to each user, or more accurately to each user's set of devices. Multiple users can participate in the same browsing session, as in traditional conferencing groupware. Depending on the access privileges of the user to the different components of content on each page, WebSplitter generates a personalized partial view. WebSplitter further splits the partial view among the devices available to each user, e.g. laptop, wireless PDA, projection display, stereo speakers, orchestrating a composite presentation across the devices. A wireless PDA can browse while remotely controlling the multimedia capabilities of nearby devices. The architecture consists of an XML metadata policy file defining access privileges to XML tags on a Web page, a middleware proxy that splits XML Web content to create partial views, and a client-side component, e.g. applet, enabling user login and reception of pushed browsing data. Service discovery finds and registers proxies, browsing sessions, and device capabilities. We demonstrate the feasibility of splitting the different tags in an XML Web page to different end users browsers, and of pushing updates from the browsing session to heterogeneous devices, including a laptop and a PDA.

There is a growing interest in deploying multimedia services in mobile cellular networks. Call admission control (CAC) is a key factor in quality of service (QoS) provisioning for these services. We propose a CAC which maximizes the revenue by determining the call admission decisions for each state. A semi-Markov decision process is employed to model the cellular system. Also, the QoS requirement such as the handoff dropping probability is taken into consideration. Simulations reveal that our CAC outperforms the upper limit CAC while satisfying the QoS requirements.

Due to scarce radio resources (e.g., wireless link bandwidth), the cell size is likely to become smaller to increase the frequency reuse ratio, especially in next generation mobile cellular networks. In this scenario, the analysis of handoff traffic is crucial for adequate QoS provisioning. With the assumption that incoming handoff traffic into a cell is equivalent to outgoing handoff traffic from the cell, we propose an iterative computation technique to analyze handoff traffic approximately. The iterative relaxation method is employed to calculate the steady state probabilities. The call blocking probability and the handoff dropping probability are accordingly obtained. We conduct the approximate analysis of handoff traffic in the case of trunk reservation call admission policy and compare the results with those of simulation experiments.

In wireless ATM-based networks, admission control is required to reserve resources in advance for calls requiring guaranteed services. In the case of a multimedia call, each of its substreams (i.e., video, audio, and data) has its own distinct quality of service (QoS) requirements (e.g., cell loss rate, delay, jitter, etc.). The network attempts to deliver the required QoS by allocating an appropriate amount of resources (e.g., bandwidth, buffers). The negotiated QoS requirements constitute a certain QoS level that remains fixed during the call (static allocation approach). Accordingly, the corresponding allocated resources also remain unchanged. We present and analyze an adaptive allocation of resources algorithm based on genetic algorithms. In contrast to the static approach, each substream declares a preset range of acceptable QoS levels (e.g., high, medium, low) instead of just a single one. As the availability of resources in the wireless network varies, the algorithm selects the best possible QoS level that each substream can obtain. In case of congestion, the algorithm attempts to free up some resources by degrading the QoS levels of the existing calls to lesser ones. This is done, however, under the constraint of achieving maximum utilization of the resources while simultaneously distributing them fairly among the calls. The degradation is limited to a minimum value predefined in a user-defined profile (UDP). Genetic algorithms have been used to solve the optimization problem. From the user perspective, the perception of the QoS degradation is very graceful and happens only during overload periods. The network services, on the other hand, are greatly enhanced due to the fact that the call blocking probability is significantly decreased. Simulation results demonstrate that the proposed algorithm performs well in terms of increasing the number of admitted calls while utilizing the available bandwidth fairly and effectively.

Palmtop computers are rapidly becoming the popular platform of choice for running personal productivity applications, but their networking capabilities are still lagging behind user expectations.Due to memory size, cost, and power considerations most palmtop computers support only limited form of point-to-point communication, namely, connection via a PSTN modem or a serial RS-232 cable.Due to wired point-to-point nature of these interfaces, satisfactory solutions for multi-point communication and direct LAN connection do not yet exist..The BlueSky project aims at providing a low-cost, lowpower, indoor RF wireless networking solution for handheld devices.Our solution consists of two components; a BlueSky wireless attachment that plugs into the standard serial port of any palmtop device, and a LAN access point that acts as a layer 213 bridge between the wireless and the wired part of the network.Palmtop devices use dialup networking software and PPP to connect via BlueSky to the network.BlueSky enables MobileIP style seamless roaming between access points without requiring any changes to the communication stack of plamtop devices.Our roaming enhancements to PPP are implemented in the PPP server, the access point, and the BlueSky attachment, all of which are completely transparent to the palmtop device.In this paper, we present our design rationale and implementation experience of building the BlueSky system.We show in what ways low cost, low power, and form factor constraints affect the choice of protocols and function placement in wireless networking systems.We use PPP based mobility solution as an example to illustrate what design adjustments and compromises one has to make to build a working solution.Based on our experience, we recommend making a slight modification to the dialup networking stack of palmtop computers.Our proposed modification offers an efficient, lower cost, and lighter-weight wireless networking solution which is particularly attractive for enabling mobility over emerging short range, low power wireless technologies, such as IrDA, HomeRF and Bluetooth.Permission Lo make digital or hard copies of all or part of this work for personal or classroom use is granted without fee provided that copies are not made or distributed for profit or commercial advantage and that copies bear this notice and the full citation on the tirst page.To copy otherWe, to republish, to post on servers or to redistribute to lists.rcquircs prior specific permission and/or a fee.

Recently there is a growing interest in adaptive multimedia networking where the bandwidth of an ongoing multimedia call is variable. In this paper, we propose a measurement-based call admission control (CAC) which guarantees the upper bound of the call degradation probability defined herein. Our CAC measures the state of the cellular system in a distributed manner and reflects the observed history of the system on making call admission decisions. Also, a bandwidth adaptation algorithm to minimize the call degradation probability is presented. Simulations reveal that our CAC algorithm outperforms the upper limit CAC algorithm while maintaining quality of service.

In this paper we propose a generic call admission control scheme based on genetic algorithms. The scheme is applied to a wireless network using an adaptive resource allocation framework. A multimedia connection is represented in terms of three substreams (video, audio and data) each with a pre-specified range of acceptable quality of service (QoS) requirements. This range of requirements make the application adaptive in the sense that each of its substreams specify to the network a number of acceptable QoS levels instead of just a single one. For example, MPEG-based video can be transmitted at different levels of varying qualities. As resource availability in the wireless network changes, the control scheme selects the best possible QoS level that each substream can obtain for a certain period of time. In case of congestion, the control scheme attempts to free up some wireless resources by degrading the QoS of existing calls to a lesser level in order to accommodate new calls. This is done, however, under the constraint of achieving maximum utilization of the resources; and sharing them fairly among the calls. The degradation in the QoS is limited to a minimum value for each multimedia substream defined in a user-defined profile. From the user perspective, the perception of the QoS degradation is very graceful and happens only during over-load periods. The network services, on the other hand, are greatly enhanced due to the fact that the call blocking probability is significantly decreased. Genetic algorithms have been used to solve the optimization problem. Simulation results for a wireless ATM network demonstrate that the proposed scheme performs well in terms of increasing the number of calls admitted while effectively utilizing the available bandwidth fairly among the calls.

The wireless medium is a s=ce shared resource in mobde computing.Consequently, the media access control (MAC) layer influences the fairness and robustness of the wireless network.According to the current MAC protocok, stations are not able to gain access equdy to the shared wireless medium.This problem is conunody known as the fairness problem.The fairness problem occurs mostly bemuse of the existence of hidden stations and the presumption of a non-Wy connected wird=s network topology.This paper addresses solutions to the fairness problem in wireless networks.persistent carrier sense mdtiple access based dg~ rithms are proposed in which a fair wirel=s access for each user is accomplished using a precsdtiated N acce~Probabfity, Pij, . .that represents the Eti access probabtity horn station i to j. Lii access probabfities are dcdated at the source station in two ways using connection-based and time-based media access methods.According to the used methods, each active user broadcasts information on either the number of Iogicd connections or the average cent ent ion time to the stations within the communication reach.This information exchange provides partird understanding of the topology of the network to the stations.Each station reserves a specific priority for itse~to gain access to the shared medium.It is suggested that the information is exchanged during the W access discovery procedure for the connection-based method, and periodidy for the time-based method.Link access probabtit ies are modified every time the exchanged information is received.The proposed algorithms are dynamic and sensitive to the changes in the network topology.The sdgoritb have been implemented in a specific media access control protocol [1], but they are app~mble to d media access control protocok.Simdation restits show that the algorithms restit in an order of magnitude performance improvement in terms of throughput in a wirel= network.

In this paper, we present the performance of go-back-N and selective-reject automatic repeat request modes of high level data link control (HDLC) and the effect of the window size, packet error rate and processor speed in the maximum utilization in infrared wireless transmission systems. The maximum utilization (efficiency) performance of these protocols in half-duplex and full-duplex environments are being driven under different channel conditions. We demonstrate which link parameter is dominant in the performance under different link conditions, and also optimal packet sizes for ARQ schemes. It is also concluded that processing time affects the performance more than channel capacity after accomplishing a certain bit error rate.

In this paper, we evaluate the performance of infrared communication methods using repetition-rate coding over L-PPM wireless links to provide reliable infrared transmission throughout indoor environments. All receivers adapt the communication data rate by monitoring the channel, and they instruct transmitters to use highest rate if it is possible. Reducing the user data rate by a factor of RR (rate reduction) results in an RR-fold SNR improvement. We analyze how repetition-rate coding improves the capture probability of the infrared modem over L-PPM links. We also discuss the effect of RR-fold transmission in utilization using go-back-N and selective reject ARQ schemes.

The shadow cluster concept can be used to estimate future resource requirements and to perform call admission decisions in wireless networks. Shadow clusters can be used to decide if a new call can be admitted to a wireless network based on its quality-of-service (QoS) requirements and local traffic conditions. The shadow cluster concept can especially be useful in future wireless networks with microcellular architectures where service will be provided to users with diverse QoS requirements. The framework of a shadow cluster system is completely distributed, and can be viewed as a message system where mobile terminals inform the base stations in their neighborhood about their requirements, position, and movement parameters. With this information, base stations predict future demands, reserve resources accordingly, and admit only those mobile terminals which can be supported adequately. The shadow cluster concept involves some processing and communication overheads. These overheads have no effect on wireless resources, but only on the base stations and the underlying wireline network. It is shown how base stations determine the probabilities that a mobile terminal will be active in other cells at future times, define and maintain shadow clusters by using probabilistic information on the future position of their mobile terminals with active calls, and predict resource demands based on shadow cluster information. In addition, a call admission algorithm is introduced, which uses current traffic and bandwidth utilization conditions, as well as the amount of resources and maximum allowable "dropping probability" being requested. Performance results showing the advantages of the shadow cluster concept are also included.

The major focus of this paper is distributed call admission control in mobile/wireless networks, the purpose of which is to limit the call handoff dropping probability in loss systems or the cell overload probability in lossless systems. Handoff dropping or cell overload are consequences of congestion in wireless networks. Our call admission control algorithm takes into consideration the number of calls in adjacent cells, in addition to the number of calls in the cell where a new call request is made, in order to make a call admission decision. This is done by every base station in a distributed manner without the involvement of the network call processor. The admission condition is simple enough that the admission decision can be made in real time. Furthermore, we show that our distributed call admission control scheme limits the handoff dropping or the cell overload probability to a predefined level almost independent of load conditions. This is an important requirement of future wireless/mobile networks with quality-of-service (QoS) provisioning.

This paper describes the Shadow Cluster concept, a novel idea that can be used to improve the resource allocation and the call admission procedures in wire less networks.Shadow clusters can be used to allocate resources that need to be reserved for call hand-offs, and to determine if a new call should be admitted to a wireless network based on the call's requirements and local traffic conditions.The shadow cluster concept is targeted for ATM-based wireless networks with a micro/n.ano-cellulararchitecture, where service will be provided to users with very diverse requirements.In these n.etworks, and as a consequence of the small cell sizes, mobile users will typically experience a high number of cell hand-offs during their connections' lifetime.'With shadow clust.ers, the quality of service of mobile calls can be improved by reducing the number of dropped calls during hand-offs, and by disallowing the establishment of new calls that are highly likely to later result in a dropped call.The framework of a shadow cluster system is completely distributed, and can be viewed as a message system where a mobile terminal informs the base stations in the neighborhood about i.ts requirements, position, and movement parameters, so that the base stations project future demands, reserve resources accordingly, and admit onlv

The major focus of this paper is distributed call admission control in mobile/wireless networks the purpose of which is to limit the call hand-off dropping probability in loss systems or the cell overload probability in lossless systems. Hand-off dropping or cell overload are consequences of congestion in wireless networks. Our call admission control algorithm takes into consideration the number of calls in adjacent cells in addition to the number of calls in the cell where a new call request is made, in order to make a call admission decision. This is done by every base station in a distributed manner without the involvement of the network call processor. The admission rule is simple enough that the admission decision can be made in real time. Furthermore, we show that our distributed call admission control scheme limits the hand-off dropping or the cell overload probability to a predefined level almost independent of load conditions. This is an important requirement of future wireless/mobile networks with quality-of-service provisioning.

An architecture is presented for a high-speed cellular radio access network based on ATM transport technology. Central to this approach is a new concept known as the virtual connection tree which avoids the need to involve the network call processor for every cell handoff attempt. Such an approach can readily support a very high rate of handoffs, thereby enabling use of physically small radio cells to provide very high system capacity, but may occasionally cause the volume of traffic to be handled by one cell site to exceed that cell site's capacity. A simple analytical methodology is developed which can be used for admission control, the purpose of which is to limit the number of in-progress calls such that two new quality of service metrics (overload probability and average time in overload) can be kept suitably low. Finally, a general framework is presented for overall system organization and signaling.>

The authors investigate various performance measures of interest, when fast packet-switched networks that operate with either fixed or variable packet sizes are compared. These performance measures include queue length distribution, packet loss probability, and user frame loss probability. The focus is on identifying key parameters that influence the outcome of this comparison, and on quantifying the potential benefits of each approach.>

The authors propose a computationally simple approximate expression for the equivalent capacity or bandwidth requirement of both individual and multiplexed connections, based on their statistical characteristics and the desired grade-of-service (GOS). The purpose of such an expression is to provide a unified metric to represent the effective bandwidth used by connections and the corresponding effective load of network links. These link metrics can then be used for efficient bandwidth management, routing, and call control procedures aimed at optimizing network usage. While the methodology proposed can provide an exact approach to the computation of the equivalent capacity, the associated complexity makes it infeasible for real-time network traffic control applications. Hence, an approximation is required. The validity of the approximation developed is verified by comparison to both exact computations and simulation results.>