Location based services(LBS) are becoming increasingly important to the success and attractiveness of next generation wireless systems. However, a natural tension arises between the need for user privacy and the flexible use of location information. In this paper we present a framework to support privacy enhanced location based services. We classify the services according to several basic criteria and we propose a hierarchical key distribution method to support these services. The key idea behind the system is to hierarchically encrypt location information under different keys, and distribute the appropriate keys only to group members with the necessary permission. Four methods are proposed to deliver hierarchical location information while maintaining privacy. We propose a key tree rebalancing algorithm to maintain the re-keying performance of the group key management. Furthermore, we present a practical LBS system implementation. Hierarchical location information coding offers flexible location information access which enables a rich set of location based services. Our load tests show such a system is highly practical with good efficiency and scalability.

Location based services are becoming increasingly important to the success and attractiveness of next generation wireless systems. Service providers will use location information to introduce new services and greatly enhance many existing services. Maintaining location privacy is an important requirement that must be met for these services to be widely deployed. It is a challenge to maintain location privacy while still providing the flexible access to location information required to enable a rich set of location based services. In this paper we define a high-level architecture for providing LBS and classify services according to several basic criteria. To support these services we propose a hierarchical key distribution method. Four methods are proposed to deliver hierarchical location information while maintaining privacy. We evaluate the efficiency of the system in terms of message delivery and key management overhead

Typical distributed transaction environments are a heterogeneous collection of hardware and software resources. An example of such an environment is an electronic store front where users can launch a number of different transactions to complete one or more interactions with the system. One of the challenges in managing such an environment is to figure out the root cause of a performance or throughput problem that manifests itself at a user access point, and to take appropriate action, preferably in an automated way. Our paper addresses this problem by analyzing the dependency relationship among various software components. We also provide a theoretical insight into how a set of transactions can be generated to pinpoint the root cause of a performance problem that is manifested at the user access point.

Studies have shown that e-Learning is one of the fasted growing trends in higher education. This evolving trend can be classified with respect to the underlying structure and content of these environments. Of particular note is an e-Learning environment which is structured asynchronously and whose content is based solely on an electronic medium. In such environments, classes may never actually meet and the only form of delivery may be via an electronic lecture system. Given this environment, how do we maintain the benefits of the traditional classroom setting while still offering students the flexibility they may need in having the asynchronous environment in the first place? Furthermore, given an electronic-only delivery mechanism, how can both the student and the instructor tailor e-Learning content to suit their needs. We propose a flexible media-rich instructional method which: 1) facilitates collaboration in an asynchronous, electronic lecture environment and 2) supports the context-based, post-production augmentation and exchange of user-defined supplements which allows an author to refine and tailor e-Learning content.

E-learning or learning via the Web is now changing the way universities and coprporations offer education and training. Rather than using conventional means for remotely distributing and delivering courses and classroom lectures, universities and enterprises are moving towards Web-based academic and training offerings. While content acquisition and distribution can be automated in a systematic fashion, research challenges remain in the area of personalized content delivery, access, and augmented interaction. Automated content indexing and annotation for easy and efficient access and navigation of various media elements associated with the course material become key takss in flexible content delivery and use. E-Learning Media Navigator (ELM-N) from IBM Research is a system with which a user can access and interact with online courses presented as audiovisual material, along with related slides and white broad contents in a time-synchronized fashion. In this demonstration, the system capabilities and research technologies behind them will be presented.

The bulk of proposed pervasive computing devices such as PDAs and cellular telephones operate as thin clients within a larger infrastructure. To access services within their local environment, these devices participate in a service discovery protocol which involves a master directory that registers all services available in the local environment. These directories typically are isolated from each other. Devices that move across service discovery domains have no access to information outside their current local domain. In this paper we propose an application-level protocol called VIA that enables data sharing among discovery domains. Each directory maintains a table of active links to other directories that share related information. A set of linked directories forms a data cluster that can be queried by devices for information. The data cluster is distributed, self-organizing, responsive to data mobility, and robust to failures. Using application-defined data schemas, clusters organize themselves into a hierarchy for efficient querying and network resource usage. Through analysis and simulation we describe the behavior of VIA under different workloads and show that the protocol overhead for both maintaining a cluster and handling failures grows slowly with the number of gateways.

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.

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 problem of obtaining a minimum cost topology for a mesh network given matrices specifying the cost of links between all pairs of nodes and the internode requirements is considered. A heuristic algorithm which works in terms of general network design principles and uses utilization as a figure of merit is presented. The procedure is applicable to a wide variety of networks, especially to the problem of obtaining starting topologies for other network design procedures. The algorithm's computational complexity is shown to be of order N/sup 2/, a significant improvement over currently used algorithms and fast enough to be embedded in the inner loop of other more general design procedures, e.g., node selection procedures. Computational experience is presented which shows that the procedure is fast and simple and yields solutions of a quality competitive with other much slower procedures.>

The performance evaluation tool (PET) is a package to aid the designers of architecture and protocols of computer networks. It is specially targeted, but not limited, to IBM's systems network architecture (SNA) networks. The tool allows its users the capability to easily construct simulation models of networks and conduct traffic related performance studies. PET is based on the research queueing package (RESQ)-version 2, which is a package for constructing and analyzing extended queueing network models. The two key features of PET are ease of use and flexibility. The user specifies the network configuration with the aid of an interactive "networking language." This includes a description of the type of representation he/she desires for the various protocols at different layers of the architecture. Based on these specifications, PET builds a RESQ simulation model for the network by piecing together many "submodels." Each submodel is a model of a certain network component (e.g., communication link) or a protocol (e.g., data link control). The submodels may either be prebuilt (stored in the PET library) or built dynamically according to user specifications. There can be several submodels for the same function capturing different levels of detail. Thus, a variety of network models with different types of components and with varying levels of detail can be constructed. Facilities to perform parametric studies and provide user specified output statistics (such as link utilizations and buffer usage statistics) are also available in PET.

This paper is concerned with a comparison study of three switching techniques used in computer-based communication networks: circuit switching, message (packet) switching, and cut-through switching. Our comparison is based on the delay performance as obtained through analytic models of these techniques. For circuit switching, the model reflects the phenomenon of channel reservation through which it can be shown that when circuit switching is used, data communication networks saturate rapidly. Through numerical examples, it is shown that the boundary between the areas of relative effectiveness of these switching techniques depends very much on the network topology (more precisely the path length of communication), the message length, and the useful utilization.

In a recent paper we presented an analysis of flow control in store-and-forward computer communication networks using a token mechanism. The analysis assumed equilibrium conditions for a selected set of system parameters which were not dynamically adjusted to stochastic fluctuations in the system load; this mechanism was referred to as "static flow control." In this paper we study a "dynamic flow control" in which parameters of the system are dynamically adjusted to match the availability of resources in the network. Based on Markov decision theory, an optimal policy to dynamically select the number of tokens is formulated. Because an exact solution to the problem is extremely difficult, an effective heuristic solution to the problem is presented. Numerical results are given and it is shown that the throughput-delay performance of a network is better with dynamic control than with static control.

In this paper we develop an analytic model for end-to-end communication protocols and study the window mechanism for flow control in store-and-forward (in particular message-switching) computerbased communication networks. We develop a static flow control model in which the parameters of the system are not dynamically adjusted to the stochastic fluctuation of the system load. Numerical results are presented and it is shown that the throughput-delay performance of a network can be improved by proper selection of the design parameters, such as the window size, the timeout period, etc.