The Active HYpermedia Delivery System (AHYDS) facilitates the access to multimedia information over a large-scale network and wide spectrum of media. We developed intelligent access facilities that build on the access paradigms supported by current web applications. This facility generalizes not only different kinds of logical data models (relational, object, hyperlink), but also access mechanisms of multimedia applications to make them customizable and scalable. This paper proposed the distributed management mechanism of the AHYDS platform. The major contribution of this paper is the mechanism for distributed multimedia delivery management over large-scale network and heterogeneous environment. We also propose the mechanism to manage huge multimedia data.

A remote auscultation support system was developed that compresses and records in real time the patient's breath sound and heart sound, obtained using a stethoscope, and sends this data to an attending doctor at a hospital via network. For real-time recording of the breath sound and heart sound, special-purpose, high-quality sound coding technology was developed and incorporated in the system. This sound coding technology enables the amount of data to be reduced to about 1/18 with virtually no deterioration of the properties of the auscultation sound, high-speed transmission of this data using network, and remote diagnosis of the auscultation sound by a medical specialist. The auscultation locations of each patient, together with the doctor, stethoscoper, and patient database are input into the system in advance at the hospital. At the patient's home or sanatorium, the auscultation sound is recorded according to a human body display that shows auscultation locations, and then sent to the hospital. To ensure patient confidentiality when the auscultation data is transmitted via network, the system scrambles the auscultation data and allows only the attending doctor to play and diagnose the auscultation sound. These features not only support an understanding of the condition of patients being treated at home, but they also enable the construction of an auscultation database for electronic charts that allows auscultation results to be shared within the hospital. When this remote auscultation support system was manufactured and its performance was assessed, virtually the same waveform was obtained for the recorded and played breath sound as for the original breath sound. Results showed that even at a sampling frequency of 11 kHz, remote diagnosis by a medical specialist was in fact possible. Furthermore, if auscultation data of 10 seconds per location for 10 locations is sent, the amount of data sent is only about 120 Kbytes. Since this amount of data converts to only about 25 pages of electronic mail text, even via the existing mobile network the auscultation sounds of many patients can be sent efficiently.

Certificate Revocation is a critical issue for a practical, public-key infrastructure. A new efficient revocation protocol using a one-way hash tree structure (instead of the classical list structure, which is known as a standard for revocation), was proposed and examined to reduce communication and computation costs. In this paper, we analysis a k-ary hash tree for certificate revocation and prove that k = 2 minimizes communication cost.

We propose the Geographical Location Information (GLI) system that maps a mobile entity on the Internet to a geographical position. Users can look up the latest geographical location information of registered mobile entities (forward-lookup) and can also search for mobile entities within a specified area (reverse-lookup). The GLI system consists of home and area servers. The home server maintains latest geographical location information of the mobile entities and processes forward-lookup requests. The area server maintains the latest geographical location information of the mobile entities in the area that it manages and processes reverse-lookup requests. To provide a highly scalable system, home and area servers are managed in a distributed manner based on a hierarchical server structure and delegation of authority to servers that manage lower layers. To reduce the amount of traffic due to distributed management, the delegation information of authority is cached by the servers. In our performance evaluation of the GLI system, the prototype implementation can handle 4,500-8,000 requests/sec for location lookup and location registration. We found that 52 home servers and 33 area servers are enough to handle all cars in Japan under some assumed parameters through the performance evaluation.

In this paper, a novel beam selection transmit diversity (BSTD) scheme is proposed for the downlink transmission of frequency division duplex (FDD) based DS-CDMA system. As a combination of selection transmit diversity and steering vector based beamforming, the BSTD scheme provides diversity gain as well as reducing multiple access interference in downlink. Moreover, to have a better understanding, the performance of the BSTD is also compared with other schemes. The comparison results show that the BSTD would be a promising candidate for the downlink transmission if both performance and implementation complexity are considered.

The capacity analysis of CDMA forward link is required to allocate high data-rate users. To solve this problem, we analyze the capacity of the CDMA forward link based on optimum power control. In a multi-cell environment, the intracell interference and intercell interference are derived. To show the capacity impact of high data-rate users, we introduce a location dependent factor, which is the ratio of intercell interference to intracell interference. Considering the location dependent factor, we propose a coordinated allocation scheme that can be used to allocate high data-rate users in sectorized cells. As a result, we show the capacity impact of high data-rate users according to location variations. The outage probability can be decreased when the coordinated allocation scheme is applied.

ATM networks are connection-oriented. Making a call requires first sending a message to do an admission control to guarantee the connections' QoS (quality of service) in the network. In this paper, we focus on the problem of translating a global QoS requirement into a set of local QoS requirements in ATM networks. Usually, an end-user is only concerned with the QoS requirements on end-to-end basis and does not care about the local switching node QoS. Most of recent research efforts only focus on worst-case end-to-end delay bound but pay no attention to the problem of distributing the end-to-end delay bound to local switching node. After admission control, when the new connection is admitted to enter the network, they equally allocate the excess delay and reserve the same bandwidth at each switch along the path. But, this can not improve network utilization efficiently. It motivates us to design a novel local QoS requirement allocation scheme to get better performance. Using the number of maximum supportable connections as the performance index, we derive an optimal delay allocation (OPT) policy. In addition, we also proposed an analysis model to evaluate the proposed allocation scheme and equal allocation (EQ) scheme in a series of switching nodes with the Rate-controlled scheduling architecture, including a traffic shaper and a non-preemptive earliest-deadline-first scheduler. From the numerical results, we have shown the importance of allocation policy and explored the factors that affect the performance index.

The available bit rate (ABR) is an ATM service category that provides an economical support of connections having vague requirements. An ABR session may specify its peak cell rate (PCR) and minimum cell rate (MCR), and available bandwidth is allocated to competing sessions based on the max-min policy. In this paper, we investigate the ABR traffic control from a different point of view: Based on the decentralized bandwidth allocation model studied in [9], we prove that the max-min rate vector is the equilibrium of a certain system of noncooperative optimizations. This interpretation suggests a new framework for ABR traffic control that allows the max-min optimality to be achieved and maintained by end-systems, and not by network switches. Moreover, in the discussion, we consider the constrained version of max-min fairness and develop an efficient algorithm with theoretical justification to determine the optimal rate vector.

This paper presents a DFT controller called as a TCU (Test Control Unit), which considerably improves the efficiency of the instruction-based functional test. Internal program/data buses are completely controllable and observable by the TCU during the test cycle. Diverse test modes of the TCU can increase the test efficiency and also provide complete access to program/data memories for functional test.

The Tropical Rainfall Measuring Mission (TRMM) is a United States-Japan joint project to measure rainfall from space. The first spaceborne rain radar is aboard the TRMM satellite. Rain/no-rain discrimination for the TRMM provides useful information for on-line data processing, storage, and post-processing analysis. In this paper, rain/no-rain discrimination for the TRMM has been validated through simulation and theory for the no-rain condition and by comparison with the ground-based radar data for rain conditions.

As Information Technology progresses, our daily lives are getting "connected" more and more. At the same time, however, problems are appearing. The center of these problems can be captured as the "Communication Overflow. " To cope with such problems, we propose an approach that tries to provide a communication environment that assists users in managing their communication activities. The key notion of this approach is to enhance the "Awareness of Connectedness. " Here, agents which are suggestive of awareness of connectedness play an important role. In this paper, we describe the key notion and introduce a brief road-map towards the environment for the awareness of connectedness. Two candidate tools for the environment are described. The first one is a visualization tool for communication media that provides feedback of users' communication activities. Its purpose is to enhance the awareness for communication. The second tool is a simple, intuitive interactive media that exchanges the statuses of users. It is an alternative network communication media that might be suitable for very light-weight, almost-acknowledge-only communication mode. Some results on an experiment of these two tools are also reported.

New equivalent characterizations are derived for Schur stability property of real polynomials. They involve a single scalar parameter, which can be regarded as a freedom incorporated in the given polynomials so long as the stability is concerned. Possible applications of the expressions are suggested to the latest results for stability robustness analysis in parameter space. Further, an extension of the characterizations is made to the matrix case, yielding one-parameter expressions of Schur matrices.

A lot of efforts have been made to develop the international Internet environment. In order to achieve a better and efficient information infrastructure around the globe, there are very strong urgent demands in the area of Asia and Pacific for a practical working environment where engineers and researchers in related fields work cooperatively. Our AI3 Project was started in 1995 by WIDE Project and JSAT. It has been operating a satellite based testbed network in Southeast Asia and conducting a series of research activities using the testbed. In this paper, we explain AI3 satellite Internet infrastructure and describe our efforts of the deployment in Asia using our developed technology. Furthermore, we evaluate our testbed network from the aspect of sustainable international collaboration.

In the future Internet, various kinds of services will rapidly increase its volume and require different qualities. Thus, it is main technical problem to guarantee various QoS to each connection. However, in the current IP networks where most applications use TCP as transport protocol, most typical packet discarding scheme, RED (Random Early Detection), causes unfairness such as the difference of bandwidth sharing among flows traversing the same router. To dissolve this problem, we consider that two viewpoints are important associated with flow-base fairness. One is instantaneous flow condition and the other is historical flow condition. In this paper, we propose the packet discarding scheme considering both instantaneous and historical use of network resources for the purpose of dissolving unfairness of each flow and improving the flow-base QoS. We call this proposal method "Random Early Drop with Dual-fairness metrics (DRED). " DRED can improve whole throughput and transfer completion time of information such as a file, etc.

Recently, the Guaranteed Frame Rate (GFR) service was proposed as a new service category of ATM to support non-realtime data applications and to provide the minimum rate guarantee. To keep the simplicity of GFR as much as possible and overcome defects of FIFO-based mechanisms, we propose a FIFO-based algorithm extending DFBA one to improve the fairness and provide the minimum rate guarantee for a wider range of Minimum Cell Rate (MCR). The key idea is controlling the number of CLP1 cells which are occupying more buffer space than the fair share even when the queue length is below Low Buffer Occupancy (LBO).

For the microwave level meter based on the FM-CW radar, we analyze the spectrum correlation of beat signals and propose a measurement algorithm using the fact that there exists a peak in the spectrum correlation of beat signals when range difference is sufficiently small. This algorithm can compensate the nonlinear effect of VCO frequency sweep, making it possible to determine the range difference in a precise manner even using a practical VCO. We present some experimental results to show the validity of this algorithm.

Wavelength converters are usually used for improving the performance of WDM optical networks. From the viewpoint of network economics and current technologies, the wavelength converters with a limited conversion range are necessary to be used sparsely in real applications. However, there have been little efforts for developing wavelength assignment algorithm that achieves a guaranteed high performance with either limited or sparse wavelength conversion. In this paper, we propose a new wavelength assignment algorithm that can be applied to both sparse and limited wavelength conversion. Through the results of simulation program, we show that the proposed algorithm outperforms other ones presented until now.

In the past, a number of scheduling algorithms that approximate GPS, such as WFQ, have been proposed and have received much attention. This class of algorithms provides per-flow QoS guarantees in terms of the bounded delay and minimum bandwidth guarantee. However, with O(log N) computational cost for each new arrival scheduling, where N is the number of backlogged flows, these algorithms are expensive to implement (e.g., in terms of scalability). Moreover, none of them addresses the issues of delay distribution and jitter. In this paper, we propose a new traffic scheduling discipline called Jitter Control Frame-based Queueing (JCFQ) that provides an upper bound for delay jitter in the case of rate-controlled connections, such as packet video streams and IP telephony, while guaranteeing bounded delay and worst-case fair weighted fairness, such as in the WF2Q algorithm, but with O(1) complexity in selecting the next packet to serve, assuming that the number of flows is fixed. Three different algorithms for slot or service order assignment between flows are proposed: Earliest Jitter Deadline First (EJDF), Rate Monotonic (RM) and Maximum Jitter First (MJF). In these algorithms, delay jitter is formulated into the virtual finish time calculation. We compare the fairness, delay and jitter performance of the JCFQ with that of the MJF algorithm with WF2Q via simulation. The results show that with proper choice of the slot size, JCFQ can achieve better flow isolation in delay distribution than can WF2Q.

It has been reported that IP packet traffic exhibits the self-similar nature and causes the degradation of network performance. Therefore it is crucial for the appropriate buffer design of routers and switches to predict the queueing behavior with self-similar input. It is well known that the fitting methods based on the second-order statistics of counts for the arrival process are not sufficient for predicting the performance of the queueing system with self-similar input. However recent studies have revealed that the loss probability of finite queuing system can be well approximated by the Markovian input models. This paper studies the time-scale impact on the loss probability of MMPP/D/1/K system where the MMPP is generated so as to match the variance of the self-similar process over specified time-scales. We investigate the loss probability in terms of system size, Hurst parameters and time-scales. We also compare the loss probability of resulting MMPP/D/1/K with simulation. Numerical results show that the loss probability of MMPP/D/1/K are not significantly affected by time-scale and that the loss probability is well approximated with resulting MMPP/D/1/K.

This paper considers a high-rate turbo code which employs high-rate convolutional codes as component codes, and presents a novel method of reducing the decoding complexity of the codes. By eliminating some of branches that have the lowest reliabilities among all the branches entering each node, the proposed algorithm reduces the complexity in the process of the add-compare-select (ACS) between the consecutive stages of iterative decoding. That is, the complexity gradually decreases as the number of iterations increases. We compare the unpunctured high-rate turbo code with a classical punctured high-rate turbo code in terms of performance/complexity trade-off under the same code rate. Simulation results show that the proposed approach with a good trade-off provides an alternative coding scheme to the classical punctured high-rate turbo coding for the application to high-data-rate wireless communication systems.