In this paper, we propose a new consolidation algorithm called the Selective Backward Resource Management (BRM) cell Feedback (SBF) algorithm. It achieves a fast response and low consolidation noise by selectively forwarding BRM cell from the most congested branch to the source instead of waiting from all branches. Mathematical models are derived to quantitatively characterize the performance, i.e. the response time and ACR of the source, of SBF and previously proposed algorithms. The interoperation of consolidation algorithms in point-to-multipoint available bit rate (ABR) is investigated. We address response time, consolidation noise and the effect of asymmetrical round trip delay (RTD) from branch point to destinations aspects. All combinations of four different consolidation algorithms are interoperated in both local/metropolitan area network (LAN/MAN) and wide area network (WAN) configuration. By a simulation method, we found that the consolidation algorithm used at the uppermost stream branch point, especially in WAN configuration, plays an important role in determining the performance of the network. However, consolidation algorithm used at the lower stream branch point affects the network performance insignificantly. Hence, in order to achieve a good and effective performance of the consolidation algorithms interoperated network, a fast response with low consolidation noise algorithm should be used at the uppermost stream branch point and a simple and easy to implement algorithm should be used at the lower stream branch point.

Most conventional studies on self-stabilization have been indifferent to the vulnerability under convergence. This paper investigates how mutual exclusion property can be achieved in self-stabilizing rings even for illegitimate configurations. We present a new method which uses a state with a large state space to detect faults. If some faults are detected, every process is reset and not given a privilege. Even if the reset values are different between processes, our protocol mimics the behavior of Dijkstra's unidirectional K-state protocol. Then we have a fast and safe mutual exclusion protocol. Simulation study also examines its performance.

It is a fundamental problem to construct a virtual path layout minimizing the hop number as a function of the congestion for a communication network. It is known that we can construct a virtual path layout with asymptotically optimal hop number for a mesh of trees network, butterfly network, cube-connected-cycles network, de Bruijn network, shuffle-exchange network, and complete binary tree network. The paper shows a virtual path layout with minimum hop number for a complete binary tree network. A generalization to complete k-ary tree networks is also mentioned.

This paper proposes a credit-based congestion control scheme for multicast communication which employs application-specific processing at intermediate network nodes. The control scheme was designed not only to take advantage of credit-based flow control for unicast communication, but also to achieve flexibility supported by active network technology. The resultant active multicast congestion control scheme is able to meet the different requirements of various multicast applications in terms of reliability and end-to-end latency. The performance of the proposed control scheme was evaluated using both discrete-event simulations and experiments on a prototype active network implementation. The results show that the proposed scheme performs very well in terms of fairness, responsiveness, and scalability. The implementation experiences also confirmed the feasibility of the scheme in practice.

In the paper, we propose a congestion control scheme for reliable multicast communication which enables TCP fairness and prevents a drop-to-zero problem. The proposed congestion control scheme is rate-based one based on NAKs from receivers and cooperatively works with a flow control scheme. The congestion control scheme consists of two components of a rate-based controller and a selection mechanism of a representative. The rate-based controller runs between the sender and the representative and achieves TCP fairness and fast response to losses at the representative. The selection mechanism of the representative allows the sender to select the representative in a scalable manner, in which the sender makes use of NAKs from receivers to select it. In the paper, we also propose the switchover mechanism of the flow and congestion control schemes which enables the sender to use either of them adaptively based on network situations. When the network is congested, the congestion control scheme works to share network resources fairly with competing TCP flows. Otherwise, the flow control scheme works to adapt the transmission rate to the slowest receiver. We verify the performance of our proposed schemes by using computer simulation.

This paper describes IP traffic, especially the control of VoIP traffic, on the carrier-scale, and proposes algorithms for it. It examines a case that has already been introduced in the United States and discusses the trend of standardization for this control. Control techniques that will be introduced into the IP network in the future are considered from the viewpoints of both "quality" that users receive and the "control" that carriers perform.

We propose new lossless medical image compression method based on hierarchical sorting technique. Hierarchical sorting is a technique to achieve high compression ratio by detecting the regions where image pattern varies abruptly and sorting pixel order by its value to increase predictability. In this method, we can control sorting accuracy along with size and complexity. As the result, we can reduce the sizes of the permutation-tables and reuse the tables to other image regions. Comparison using experimental implementation of this method shows better performance for medical image set measured by X-ray CT and MRI instruments where similar sub-block patterns appear frequently. This technique applies quad-tree division method to divide an image to blocks in order to support progressive decoding and fast preview of large images.

Layered multicast approach enables IP multicast to adapt to heterogeneous networks. In layered multicast, each layer of a session is sent to separate multicast groups. These layers will be transmitted on the same route, or on different routes. However, traditional congestion control schemes of layered multicast do not consider the case when layers of a session are transmitted on different routes. In this paper, at first we show that in sparse-mode routing protocols like PIM-SM and CBT, layers of a session can be mapped to different Rendezvous Points or cores due to the bootstrap mechanism. It means that layers of a session can be transmitted on different routes. We then show that traditional congestion control schemes of layered multicast do not work properly in sparse-mode routing regions. At last we introduce Rendezvous Point based Layered Multicast (RPLM), a novel congestion control scheme suitable for sparse-mode routing regions, and show that RPLM works efficiently in regions using sparse mode routing protocols. RPLM uses per-RP packet loss rate instead of the overall one to detect congestion on each route, and can react to congestion quickly by dropping the highest layer on the congested route. In addition, RPLM simultaneously drops all the layers those are useless in quality's improvement to prevent bandwidth waste.

The problem of detecting straight lines arises frequently in image processing and computer vision. Here we consider the problem of extracting lines from range images and more generally from sets of three-dimensional (3D) points. The problem is stated as follows. Given a set Γ of points in 3D space and a non-negative constant , determine the line that is at a distance at most ε from the maximal number of points of . The extraction of multiple lines is achieved iteratively by performing this best line detection and removing at each iteration the points that are close to the line found. We consider two approaches to solve the problem. The first is a simple approach that selects the best line among a randomly chosen subset of lines each defined by a pair of edge points. The second approach, based on tabu search, explores a larger set of candidate lines thus obtaining a better fit of the lines to the points. We present experimental results on different types of three-dimensional data (i) range images of polyhedral objects (ii) secondary structures (helices and strands) of large molecules.

We have developed an airport monitoring system that traces the movement of airplanes in the parking areas of airports. For this system, we have developed an image processing method, a two-stage normalized background subtraction method that can detect moving objects and determine the sizes of those objects under illumination changes, which are inevitable for outdoor monitoring systems. The two-stage method consists of local and global normalized subtraction. With this method, airplanes can be detected in a stable manner under illumination changes, which means that the brightness in each pixel is not constant due to changes in atmospheric phenomena, such as the shadows of clouds. And false detection problems due to the presence of boarding bridges are solved by utilizing differences in motion between an airplane and the boarding bridge, such as the direction of movement. We have evaluated this method using 140 hours of video images that contain scenes with a variety of conditions, such as the presence of cloud shadows, the turning on and off of lights, night, rainfall and so on. As a result, we have confirmed a 95% level of accuracy of airplane detection. This system is now in operation at Kansai International Airport and is performing most satisfactorily.

This paper presents a novel image coding scheme based on separate coding of region and residue sources. In a subband image coding scheme, quantization errors in each subimage spread over the reconstructed image and result in a blurring or a boundary artifact. To obtain high compression ratio without considerable degradation, an input image, in our scheme, is separated into region and residue sources which are coded using different coding schemes. The region source is coded by adaptive arithmetic coder. The residue source is coded using multiresolution subimages generated by applying a subband filter. Each block in the subimages is predicted by an affine transformation of blocks in lower resolution subimages. Experimental results show that a high coding efficiency is achieved using the proposed scheme, especially in terms of the subjective visual quality and PSNR at low bit-rate compression.

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.

In this paper, an attempt was made to evaluate mental workload using chaotic analysis of EEG. EEG signals registered from Fz and Cz during a mental task (mental addition) were recorded and analyzed using attractor plots, fractal dimensions, and Lyapunov exponents in order to clarify chaotic dynamics and to investigate whether mental workload can be assessed using these chaotic measures. The largest Lyapunov exponent for all experimental conditions took positive values, which indicated chaotic dynamics in the EEG signals. However, we could not evaluate mental workload using the largest Lyapunov exponent or attractor plot. The fractal dimension, on the other hand, tended to increase with the work level. We concluded that the fractal dimension might be used to evaluate a mental state, especially a mental workload induced by mental task loading.

This paper proposes two modes of the congestion control scheme to improve its behavior during the start-up period of networks in current TCP over ATM-UBR implementation. The proposed two modes are a single packet loss mode and a multiple packet losses mode. The proposed algorithm is to minimize the number of cell losses in the ATM switch during specially the start-up period. During the start-up period, multiple packet losses often happens because a TCP sender starts with default parameters. It often ends up sending too many packets and too fast, leading to multiple losses is packet burstiness which occurs right after fast recovery ends. We analyze the transition behavior during fast recovery algorithm and estimate the number of new packets sent when multiple packet losses detected. We present a simple simulation model and numerical results to investigate its performance of the proposed algorithms.

In this paper, we survey the fairness issues in the congestion control mechanisms of TCP, which is one of most important service aspects in the current and future Internet. We first summarize the problems from a perspective of the fair service among connections. Several solution methods are next surveyed. Those are modifications of TCP congestion control mechanism and router support for achieving the fair service among TCP connections. We finally investigate the fair share of resources at endhosts.

In order to increase the flexibility and the extensibility for packet transmission, the active network approach, that makes network nodes programmable, is very promising. Every packet includes a program specifying behavior of the packet at network nodes, and the network nodes only have the functions to execute programs of the packet. In an active network, network congestion management is achieved more flexibly and intelligently than that in the conventional internet. A number of mechanisms for congestion management are easily developed because various network functions such as QoS routing and congestion detection are easily combined in the same network nodes. By executing different routing protocols that use different levels of the link information, a congestion management scheme can be customized by users. This paper shows how QoS routing and congestion detection achieve a congestion management over an active internetwork system.

We propose a fast and compact longest match table look-up method for very long network addresses like IP version 6. This method uses two ideas for a routing-table arranged in a tree-structure. The first idea is to make table look-up fast by caching pointers to intermediate nodes in the tree, reducing the number of node traversals. The second idea is to reduce the memory size required for each node in the tree by one-third by eliminating common parts of addresses of adjacent nodes. Evaluating the performance of this method by using actual routing table data of an IP backbone network, we found it was five to ten times faster than a conventional method.

This paper proposes a unified coding algorithm for the lossless and near-lossless compression of still color images. The proposed unified color image coding scheme can control the Peak Signal-to-Noise Ratio (PSNR) of the reconstructed image while the level of distortion on the RGB plane is suppressed to within a preset magnitude. In order to control the PSNR, the distortion level is adaptively changed at each pixel. An adaptive quantizer to control the distortion is designed on the basis of psychovisual criteria. Finally, experiments on Super High Definition (SHD) images show the effectiveness of the proposed algorithm.

This paper describes a 1-GHz portable digital delay-locked loop (DLL) with 0.15-µm CMOS technology. There are three factors contributing to jitter in digital DLLs. One is supply-noise induced jitter, another is jitter caused by delay time resolution and phase step in the delay line, and the third is jitter caused by the sensitivity of the phase detector. In order to achieve a low jitter digital DLL, we have developed a master-slave architecture that achieves infinite phase capture ranges and low latency, a delay line that improves the delay time resolution, a phase step suppression technique and a dynamic phase detector with increased sensitivity. These techniques were used to fabricate a digital DLL with improved jitter performance. Measured results showed that the DLL successfully achieves 29-ps peak-to-peak jitter with a quiet supply and 0.2-ps/ mV supply sensitivity.

In order to ease the impact of the packet fragmentation problem and to avoid network congestion in TCP over UBR, packet discard schemes in ATM layer (such as PPD and EPD) have been proposed. These schemes drop packets before they reach their intended destinations if the network is congested and the packets are to be partially discarded. On the other hand, TCP also regulates data flow with its own flow control method. Due to restriction of data flow at the TCP layer, buffer space is not fully used in an ATM switch. In order to make use of more buffer resources, this paper generalizes the PPD and EDP schemes. From this generalization, an optimistic packet discard scheme named the "Probability-Based Delayed Packet Discard" (PDPD) scheme is proposed. Depending on a particular probability, this scheme sets a discard flag to delay actual discard operation. This paper presents the results of several simulated models to find out the potential of improvement of goodput by PDPD. The results of these simulations indicate that PDPD obtains higher goodput than ordinary schemes when the packet size is large and the input load is not light. This author concludes that a PDPD scheme should achieve effective goodput and link utilization while using more buffer resources effectively.