掲載種別：研究論文（学術雑誌）  

Current IaaS (infrastructure as a service) cloud service may not satisfy communication QoS (quality of service) requirements of delay-sensitive network applications, if there is a significant physical distance between a server of the network application (NetApp server) at a data center and its network application clients (NetApp clients). In order to improve communication QoS of NetApp clients, we propose server migration service (SMS) in this article. SMS allows NetApp servers to migrate among different locations in the network (1) to optimally locate themselves in relation to NetApp clients and mitigate the QoS degradation caused by location-related factors (ie, propagation delays on network links) and (2) to optimally distribute traffic load over routers and processing load over (physical) computers and decrease the energy consumption. We develop a mixed-integer programming model that determines when and to which locations NetApp servers migrate to minimize the total operating cost of SMS, that is, the sum of the monetary penalty incurred due to QoS violation and energy cost incurred due to energy consumption, while preventing NetApp servers from excessively migrating and adversely impacting QoS of the non-SMS service that share the resource of the substrate network with SMS. Simulation results show that the model developed in this article achieves up to 42% lower total operating cost of SMS compared to the model that only minimizes the monetary penalty of SMS without considering the energy cost of SMS.

DOI： 10.1002/ett.4511

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掲載種別：研究論文（国際会議プロシーディングス）  

DOI： 10.1109/ICTC55196.2022.9952427

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その他リンク： https://dblp.uni-trier.de/db/conf/ictc/ictc2022.html#DeguchiKTFY22

掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

DOI： 10.1109/ictc52510.2021.9621088

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掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

Edge computing is a promising computing paradigm that satisfies QoS requirements of delay-sensitive applications. In edge computing, server migration control is indispensable for managing client mobility. As a server migration method for edge computing, the method based on Q-learning has been proposed. However, the method assumes that there is only one application client and the number of destination edge servers is limited to one. In this paper, we propose a server migration method using Q-learning that copes with realistic situations where there are multiple application clients and destination edge servers. The contributions of this paper are as follows: 1) we clarify that, under the situation with multiple application clients and multiple destination edge servers, a straightforward server migration method using Q-learning (RL method) does not scale due to state space explosion, and 2) we propose a server migration method using Q-learning (RL-DR method) that reduces the dimensionality of state space by abstracting the numbers of application clients at all locations into a center of the gravity (COG) of application clients. The simulation results show that 1) RL method shows up to 248% worse performance than conventional server migration methods because of state space explosion and 2) RL-DR method achieves up to 38.3% better performance than the conventional methods.

DOI： 10.1109/ICOIN50884.2021.9333965

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その他リンク： https://dblp.uni-trier.de/db/conf/icoin/icoin2021.html#UrimotoFTMY21

掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

Various information is collected from IoT devices through the network. As such device becomes more familiar to the user, services are required to consider the influence of user. However, it is difficult to set the parameters of actuators that build consensus among all users in an environment where people with various preferences coexist. The conventional method minimizes the power consumption under the constraints of the user stress. However, this method has a problem that the calculation overhead is increased as the number of devices and users is increased. In this study, we propose a device control method based on consensus building with reinforcement learning. In the proposed method, the state is reduced by applying reinforcement learning for reducing the calculation overhead. As a result of evaluation, we clarified that our method obtains the device parameters that improve the reward by 1.5 times compared with the conventional method. Moreover, we also clarified that a reward value of 98.6% can be achieved compared to the optimum value.

DOI： 10.1109/ICOIN50884.2021.9333958

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その他リンク： https://dblp.uni-trier.de/db/conf/icoin/icoin2021.html#OishiTFY21

掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

Since network failures have a huge negative impact on our daily life, networks should be recovered from the failures as soon as possible. Although a lot of IP fast reroute methods have been proposed, most methods focus on multiple link failures or a single node failure. This paper proposes an IP fast reroute method which can reroute packets against multiple node failures. The paper is the first paper which deals with multiple node failures in the research area on IP fast reroute. The proposed method generates spanning trees to bypass the failures from a given network topology in network design stage, and reroutes a packet using one of the generated spanning trees every time the packet encounters a node failure in network operation stage. Numerical example shows that such spanning trees can be easily generated using our proposed method.

DOI： 10.1109/ICTC49870.2020.9289623

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その他リンク： https://dblp.uni-trier.de/db/conf/ictc/ictc2020.html#NumataITFY20

掲載種別：研究論文（国際会議プロシーディングス）  

© 2020 ACM. Recurrent neural network method considering traffic volume, timestamp and day of the week (RNN-VTD method) is a promising network traffic prediction method because of its high prediction accuracy. The RNN-VTD method encodes timestamp and day of the week, which are categorical data, to numerical data using label encoding. The label encoding, however, gives magnitude to the encoded values, which may cause misunderstanding of recurrent neural network models, and consequently, the prediction accuracy of the RNN-VTD method may be degraded. In this paper, we investigate the effect of using one-hot encoding instead of label encoding for a feature encoding method in the RNN-VTD method. In the one-hot encoding, each input data is encoded to an k-dimensional 0 - 1 vector where k is the number of category types. Because the encoded data do not have magnitude, it is expected that the prediction accuracy of the RNN-VTD method is improved.

DOI： 10.1145/3395245.3396441

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掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

DOI： 10.1109/ICTC46691.2019.8939877

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その他リンク： https://dblp.uni-trier.de/db/conf/ictc/ictc2019.html#KishimotoOTFY19

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：Institute of Electrical and Electronics Engineers Inc.  

It is crucial for network operators to predict network traffic in the future as accurate as possible for appropriate resource provisioning and traffic engineering. Recurrent neural network (RNN) methods are considered to be the most promising prediction methods because of their high prediction accuracy. In conventional studies, RNN methods use only time series of traffic volume as input, and do not use any attribute information (e.g., timestamp and day of the week) of the time series data. However, traffic volume changes depending on both time and day of the week. Therefore, it is possible that we can further improve the prediction accuracy of the RNN methods by using the attribute information as input, in addition to the time series of traffic volume. In this paper, we investigate the effect of using the attribute information of time series of traffic volume on prediction accuracy in network traffic prediction. We propose two RNN methods: RNN-VT method and RNN-VTD method. The RNN-VT method uses timestamp information and the RNN-VTD method uses both timestamp and day of the week information as input, in addition to the time series of traffic volume. Experimental results show that day of the week information is significantly effective for improving prediction accuracy of the RNN methods while timestamp information is not effective.

DOI： 10.1109/ICTC.2018.8539488

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：Springer New York LLC  

In this paper, we propose a new class of service called server migration service (SMS) to augment the existing IaaS (Infrastructure as a Service). SMS allows servers (server-side processes of a network application) to dynamically and automatically migrate as their clients (client-side processes of a network application) change their locations in order to reduce the total monetary penalty that the SMS provider pays to its SMS subscribers when failing to provide them with the guaranteed level of QoS. In this paper, we consider the monetary impact that arises from QoS degradation due to server migration and build an integer programming model to determine when and to which location servers should migrate to minimize the total monetary penalty incurred by the SMS provider. Numerical examples show that SMS achieves up to 96% lower total monetary penalty compared to that without server migration. Numerical examples also show that the integer programming model developed in this paper requires reasonable computation time under realistic parameter settings.

DOI： 10.1007/s10922-018-9451-6

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掲載種別：研究論文（国際会議プロシーディングス）  

DOI： 10.1109/CloudNet.2018.8549545

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DOI： 10.1109/NTMS.2018.8328725

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：Institute of Electrical and Electronics Engineers Inc.  

When a standard TCP implementation using the minimum retransmission timeout (RTOmin) of 200 ms is used in distributed file systems in data centers, a well-known throughput degradation called TCP Incast occurs, because 200 ms is too large as an RTOmin in data centers. In order to avoid TCP Incast, a TCP implementation using a much smaller RTOmin attained by a fine-grained kernel timer is proposed. However, throughput still decreases in the implementation when the number of servers in distributed file systems is large. In this paper, we propose an improvement method introducing a new TCP option which enables a client to request servers to retransmit packets in a better timing. Numerical examples obtained by extensive simulation runs show the effectiveness of the improved method.

DOI： 10.1109/ICTC.2017.8190962

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：Institute of Electrical and Electronics Engineers Inc.  

The server migration service (SMS) is an optional service that improves communication QoS of the IaaS cloud service. In the SMS, small-scale data centers (micro data centers) called work places (WPs) are deployed at various locations in the network. In the SMS, a network application (a NW-App) consists of one or more server-side processes of the application (servers) and one or more client-side processes of the application (clients). In the SMS, servers migrate among WPs in order to improve communication QoS between the servers and their clients, unlike in the IaaS cloud service where locations of the servers are always fixed at a data center in the network. In our previous study, we developed an integer programming model and solved it to determine when and to which WPs servers should migrate in the SMS in order to minimize the financial penalty, i.e., SMS provider's financial loss due to degradation of the communication QoS. In this paper, we consider the electricity power cost, an important component of the operational cost of the SMS, in addition to the financial penalty considered in our previous study, in determining when and to which WPs servers should migrate in the SMS. In this paper, we define the operational cost of the SMS as the sum of the financial penalty and electricity power cost and consider a power-Aware server location decision problem in order to minimize the operational cost of the SMS. We formulate the problem as a mixed-integer programming model and solve the model numerically. Numerical examples show that our mixed-integer programming model optimally determines the locations of the servers and decreases the operational cost by up to 15.5% compared to our previous integer programming model, where the power cost was not considered.

DOI： 10.1109/ICTC.2016.7763457

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：Academy and Industry Research Collaboration Center (AIRCC)  

In distributed file systems, a well-known congestion collapse called TCP incast (Incast briefly) occurs because many servers almost simultaneously send data to the same client and then many packets overflow the port buffer of the link connecting to the client. Incast leads to throughput degradation in the network. In this paper, we propose three methods to avoid Incast based on the fact that the bandwidth-delay product is small in current data center networks. The first method is a method which completely serializes connection establishments. By the serialization, the number of packets in the port buffer becomes very small, which leads to Incast avoidance. The second and third methods are methods which overlap the slow start period of the next connection with the current established connection to improve throughput in the first method. Numerical results from extensive simulation runs show the effectiveness of our three proposed methods.

DOI： 10.5121/ijcnc.2016.8406

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：Academy and Industry Research Collaboration Center (AIRCC)  

Minimum Physical Hop (MPH) has been proposed as a peer selection algorithm for decreasing inter-AS (Autonomous System) traffic volume in P2P live streaming. In MPH, a newly joining peer selects a peer whose physical hop count (i.e., the number of ASes traversed on the content delivery path) from it is the minimum as its providing peer. However, MPH shows high inter-AS traffic volume when the number of joining peers is large. In this paper, we propose IMPH that tries to further decrease the inter-AS traffic volume by distributing peers with one logical hop count (i.e., the number of peers or origin streaming servers (OSSes) traversed on the content delivery path from an OSS to the peer) to many ASes and encouraging the following peers to find their providing peers within the same AS. Numerical examples show that IMPH achieves at the maximum of 64% lower inter-AS traffic volume than MPH.

DOI： 10.5121/ijcnc.2016.8103

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

In this paper, we design LM-MIMO system using high-order m-PSK modulation and evaluate performance of the system. As simulation results, we can confirm that LM-MIMO system with high-order m-PSK modulation has a similar performance to conventional MIMO system with a number of RF chains.

DOI： 10.1109/ICCE.2016.7430711

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：SPRINGER  

Light trail networks have been proposed as an extension of lightpath networks in order to enable sub-wavelength bandwidth allocation in WDM networks. The networks need a medium access control (MAC) method to avoid collisions of data transmissions. We have proposed a MAC method which adopts a token-passing mechanism. The method splits one trail into two trails: one upstream trail and one downstream trail and permits independent data transmissions on the trails. However, the method leaves bandwidths of links upstream from the token-holding node idle because the method only splits the original trail into two trails and leaves the upstream trail unsplit. In this paper, we propose a novel token-passing MAC method for further improving trail throughputs. Our proposed method recursively splits the upstream trail, and consequently the original trail can be split into more than two trails. In the proposed method, an appropriate setting of upstream/downstream token-holding times of all nodes is a key factor to successfully accommodate an input traffic. Therefore, we tackle a problem (called a token-holding time decision problem) where we optimize token-holding times of all nodes so that input traffic is successfully accommodated. We formulate the problem as a linear programming model. Numerical examples on maximum effective throughput show that the proposed method is more superior as the ratio of short-hop traffic is higher in input traffic patterns.

DOI： 10.1007/s11107-015-0509-z

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：Information Processing Society of Japan  

We have developed a system (traditional system) to flexibly provide the requested applications environment on educational Windows PCs. The traditional system dynamically controls the execution of applications installed on each educational PC depending on the rules defined by teachers as well as by administrators. The traditional system, however, has a low tolerance for malicious attacks. If the execution file of a certain application is falsified, the corresponding rules already applied become invalid. In addition, though the traditional system has a function to define groups of controlled applications, it does not support hierarchical groups. This reduces the usability of the traditional system. In order to address these issues, this paper proposes a control method of application execution using digital certificates. The proposed method has a high tolerance for the falsification of execution files by controlling their executions based on the reliability of the corresponding digital certificates. It also improves its usability by introducing hierarchical group management utilizing hierarchical structure for digital certificates.

DOI： 10.2197/ipsjjip.23.449

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

In IaaS cloud services, QoS of network applications (NW-Apps) may degrade due to location factors such as significant distance between a server-side application (server) of a NW-App at a data center and a client-side application (client) of the NW-App at a client terminal. In order to shorten the distance and to improve the QoS, server migration services (SMSes) have been proposed. In SMSes, servers may migrate between different computers (called work places, WPs) on a network to prevent QoS degradation caused by the changes of client locations. Although server migrations can improve QoS of NW-Apps, they also generate a huge amount of traffic (server migration traffic) in the network. This paper focuses on a server location decision problem where the location of a server is decided in an on-line manner so that QoS of a NW-App is improved under the constraint that the server migration traffic has to be suppressed below an acceptable level. For the problem, we propose a practical on-line algorithm. The key idea behind the proposed algorithm is that the location of the server is decided with consideration of the QoS degradation in the future. The algorithm defines the averagely good location for the server where the QoS is expected to be relatively good for various client locations. Then, it keeps the range of the server's migration within the returnable range where the server can soon come back to the averagely good location. As a result, the QoS can be always kept as good as the one under the averagely good location. Simulation results show that the proposed algorithm improves QoS of the NW-App by up to 30% compared to a greedy algorithm.

DOI： 10.1109/CCNC.2015.7158070

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

The demand for high Internet availability has increased. As routing methods to recover fast from a single network component (link or node) failure, methods using multiple routing tables have been proposed. In the methods, when a component failure occurs, packets which are supposed to pass through the component in the failure-free state are rerouted using a routing table (backup table) which does not use the component for packet forwarding. Thus, it is important to decrease the increased routing cost (for example, the number of increased hops of rerouted packets if every link cost is one) due to the reroute as much as possible. In this paper, we propose a new routing method to decrease the increased routing cost. In the conventional methods, there are many components that do not fail but are not used for packet forwarding in a backup table. In our proposed method, we use such components as much as possible, and consequently we try to decrease the increased routing cost. Numerical examples show the effectiveness of our proposed method.

DOI： 10.1109/ICTC.2014.6983329

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

When the standard TCP implementation is used in distributed. le systems in data center, a well-known throughput degradation called TCP Incast occurs. In order to avoid TCP Incast, a TCP implementation using a specialized. fine-grained timer is proposed. However, the throughput still decreases in the implementation when the number of servers in distributed. le systems is large. In this paper, we clarify that the cause of the decreasing is that maximum timeout value is too large and optimize the value.

DOI： 10.1109/ICTC.2014.6983174

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

In recent years, with proliferation of smart phones and tablet PCs, speedup of wireless LAN communication is required for dealing with increase of traffic in wireless networks. However, transmission speed through a wireless network often slows down in comparison with that through a wired network since packets of wireless networks frequently drop due to the influence of surrounding environment such as electromagnetic noise. In this paper, we propose a method to mitigate the impacts caused by packet loss by virtue of SCTP bicasting in lossy multihoming environment with two or more wireless networks. This method bicasts not all packets but only important packets concerning retransmission for efficiency since bicasting all packets would cause congestion. We also implemented a prototype system based on the proposed method. According to the result of performance evaluation experiment, we confirmed the effectiveness of the proposed method by the fact that the prototype system performed faster transmission than normal SCTP transmission even in high packet loss rate environment.

DOI： 10.1109/COMPSAC.2014.78

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

Split-connection type proxies have been proposed to improve TCP (Transmission Control Protocol) throughput in network environment with wireless LANs. A proxy is placed at the boundary of a wireless LAN (WLAN) and a wired network and it splits a TCP connection into two virtual TCP connections. We proposed a split-connection type proxy called MDA-Proxy (Modified Decreasing-ACK Proxy), which increases throughput of upflows (flows from wireless hosts to wired hosts) by decreasing the number of ACKs in a WLAN. MDA-Proxy uses two parameters, ack_stock and delay_timer, for each connection. The former controls the number of ACKs sent to a WLAN and the latter is used to recover from losses and delays of ACKs in a wired network. However, it is assumed that MDA-Proxy uses fixed values of the two parameters and their automatic tuning in response to network situation remains to be future work. In this paper, we propose a new type of proxy called ASA-Proxy (Ack_Stock Autotunning Proxy), which automatically tunes the two parameters in MDA-Proxy. Ack_stock is tuned comparing throughputs before and after its changing so that throughput of the connection becomes large, and delay_timer is replaced by the normal TCP retransmission timer because it is better than a fixed value of delay_timer for the inference of losses and delays of ACKs in a wired network. Extensive simulation runs show that ASA-Proxy is superior to MDA-Proxy.

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

The demands on Internet reliability and availability have increased. As a routing method to recover fast from a single network component (link or node) failure, a method using multiple routing tables has been proposed. In the method, when a component failure occurs, packets which are supposed to pass through the component in no failure state are rerouted using a routing table (backup table) which does not use the component for packet forwarding. Thus, it is important to decrease the increased routing cost (for example, the number of increased hops of rerouted packets if every link cost is one) due to the reroute as much as possible. In this paper, we propose a new routing method to decrease the increased routing cost. In the conventional method, there are such components that do not fail but are not used for packet forwarding in a backup table. In our proposed method, we use such components, and consequently we try to decrease the increased routing cost. Numerical examples show the effectiveness of our proposed method.

DOI： 10.1109/APCC.2013.6765984

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

In distributed file systems, a well-known congestion collapse called TCP Incast occurs because many servers send data to the same client and then many packets overflow the port buffer of the client link. Incast leads to throughput degradation in the network. In this paper, we propose two methods to avoid Incast based on the fact that only one connection can almost fully use the network bandwidth because the bandwidth-delay product is not so large in current data center networks. The first method is a method which completely serializes connection establishments. By the serialization, the number of packets in the port buffer becomes very small, which leads to Incast avoidance. The second method is a method which overlaps the slow start period of the next connection to the current established connection to improve throughput in the first method. Numerical results from extensive simulation runs show the effectiveness of our two proposed methods.

DOI： 10.1109/APCC.2013.6765931

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

In distributed file systems, a well-known congestion collapse called TCP Incast occurs because many servers send data to the same client and then many packets overflow the port buffer of the client link. Incast leads to throughput degradation in the network In our previous work, we have proposed a method to avoid Incast. In the method, we limit the maximum number of simultaneously existing connections to a predetermined constant value. However, we cannot use the method when the data size is small, and because we have not investigated how to optimize the maximum value, Incast may occur if the maximum value is not appropriate. In this paper, we improve the method so that it is applicable regardless of the data size and the maximum value is optimized Numerical results show the effectiveness of our proposed method.

DOI： 10.1109/ICTC.2013.6675396

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

Light trail architecture is attracting attention as a new optical wavelength-division multiplexing network architecture that can be built with currently available devices and can achieve bandwidth allocation with granularity finer than a wavelength. Because a light trail is a shared medium, we need a medium access control (MAC) protocol to prevent collisions. Although MAC protocols using token passing can prevent collisions, the bandwidths of links that are located upstream of the token holding node are kept idle. We first propose a dynamic light trail splitting method for increasing throughput of a light trail by using such idle bandwidths. Our method splits a trail into upstream and downstream trails at the token holding node, and independent data transmission on the two trails are permitted. As a result, we expect that the split trail architecture will achieve higher throughput than the original non-split trail architecture. The degree of performance improvement with the split trail architecture depends on how appropriately we determine the upstream and downstream token holding times of every transmission node. Thus, we formulate a problem in which we optimize the token holding times to accommodate requested traffic volume as a linear programming problem. We then derive the throughput of the split trail architecture by solving the problem using the NUOPT solver and investigate the degree of improvement over the original architecture. In addition, we evaluate the end-to-end delay of the split trail architecture by simulation. According to numerical examples, the split trail architecture achieves 1) almost the same throughput as the original one for the worst-case traffic pattern where every transmission node sends data to the terminating node of the trail only, 2) about 1.6 times higher throughput for a uniform traffic pattern where every node pair requests the same traffic volume and an extremely unbalanced traffic pattern where only a few node pairs request huge traffic volume, 3) about 1.9 time higher throughput for the split trail architecture's good-case traffic pattern where every transmission node sends data to its adjacent downstream node only, and 4) the end-to-end delay enough to satisfy any application's QoS requirement according to ITU-T Recommendation Y.1541.

DOI： 10.1587/transcom.E95.B.2330

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

To improve TCP throughput even if the maximum receiving window size is small, a TCP performance enhancing proxy (PEP) using a UDP-like packet sending policy with error control has been proposed. The PEP operates on a router along a TCP connection. When the PEP receives a data packet from the source host, it transmits the packet to the destination host, copies the packet into the local buffer (PEP buffer) in case the packets need to be transmitted and sends a premature ACK acknowledging receipt of the packet to the source host. In the PEP, the number of prematurely acknowledged packets in the PEP buffer is limited to a fixed threshold (watermark) value to avoid network congestion. Although the watermark value should be adjusted to changes in the network conditions, watermark adjusting algorithms have not been investigated. In this paper, we propose a watermark adjusting algorithm the goal of which is to maximize the throughput of each connection as much as possible without excessively suppressing the throughputs of the other connections. In our proposed algorithm, a newly established connection uses the initial watermark value of zero to avoid drastic network congestion and increases the value as long as its throughput increases. In addition, when a new connection is established, every already-established connection halves its watermark value to allow the newly established connection to use some portion of the bandwidth and increases again as long as its throughput increases. We compare the proposed algorithm (CW method) with other methods: the FW method that uses a fixed large watermark value and the NP method that does not use the PER Numerical results with respect to throughput and fairness showed that the CW method is generally superior to the other two methods.

DOI： 10.1587/transcom.E95.B.2344

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

In recent years, there is a big demand for live streaming service over the Internet. As one of the service model, peer-to-peer (P2P) live streaming has been utilized. In the system, it is important to decrease inter-AS (Autonomous System) traffic volume and content delivery delay because the larger volume leads to higher transit cost of ASes and this system treats the live contents, respectively. Therefore, as a conventional peer selection algorithm that tries to decrease them, Minimum Physical Hop (MPH) has been proposed. In MPH, to decrease inter-AS traffic volume, a newly joining peer selects a providing peer with the minimum physical hop count, which is the minimum number of ASes between the providing peer and the newly joining peer. Furthermore, to decrease content delivery delay, if there are several such candidates, it selects a providing peer with the minimum logical hop count, which is the number of Origin Streaming Servers (OSS) or peers passed from an OSS to the peer, from the candidates. However, MPH shows high inter-AS traffic volume when the number of joining peers is large. When we use MPH, candidates for providing peer with large logical hop count tend to swarm in ASes without an OSS while those with small logical hop count do in ASes with an OSS because of the peer selection policy of MPH. As the number of joining peers becomes large, the logical hop counts of the candidates for providing peer in every AS increase. As a result, in the former ASes, the logical hop counts of all the candidates for providing peer reach the upper bound frequently, and the newly joining peer cannot often find its providing peer in the AS, and consequently have to select providing peers in different ASes. In this paper, we improve MPH in order to further decrease the inter-AS traffic volume. In the improved algorithm called Improved Minimum Physical Hop (IMPH), we try to decrease the volume by distributing providing peers with one logical hop count to many ASes. As a result of the evaluation, we show that IMPH achieves to reduce the inter-AS traffic volume by at a maximum of 38% compared to MPH under uniform peer generation distributions.

DOI： 10.1109/ICNC.2012.31

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

TCP-fox is a new flow control method for TCP (Transmission Control Protocol) which allows the sender of a connection to send packets over the receive window advertised by the receiver to obtain higher throughput. However, TCP-fox has the problem that it halts sending packets when the receiver uses a receive buffer auto tuning mechanism. In this paper, we investigate the cause of the halting and present a way to avoid it.

DOI： 10.1109/APCC.2012.6388193

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）  

In a network environment including a wireless LAN and a wired network, split-connection type proxies have been proposed to improve TCP (Transmission Control Protocol) throughput. A proxy is placed at the boundary between a wireless LAN and a wired network, and it splits a TCP connection which is established between a host (WSTA) in a wireless LAN and a host (STA) in a wired network into two virtual TCP connections. We have proposed a split-connection proxy called DA-Proxy, and simulation results in our previous works show that although the DA-Proxy can increase throughput of downflows (flows from STAs to WSTAs), throughput of upflows (flows from WSTAs to STAs) is not improved and is almost the same as that of upflows without the proxy. In this paper, we propose a new proxy called MDA-Proxy which tries to increase throughput of upflows by decreasing the number of ACKs in a WLAN and show its effectiveness using simulation results. © 2012 IEEE.

DOI： 10.1109/GCCE.2012.6379960

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）  

Using network virtualization technologies for network applications (NW-Apps) consisting of server and clients, we can implement a server migration service where there are many server running environments (working places: WPs) inside a network and servers onWPs can migrate to otherWPs when communication QoSs in some NW-Apps fall off. Because server size is large, the traffic for server migration causes QoS degradation of its background traffic. Thus, it is important to decrease the degree (network impact) of the QoS degradation. In this paper, we propose three destination selection algorithms, MIA (Minimum Impact Algorithm), MRA (Maximum Remaining space Algorithm) and MCA (Maximum Covering Algorithm) that try to decrease the impact while increasing the number of NW-Apps' clients whose SLA (service level agreement) are satisfied. When a server migration is triggered, MIA, MRA and MCA move the server in the WP with the minimum impact, the WP with the maximum remaining space for servers and the WP with the maximum coverage, which is the number of routers under which the corresponding clients can communicate with the server while satisfying the SLA, respectively. Numerical examples show that when the number of accommodatable servers in full-cover WP (a full-cover WP is such WP that communicates with an arbitrary client while satisfying the SLA) is small or there is no full-cover WP, MIA outperforms the others, otherwise MCA outperforms the others. Copyright © 2012 by the Association for Computing Machinery, Inc.

DOI： 10.1145/2377310.2377315

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：ELSEVIER SCIENCE BV  

Change-point detection schemes, which represent one type of anomaly detection schemes, are a promising approach for detecting network anomalies, such as attacks and epidemics by unknown viruses and worms. These events are detected as change-points. However, the schemes generally also detect false-positive change-points caused by other events, such as improper parameter setting of detectors. Therefore there is a requirement for a scheme that detects only true-positive change-points caused by attacks and epidemics by unknown viruses and worms. The true-positive change-points tend to occur simultaneously and intensively in very large numbers, while the false-positive change-points tend to occur independently. Therefore, we expect that the multi-stage change-point detection scheme, which performs change-point detection in a distributed manner and takes account of the correlation among multiple change-points, can exclude false-positive change-points by neglecting those that occur independently. In this paper, we propose the multi-stage change-point detection scheme and introduce a weighting function that gives smaller weight to LDs with higher false-positive rate inferred by GD in order to avoid a set of false-positive alerts generated by the low-accuracy detectors from causing high false-positive rate of the scheme. We evaluate the performance of the scheme by a simulation using the parameter values obtained in an experiment using real random scan worms. In the evaluation, we modify AAWP (Analytical Active Worm Propagation) model so that it can derive the number of infected hosts (i.e., attack hosts) more accurately by considering a failure of infection behavior by random scan worms. The simulation results show that our scheme can achieve an optimal performance (detection rate of 1.0 and false-positive rate of 0) while the stand-alone change-point detection scheme, which does not use the correlation among multiple change-points, cannot attain such optimal performance, and our scheme with alert weighting always shows better detection performance than the scheme without alert weighting. (C) 2011 Elsevier B.V. All rights reserved.

DOI： 10.1016/j.comcom.2011.05.001

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

As a new optical WDM network architecture that can be built with currently available devices and can achieve bandwidth allocation with granularity finer than a wavelength, a light trail architecture attracts attention. Because a light trail is a shared medium, we need a medium access control (MAC) protocol to avoid collisions. Although MAC protocols using token passing can avoid collisions, bandwidths of links that locate upstream of the token holding node are kept idle. In this paper, we first propose a dynamic light trail splitting method in order to increase throughput of a light trail by using those idle bandwidths. Our method splits a trail into the upstream trail and the downstream trail at the token holding node and independent data transmissions on the two trails are permitted. As a result, we expect that the split trail architecture achieves higher maximum throughput than the original non-split trail architecture. The degree of throughput improvement by the split trail architecture depends on how appropriately we set upstream and downstream token holding times of every transmission node. Thus, we formulate a problem to decide the token holding times as a nonlinear programming problem, derive the maximum throughput of the split trail architecture by solving the problem using NUOPT solver, and investigate the degree of improvement compared to the original architecture. According to numerical examples, the split trail architecture achieves 1) almost the same maximum throughput as the original one for its unfavorite traffic pattern where every transmission node sends data to the terminating node of the trail only, 2) about 1.6 times as high maximum throughput for a uniform traffic pattern where every node-pair requests the same traffic volume, and 3) about 1.9 time as high maximum throughput for its favorite traffic pattern where every transmission node sends data to its adjacent downstream node only.

DOI： 10.1109/GLOCOM.2011.6133810

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）  

Fixed robust routing is a promising approach for ISP networks to accommodate variable traffic patterns with low operational complexity. The routing minimizes the maximum link load by distributing traffic of every source-destination pair to multiple paths (multipath routing). The multipath routing can result in performance degradation of TCP because of frequent out-of-order packet arrivals. In this paper, we first investigate the influences of delay differences among multiple paths and ratio of shorter paths among multiple paths on TCP performance with simulation using ns-2. The simulation results clarify that smaller delay difference and lower ratio of shorter paths among multiple paths lead to higher TCP throughput. Based on the investigation results, we next propose fixed robust routing algorithms that try to improve TCP throughput in addition to decreasing the maximum link load. The first algorithm called MDD (Minimum Delay Difference) selects a set of paths with the minimum delay differences between the shortest and the longest paths as candidate paths for every source-destination pair The second algorithm called MDD-LF (Minimum Delay Difference with Limited Fraction) bounds the fraction of traffic routed on the shortest delay path in addition to selecting the same candidate paths as MDD. Simulations using ns-2 show that, compared to a straightforward fixed robust routing that selects k-shortest hop paths as the candidate paths, MDD and MDD-LF achieve about 22% and 27% higher TCP throughput while MDD and MDD-LF produce about 1.7 and 2.3 times higher maximum link load, respectively. © 2011 IEEE.

DOI： 10.1109/CQR.2011.5996088

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

With the spread of broadband access lines, many bandwidth-consuming services such as video streaming have appeared. The popularity of these services can cause problems such as the shortage of the Internet backbone capacity, so it is important to detect increasing traffic caused by these services early. Conventional detection method called aggregation method tries to detect increasing traffic by predicting its future traffic volume with linear approximation. However, the aggregation method may not cope with increasing traffic whose growth is more rapid than linear growth. In this paper, we propose an early detection method (partial aggregation method) of increasing traffic under per-subnet based distributed traffic measurement. The method predicts 1) future traffic volume for each address in each subnet and 2) the number of subnets having increasing traffic in the future with a linear approximation. Then, the method estimates future traffic volume for each address as a product of the predicted future traffic volume in each subnet and the predicted subnet number. As a result, the method is expected to cope with rapid growth in traffic volume. Simulation results show that the partial aggregation method can detect increasing traffic earlier than the aggregation method by a maximum of 90 days.

DOI： 10.1109/TENCON.2010.5686580

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

As a detection method of large-scale simultaneous events (e.g., DDoS attack), a multi-stage change-point detection scheme with alert weighting was proposed. In the scheme, local detectors (LDs), which are deployed on each monitored subnet, try to detect an event by change-point detection. If they do, they send an alert to global detector (GD). Then GD judges whether an event is occurring by comparing the weight sum of the received alerts with an predetermined threshold value. The weight of an alert is set lower for LDs with higher false-positive rate (FPR). Conventional evaluation results only showed that alert weighting improves the performance for particular combination of two kinds of LDs with different FPRs. In this paper, we investigate the effectiveness of alert weighting for various combinations of two kinds of LDs with different FPRs in detail. We first consider the situation where detection rates (DRs) of all LDs are identical. Then, we consider the situation where high-FPR LDs show higher DR than low-FPR LDs, which is more realistic. Simulation results show that 1) alert weighting does not lead to degradation of detection performance and 2) alert weighting is most effective when event scale is moderate in our numerical examples.

DOI： 10.1109/TENCON.2010.5686586

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

One promising approach for large-scale simultaneous events (e.g., DDoS attacks and worm epidemics) is to use a multi-stage change-point detection scheme. The scheme adopts two-stage detection. In the first stage, local detectors (LDs), which are deployed on each monitored subnet, detects a change point in a monitored metric such as outgoing traffic rate. If an LD detects a change-point, it sends an alert to global detector (GD). In the second stage, GD checks whether the proportion of LDs that send alerts simultaneously is greater than or equal to a threshold value. If so, it judges that large-scale simultaneous events are occurring. In previous studies for the multi-stage change-point detection scheme, it is assumed that weight of each alert is identical. Under this assumption, false-positive rate of the scheme tends to be high when some LDs sends false-positive alerts frequently. In this paper, we weight alerts based on false-positive rate of each LD in order to decrease false-positive rate of the multi-stage change-point detection scheme. In our scheme, GD infers false-positive rate of each LD and gives lower weight to LDs with higher false-positive rate. Simulation results show that our proposed scheme can achieve lower false-positive rate than the scheme without alert weighting under the constraint that detection rate must be 1.0.

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：ELECTRONICS TELECOMMUNICATIONS RESEARCH INST  

We evaluate performance of two medium access control methods with token passing, LT-FA MAC and ART MAC, in light trail networks. LT-FA MAC aims at coping with bursty packet arrivals while ART MAC aims at decreasing transmission delay. Simulation evaluation is performed in terms of the number of the maximum transmission nodes accommodated on a light trail and the maximum communication distance under the constraints that the bandwidth utilization rate of 90% and the average transmission delay of 100 ms must be achieved. Simulation results show that ART MAC permits about 500-1500km longer communication distance than LT-FA MAC with the same number of transmission nodes on a light trail and ART MAC accommodates about 15-25 more nodes than LT-FA MAC with the same communication distance.

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）  

In this paper, we evaluate the performance of peer selection algorithms: MLH (Minimum Logical Hop) and MPH (Minimum Physical Hop) to increase the number of joining peers and to decrease inter-AS traffic volume in P2P live streaming, where we assume that every peer's logical hop count (the number of providing peers between an origin streaming server and the peer) is limited by a predetermined value in order to keep real-time property of live streaming. In MLH, a newly joining peer selects such providing peers that the logical hop count is minimum to increase the number of joining peers. And then if there are several such providing peers, the newly joining peer selects such providing peers that the number (physical hop count) of ASs between the newly joining peer and each of the providing peers is minimum to decrease the inter-AS traffic volume. In MPH, a newly joining peer selects providing peers in the reverse order of MLH. Simulation results show that MLH achieves about from 20% to 60% more maximum joining peers than MPH, and MPH shows smaller inter-AS traffic volume than MLH when the number of joining peers is small while MLH shows smaller inter-AS traffic volume when the number of joining peers is large. ©2009 IEEE.

DOI： 10.1109/APCC.2009.5375467

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記述言語：英語   掲載種別：研究論文（学術雑誌）  

Although the offset-time-based scheme can differentiate loss rates of packets according to their QoS levels in optical burst switching networks, it also increases end-to-end delay of packets with higher QoS level. This is because it increases offset time of bursts in proportional to QoS levels of packets which are included in the bursts. In this paper, we propose a burst assembly method that reduces end-to-end delay. Our proposed method includes packets that arrive during the offset time into the currently assembled bursts while the conventional method includes them into the burst assembled next time. Simulation results show that our method offers 12-23% delay reduction for packets with the highest QoS level.

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

In order to improve TCP performance in networks with large round trip time, a method using a PEP (Performance Enhancing Proxy) is proposed. In the method, the PEP is located on a intermediate router along a TCP connection, and returns acknowledgment packets (premature ACKs) to the source host instead of the destination host. In previous researches, although a congestion control in the PEP which keeps the number of prematurely acknowledged packets below a threshold (watermark) value has been proposed, packet losses are not taken into account, and consequently performance of the PEP under a packet-loss environment has not been investigated. In this paper, we incorporate a retransmission control to deal with packet losses into the congestion control. The proposed retransmission control mainly has two functions. One is to return duplicate premature ACKs or partial premature ACKs to urge the source host to execute the fast retransmission. The other is to retransmit prematurely acknowledged packets to the destination host, when a timeout occurs or three duplicate acknowledgments and a partial acknowledgment from the destination host are received. Simulation results under a packet-loss environment show that the PEP improves throughput larger as the end-to-end propagation delay becomes larger and throughput with PEP is a maximum of four times as much as throughput without PEP.

DOI： 10.1109/ICACT.2008.4494152

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

Change-point detection schemes, which represent one type of anomaly detection schemes, are a promising approach for detecting network anomalies, such as attacks and epidemics by unknown viruses and worms. These events are detected as change-points. However, they generally also detect false-positive change-points caused by other events, such as hardware problems. Therefore there is a requirement for a scheme that detects only true-positive change-points caused by attacks and epidemics by unknown viruses and worms. The true-positive change-points tend to occur simultaneously and intensively in very large numbers, while the false-positive change-points tend to occur independently. We can exclude false-positive change-points by excluding those that occur independently, based on information gathered from the entire network. In this paper, we combine change-point detection schemes with a distributed IDS, and evaluate performance of the combined scheme by a simulation using the parameter values obtained by an experiment using real worms. The simulation results show that the combined scheme detects all the DDoS attacks without any false-positives while we have to tolerate false-positive rate of at least 0.02 to detect all the attacks in a stand-alone IDS scheme.

DOI： 10.1109/APSITT.2008.4653540

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記述言語：英語   掲載種別：研究論文（学術雑誌）  

In wavelength division multiplexing (WDM) technology, high-speed large-capacity communication is realized by transmitting multiple optical signals with different wavelengths in parallel on an optical fiber. In WDM technology, the logical topology of the network can generally be modified by changing the wavelength assignment to the transmitters and receivers. Thus it is important to develop an algorithm that derives the wavelength assignment that maximizes the transmission efficiency of the network. This paper considers a bidirectional Manhattan street network which has a regular topology, and proposes a hierarchical two-stage heuristic algorithm. In the first stage, the initial solution is derived by hierarchical region division and node assignment to the regions. In the second stage, the solution is improved by simulated annealing (SA). The proposed algorithm is evaluated by simulations of random instances and instances for which optimal solutions are already known. The effectiveness of the proposed algorithm is demonstrated through comparisons to the existing one. © 2007 Wiley Periodicals, Inc.

DOI： 10.1002/scj.10534

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

A method to detect open node defects that cannot be detected by the conventional IDDQ test method has previously been proposed employing a sinusoidal wave superposed on the DC supply voltage. The present paper proposes a strategy to improve the detectability of the test method by means of frequency analysis of the supply current. In this strategy. defects are detected by determining whether secondary harmonics of the sinusoidal wave exist in the supply current. The effectiveness of the method is confirmed by experiments on two CMOS NAND gate packages (SSIs).

DOI： 10.1093/ietisy/e90-d.3.685

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

In order to improve TCP performance, the use of a PEP (Performance Enhancing Proxy) has been proposed. The PEP operates on a router along a TCP connection. When a data packet arrives at the PEP, it forwards the packet to the destination host, transmits the corresponding ACK (premature ACK) to the source host on behalf of the destination host, and stores a copy of the packet in a local buffer (PEP buffer) in case the packet needs to be retransmitted. In this paper, in accordance with a strategy that keeps the number of prematurely acknowledged packets in the PEP buffer below a fixed threshold (watermark) value, we investigate the relation between the watermark value and the average throughput. Extensive simulations show that the results can be roughly classified into two cases. In the first case, the average throughput becomes larger for larger watermark values and becomes a constant value when the watermark value is over a certain value. In the second case, although the average throughput becomes larger for lager watermark value in the same way, it decreases when the watermark value is over a certain value. We also show that the latter (former) case can occur more easily as the propagation delay in the input side network of the PEP becomes smaller (larger) and the propagation delay in the output side network of the PEP becomes larger (smaller), and also show that the latter (former) case can occur more easily as the transmission speed in the input side network becomes larger (smaller) and the transmission speed in the output side network becomes smaller (larger) while the PEP buffer capacity becomes smaller (larger).

DOI： 10.1093/ietcom/e90-b.1.31

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）  

Change-point detection schemes are a promising approach for detecting network anomalies, such as attacks and epidemics by unknown viruses and worms. They detect those events as change-points. However, they generally also detect false-positive change-points, those caused by other events such as hardware trouble. A scheme is needed that only detects truepositive change-points, caused by attacks and epidemics. Truepositive change-points tend to occur simultaneously in very large numbers, while false-positive change-points tend to occur sporadically. We can exclude false-positive change-points by excluding change-points that occur sporadically, based on information gathered from the entire network. In this paper, we propose a multi-stage network event detection scheme that aggregates change-point information from distributed IDSs (Intrusion Detection Systems) and detects the true-positive change-points. Simulation results show that, compared to a scheme using only one IDS, our method always yields a smaller false-positive rate under the constraint that the detection rate of the true-positive change-points must exceed 0.99. © 2007 IEEE.

DOI： 10.1109/ICN.2007.71

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

The EDD connection admission control scheme has been proposed for supporting real-time communication in packet-switched networks. In the scheme, when a connection establishment request occurs, the worst-case link delay in each link along the connection is calculated to determine whether the request can be accepted or not. In order to calculate the worst-case link delay, we must perform a check called the point schedulability check for each of some discrete time instants (checkpoints). Therefore when there are many checkpoints, the worst-case link delay calculation is time-consuming. We have proposed a high-speed calculation method. The method finds some checkpoints for which the point schedulability check need not be performed and removes such unnecessary checkpoints in advance before a connection establishment request occurs, and the check is performed for each of the remaining checkpoints after the request occurs. However, the method is not so effective under the situation that the maximum packet length in networks is large, because the method can find few unnecessary checkpoints under the situation. This paper proposes a new high-speed calculation method. We relax the condition which determines whether or not the point schedulability check need not be performed for each checkpoint in our previous method and derive a new condition for finding unnecessary checkpoints. Using the proposed method based on the new condition, we can increase the number of unnecessary checkpoints compared to our previous method. Numerical examples which are obtained by extensive simulation show that the proposed method can attain as much as about 50 times speedup.

DOI： 10.1093/ietcom/e89-b.7.2012

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

Although the offset-time-based scheme can differentiate packet/burst loss rate according to their QoS levels in optical burst switching networks, it increases end-to-end packet delays of packets in a burst with higher QoS level. This is because it increases offset time of bursts in proportional to their QoS levels. In this paper, we propose a burst assembly method that decreases end-to-end packet delays. Our proposed method includes packets that arrive during the offset time into the currently assembled bursts while the conventional method includes them into the burst assembled next time. Simulation results show that our method offers more than 21% delay reduction for bursts with the highest QoS level.

DOI： 10.1109/ICCT.2006.341915

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

To improve TCP throughput performance, a method using a PEP (Performance Enhancing Proxy) has been proposed. The PEP operates on a router along a TCP connection. When a data packet arrives at the PEP, it forwards the packet to the destination host, transmits the corresponding ACK (premature ACK) to the source host in behalf of the destination host, and stores a copy of the packet into its own buffer (PEP buffer) in case retransmission of the packet is required. As a congestion control method on the PEP, a method which keeps the number of prematurely acknowledged packets in the PEP buffer below a threshold (watermark) value has been proposed. However, the relation between the watermark value and throughput is not sufficiently investigated, and an optimization method of the watermark value is not proposed. In this paper, we first investigate the relation between the watermark value and the average throughput. Extensive simulations show that the simulation results are roughly classified into two cases. In the first case, the average throughput becomes larger for larger watermark values and becomes a constant value when the watermark is over a certain value. In the second case, although the average throughput becomes larger for larger watermark values in the same way, it decreases when the watermark is over a certain value. Next, based on the results about the relation, we propose an watermark optimization algorithm which can adaptively maximize the average throughput of each connection and also satisfy a fairness condition that the average throughputs of connections are equal to each other.

DOI： 10.1109/ICCT.2006.341898

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

The shuffle-like network (SL-Net) is known as a logical topology for WDM-based multihop packet-switched networks. Even if we fix the logical topology to an SL-Net, we can still reposition nodes in the SL-Net by re-tuning wavelengths of transmitters and/or receivers. In conventional node placement algorithms, routes between nodes are assumed to be given. In this paper, we propose two heuristic node placement algorithms for the SL-Net to decrease the average end-to-end packet transmission delay under a given traffic matrix in the case that routes are design variables. The principal idea is to prevent too many traffic flows from overlapping on any link. To attain the idea, in one of the algorithms, a node is selected one by one in a decreasing order of the sums of sending and receiving traffic requirements in nodes, and its placement and routes between the node and all the nodes already placed are simultaneously decided so that the maximum of the amounts of traffic on links at the moment is minimum. In the other algorithm, a node is selected in the same way, and first it is placed so that the average distance between the node and all the nodes already placed is as large as possible, and then routes between the node and all the nodes already placed are decided so that the maximum of the amounts of traffic on links at the moment is minimum. Numerical results for four typical traffic matrices show that either of the proposed algorithms has better performance than conventional algorithms for each matrix, and show that the proposed algorithms, which are based on a jointed optimization approach of node placement and routing, are superior to algorithms which execute node placement and routing as two isolated phases.

DOI： 10.1093/ietcom/e88-b.12.4578

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記述言語：英語   掲載種別：研究論文（学術雑誌）  

The shuffle-like network (SL-Net) is known as a logical topology for WDM-based multihop packet-switched networks. We propose two heuristic node placement algorithms for the SL-Net to decrease the average end-to-end packet transmission delay for a given traffic matrix. The principal idea is to prevent too many traffic flows from overlapping on any link. To attain this objective, one algorithm selects nodes one by one in decreasing order of the sums of their sending and receiving traffic requirements, and places them so that the maximum traffic in the links is minimal. The other selects nodes in the same way, and places then so that the average distance among the nodes which have already been placed is as large as possible. Numerical results for four typical traffic matrices show that either of the proposed algorithms has better performance than the conventional algorithms for each matrix. © 2004 Wiley Periodicals, Inc.

DOI： 10.1002/ecja.10184

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

In order to improve TCP performance, a method using a PEP (Performance Enhancing Proxy) is proposed. The PEP operates on a router along a TCP connection. When a data packet arrives at the PEP, it forwards the packet to the destination host, transmits the corresponding ACK (premature ACK) to the source host in behalf of the destination host and stores the copy of the packet into its own buffer (PEP buffer) in case of the retransmission of the packet. In this paper, under the strategy which keeps the number of packets in the PEP buffer for which premature ACKs have been returned being less than or equal to a fixed threshold value (watermark value), we investigate the relation between the watermark value and the maximum throughput. Extensive simulation runs show that the simulation results are roughly classified into two cases. One case is that the maximum throughput becomes larger for larger watermark value and becomes a constant value when the watermark value is over a value. The other case is that though the maximum throughput becomes larger for lager watermark value in the same way, it reversely decreases when the watermark value is over a value. We also show that the latter (former) case is easier to occur as the propagation delay in the input side network of the PEP becomes smaller (larger) and the propagation delay in the output side network of the PEP becomes larger (smaller) and the PEP buffer capacity becomes smaller (larger).

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：SCRIPTA TECHNICA-JOHN WILEY & SONS  

In this paper, the authors propose a multiwindow system that can smoothly execute window operations such as moving, resizing, or scrolling by using a combination of a drawing-time composition method and display-time composition method. Memory for storing the image of the window for which the window operation is to be performed (operation target window) and memory for storing a composite image of all other windows generated by a drawing-time composition method are both established in this system. The image that is displayed on the actual screen is obtained by using a display-time composition method to combine images that were read in parallel from both of these memories to generate the multiwindow image. The authors performed a trial experiment using a 640-pixel by 480-pixel display device to verify that no flicker appeared on the display screen when updating the operation target window. From the results of this trial experiment, they suggested that smooth window operations probably can also be executed for a 1280-pixel by 1024-pixel display device. (C) 2004 Wiley Periodicals, Inc.

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE  

As one of mobile agent applications, many systems which provide continuous service for users moving on a network have been proposed. In these systems, because a movement of mobile agents is performed after a user movement, users must wait for arrival of mobile agents. To reduce users' waiting time, we propose a fundamental framework for mobile agent systems where an agent can move precedently before a user movement. In our framework, it is assumed that computers are connected on a network and users with rewritable devices move on the network. The framework supports precedent movement of mobile agents based on prediction using movement history of users. Because the prediction may be wrong, the framework also provides the following movement of mobile agents. Moreover, the framework provides a recovery method of mobile agents in service in case that mobile agents disappear due to problems such as their bugs. Because we provide some APIs, via which various functions of our framework are accessed, developers of mobile agent systems can easily use our framework using the APIs. We implemented an experimental agent system using the APIs and confirmed that the framework performed correctly using the experimental system.

DOI： 10.1109/APSITT.2005.203636

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

This paper proposes a new supply current test method for detecting floating gate defects in CMOS ICs. In the method, unusual increase of the supply current caused by defects is promoted by superposing an AC component on the DC power supply. Feasibility of the test is examined by some experiments on four DUTs with an intentionally caused defect. The results showed that our method could detect clearly all the defects, one of which may be detected by neither any functional logic test nor any conventional supply current test.

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE COMPUTER SOC  

We already proposed a supply current test method for detecting floating gate defects in CMOS ICs. In the method, increase of the supply current caused by defects is promoted by superposing a sinusoidal signal on the supply voltage. In this study, we propose one way to improve detectability of the method for the defects. They are detected by analyzing the frequency of supply current and judging whether secondary harmonics of the sinusoidal signal exists or not. Effectiveness of our way is confirmed by some experiments.

DOI： 10.1109/ATS.2003.1250846

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

For efficient use of limited electromagnetic wave resource, the assignment of communication channels to call requests is very important in a cellular network. This task has been formulated as an NP-hard combinatorial optimization problem named the channel assignment problem (CAP). Given a cellular network and a set of call requests, CAP requires to find a channel assignment to the call requests such that three types of interference constraints between channels are not only satisfied, but also the number of channels (channel span) is minimized. This paper presents an iterative search approximation algorithm for CAP, called the Quasi-solution state evolution algorithm for CAP (QCAP). To solve hard CAP instances in reasonable time, QCAP evolutes quasi-solution states where a subset of call requests are assigned channels and no more request can be satisfied without violating the constraint QCAP is composed of three stages. The first stage computes the lower bound on the channel span for a given instance. After the second stage greedily generates an initial quasi-solution state, the third stage evolutes them for a feasible channel assignment by iteratively generating best neighborhoods, with help of the dynamic state jump and the gradual span expansion for global convergence. The performance of QCAP is evaluated through solving benchmark instances in literature, where QCAP always finds the optimum or near-optimum solution in very short time. Our simulation results confirm the extensive search capability and the efficiency of QCAP.

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：IEEE COMPUTER SOC  

In this paper, we propose a new I-DDQ test method for detecting floating gate defects in CMOS ICs. In the method, unusual increase of the supply current caused by defects is promoted by superposing an AC component on the DC power supply. Feasibility of the test is examined by some experiments on four DUTs with an intentionally caused defect. The results showed that our method could detect clearly all the defects, one of which may be detected by neither any functional logic test nor any conventional IDDQ test.

DOI： 10.1109/ATS.2002.1181747

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記述言語：英語   掲載種別：研究論文（学術雑誌）  

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：Institute of Electronics, Information and Communication, Engineers, IEICE  

The programming circuit of SRAM-based FPGAs consists of two shift registers, a control circuit and a configuration memory (SRAM) cell array. Because the configuration memory cell array can be easily tested by conventional test methods for RAMs, we focus on testing for the shift registers. We first derive test procedures for the shift registers, which can be done by using only the faculties of the programming circuit, without using additional hardware. Next, we show the validness of the test procedures. Finally, we show an application of the test procedures to test Xilinx XC-1025.

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：John Wiley and Sons Inc.  

This paper describes the realization of the multi-window system, where the framed nonrectangular windows can be operated with a high speed. The body image (the image to be displayed inside the frame) and the frame image (the image to be displayed as the frame), which are stored on the multiport memory, are read out in parallel, being synchronized to the CRT scanning. By selecting pixelwise one of those images, the window image displayed in the window is generated as a framed image. The selection signal for this purpose is generated based on the window shape information stored in the dedicated memory. The multiwindow image is generated by selecting the image to be displayed for each pixel position on CRT using a hardware. A distributed priority encoder is used as the hardware for this purpose. According to the result of construction and experiment, it is expected that the multiwindow system can realize a sufficiently high-speed operation from the practical viewpoint for the 1200 × 1000 CRT screen using the existing circuit technology.

DOI： 10.1002/scj.4690271005

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：SCRIPTA TECHNICA-JOHN WILEY & SONS  

This paper proposes a design algorithm for packet-switching networks with different classes of packets under end-to-end packet delay constraint and a design algorithm for fault-tolerant networks. The former is a heuristic capacity, priority, and flow assignment algorithm which locally minimizes the network cost while satisfying a constraint that the average transmission delay of each class of packets between two arbitrary nodes never exceed a value predetermined for the class.
This algorithm is composed of three subalgorithms and the cost minimization is pursued by iterating these algorithms as long as the constraint is satisfied. The latter is a design algorithm for a fault-tolerant network which satisfies the foregoing constraint even if an arbitrary single link fails. With this algorithm, an assignment with relatively large cost but satisfying the constraint is obtained first. Then a procedure of reducing the capacity of a link whose effect on cost reduction is great but whose effect on path delay increase is not so great is iterated to reduce the network cost as long as the constraint is satisfied.

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掲載種別：研究論文（学術雑誌）  

An end‐to‐end error recovery scheme is a promising way to obtain low end‐to‐end delays in packet switching networks. However, a too‐short timeout interval may cause unnecessary packet retransmissions even when a packet is transmitted successfully to the receiver. On the other hand, a longer timeout interval may introduce large end‐to‐end delay when a packet is actually lost. Thus, it becomes important to determine an appropriate timeout period in the scheme. This paper analyzes a tandem queueing network system with blocking due to finite capacity in intermediate nodes and investigates the effect of the timeout period on the end‐to‐end delay performance. It is found that there exists the timeout period minimizing the mean packet transmission delay in the end‐to‐end error recovery scheme. The delay performance between end‐to‐end and link‐by‐link error recovery schemes also is compared and it is found that the delay performances of these schemes are influenced significantly by the node‐to‐node delay. Copyright © 1992 Wiley Periodicals, Inc., A Wiley Company

DOI： 10.1002/ecja.4410750702

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記述言語：英語   掲載種別：研究論文（学術雑誌）  

DOI： 10.1109/24.93767

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掲載種別：研究論文（学術雑誌）  

This paper proposes a queueing network model for the layered communication protocol and its approximate analysis method. The performance of the layered communication protocol is evaluated by using the proposed model and analysis. First, the queueing network model is constructed for the connection‐oriented and connectionless data transmissions between peer layers. In each queueing network model, the transmission delay in the lower layer protocol is represented by an infinite‐servor queue. The performance model for the layered communication protocol is constructed by piling up each model. Then an approximate analysis is presented for each queueing network model. Based on the relation to be satisfied by the packet transmission delay in each layer and the throughput between adjacent layers, an approximate analysis of the whole model can be executed iteratively. In numerical examples, the validity of the approximate analysis for the model is shown. The effects of window size, the time‐out period, and receiver buffer size on the transmission delay are discussed. Copyright © 1991 Wiley Periodicals, Inc., A Wiley Company

DOI： 10.1002/ecja.4410740402

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記述言語：英語   掲載種別：研究論文（学術雑誌）  

This paper contains the realization of the decentralized priority encoder (DPE) and presents two application examples. DPE is constructed by placing a module for each device issuing a request and by connecting the outputs from the modules using wired‐OR. Each module has the function to modify the priority dynamically. The request signal from the corresponding device is connected to the input of the module. When the priority of the module M issuing the request is higher than those of any other modules issuing requests, M produces the priority of its request as a binary code on the bus. A data selector can be constructed using DPE, since the data with the highest priority can be selected from a set of data with priorities. This data selector is suited especially to the data selector for image data in the enhanced multiwindow system. Using DPE as the priority comparator, a decentralized independent‐request arbiter with priority is obtained. Copyright © 1991 Wiley Periodicals, Inc., A Wiley Company

DOI： 10.1002/scj.4690221002

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記述言語：英語   掲載種別：研究論文（学術雑誌）  

In this paper, a link capacity assignment algorithm for a packet switched communication network is presented. This algorithm, called max‐delay‐link algorithm, prevents sharp performance degradation in a network even in case of link failures. An optimal link capacity assignment can be obtained with respect to link cost minimization, subject to the constraint that the average packet network transmission delay does not exceed a predetermined value. Robust networks against failures can be designed by assigning large capacities to links where overflow traffic and packet delay are expected to increase in case of a failure. Several numerical examples also are shown to examine the effectiveness of the algorithm in comparison with the conventional capacity assignment algorithm where link failures are not taken into account. Copyright © 1991 Wiley Periodicals, Inc., A Wiley Company

DOI： 10.1002/ecja.4410740902

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）   出版者・発行元：I E E E, COMPUTER SOC PRESS  

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記述言語：英語   掲載種別：研究論文（学術雑誌）  

The authors propose a queueing network model to examine the effects of the layered communication protocol implementation on the end-to-end delay. The main features of this model are as follows: based on the communication cycle, that is, data packet transmission → wait for ACK (→ timeout → retransmission) → receipt of ACK, each layer is modeled individually
and the case where one processor serves more than one layer process is considered.

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記述言語：英語   掲載種別：研究論文（学術雑誌）  

This paper compares the stability of five routing-procedures: nonalternate-routing (NAR), alternate-routing (AR), alternate-routing-trunk-reservation (AR-TR), alternate-routing-external-blocking (AR-EB), alternate-routing-trunk-reservation-and-external-blocking (AR-TR-EB), in circuit-switched network. First the stability of AR-TR, AR-EB AND AR-TR-EB routing-procedures is discussed by applying the cusp catastrophe theory, and the cusp is found in which a sudden change of end-to-end-blocking-probability (EEBP) occurs. Next, the stability of five routing-procedures is compared using the cusp. Finally, two methods are proposed that change the routing-procedure dynamically when network-traffic becomes large, and their effectiveness is shown.

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：ELSEVIER SCIENCE BV  

DOI： 10.1016/0169-7552(86)90059-0

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記述言語：日本語   出版者・発行元：電子情報通信学会  

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記述言語：日本語   出版者・発行元：電子情報通信学会  

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

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記述言語：英語  

We have developed a system (traditional system) to flexibly provide the requested applications environment on educational Windows PCs. The traditional system dynamically controls the execution of applications installed on each educational PC depending on the rules defined by teachers as well as by administrators. The traditional system, however, has a low tolerance for malicious attacks. If the execution file of a certain application is falsified, the corresponding rules already applied become invalid. In addition, though the traditional system has a function to define groups of controlled applications, it does not support hierarchical groups. This reduces the usability of the traditional system. In order to address these issues, this paper proposes a control method of application execution using digital certificates. The proposed method has a high tolerance for the falsification of execution files by controlling their executions based on the reliability of the corresponding digital certificates. It also improves its usability by introducing hierarchical group management utilizing hierarchical structure for digital certificates.------------------------------This is a preprint of an article intended for publication Journal ofInformation Processing(JIP). This preprint should not be cited. Thisarticle should be cited as: Journal of Information Processing Vol.23(2015) No.4 (online)------------------------------We have developed a system (traditional system) to flexibly provide the requested applications environment on educational Windows PCs. The traditional system dynamically controls the execution of applications installed on each educational PC depending on the rules defined by teachers as well as by administrators. The traditional system, however, has a low tolerance for malicious attacks. If the execution file of a certain application is falsified, the corresponding rules already applied become invalid. In addition, though the traditional system has a function to define groups of controlled applications, it does not support hierarchical groups. This reduces the usability of the traditional system. In order to address these issues, this paper proposes a control method of application execution using digital certificates. The proposed method has a high tolerance for the falsification of execution files by controlling their executions based on the reliability of the corresponding digital certificates. It also improves its usability by introducing hierarchical group management utilizing hierarchical structure for digital certificates.------------------------------This is a preprint of an article intended for publication Journal ofInformation Processing(JIP). This preprint should not be cited. Thisarticle should be cited as: Journal of Information Processing Vol.23(2015) No.4 (online)------------------------------

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記述言語：日本語   出版者・発行元：一般社団法人情報処理学会  

我々は先行研究として，アプリケーションの実行可否を制御することにより，教育用 PC における個別のアプリケーション環境を提供するシステム （従来システム） を開発してきた．しかし，従来システムではアプリケーションの制御に実行ファイルのハッシュ値を用いており，実行ファイルの改ざんにより設定済みのルールが適用されなくなる問題があった．また，複数のアプリケーションを一括制御するグループ化機能はサポートしていたが，複数のグループを同時に制御することはできなかった．そこで本論文では，デジタル証明書を用いることにより厳密な実行制御をするとともに，証明書の階層構造を用いて柔軟なグループ運用を実現する手法を提案する．We have developed an Application Execution Control System (traditional system) which provides individual application environments on each educational PC by controlling its execution. The traditional system uses a hash value of the execution file to control target application. If the execution file is falsified, a rule already set gets invalid. In addition, the traditional system has a function to set rules for multiple application at once. However, it cannot handle multiple groups at once. In order to solve these problems, this paper proposes a control method using digital certificates. It supports strict control and flexible group management by using hierarchical structure of digital certificates.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

近年,既存の光デバイスにより構築可能であり,かつ,波長単位よりも細かい粒度での帯域割当が可能な光ネットワークアーキテクチャとして光トレイルネットワークが注目されている.また,このアーキテクチャとトークンパッシング型メディアアクセス制御を前提として,トークン保持ノードにおいてトレイルを上流トレイルと下流トレイルへと分割し,それぞれのトレイルで独立してデータ転送を行うことによりスループットの向上を目指したトレイル二分割方式も提案されている.この方式は下流トークンの受信のみを契機としてトレイルを分割するため,上流トークン保持ノードの上流に位置するリンク上の帯域は未だ遊休状態のままである.本稿では,さらなるスループットの向上を目的として,トレイル多重分割方式を提案する.この方式は,下流トークンだけでなく上流トークンをもトレイル分割の契機とすることにより,上述した遊休状態の帯域を有効利用する.提案方式を用いた場合のトラヒック収容判定問題を線形計画問題として定式化し,それを解くことにより多重分割トレイルの最大実効スループットを導出し,従来の二分割トレイルに対する最大実効スループットの向上の程度を検討している.数値例によると,多重分割トレイルに都合の悪い(各ノードペアの送信レートが互いに等しい)トラヒックパターンでは,最大実効スループットの向上は見られないが,多重分割トレイルに都合の良い(隣接ノード間のみで送信が行われ,かつ,それらの送信レートが互いに等しい)トラヒックパターンでは,最大で約1.93倍の最大実効スループットを得ることができる.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

IaaS型クラウドサービスの一つとして,1つ以上のサーバとクライアントからなるネットワークアプリケーション(NW App)に対して,サーバが稼働できる計算機(WP:Working Place)を用意し,NW Appの通信品質が劣化したときに通信品質が良くなる別のWPにサーバを移動させるサービスが考えられている.しかし,サーバのファイルサイズは一般に非常に大きいため,その移動に伴い,背景トラヒックの通信品質の劣化を引き起こすので,この劣化の程度(インパクト)を極力低減することが重要である.本報告では,NW AppのSLA(Service Level Agreement)を満たすクライアントの総数(収容クライアント数)を多くしつつ,インパクトの低減を目指した3つの移動先決定アルゴリズムMIA,MRAおよびMCAを提案する.MIA,MRAおよびMCAでは,サーバ移動が必要なとき,それぞれ,移動に伴うインパクトが最小となるWP,全WPの残余収容サーバ数が極力均一となるWP,および,そのサーバの被覆度(SLAを満たす通信範囲)が最大となるWPに移動させる.数値例によれば,フルカバーWP(任意の位置のクライアントとSLAを満たす通信が行えるWP)が存在しない,もしくは,その収容サーバ数が小さい場合はMIAが最も優れており,それ以外の場合,MCAが最も優れている.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

IaaS型クラウドサービスの一つとして,1つ以上のサーバと各サーバに属する1つ以上のクライアントからなるネットワークアプリケーション(NW App)に対して,サーバ移動サービスを提供することが考えられている.このサービスでは,ネットワーク内に複数設置された計算機(WP: Working Place)上でサーバが稼働することができ,サーバは必要とあれば別のWPに移動(自発移動)することができる.ただし,移動先のWP内のサーバ数があらかじめ決められた上限値に等しい場合,自発移動のサーバを格納するために,そのWPに存在するいずれかのサーバを追い出して別のWPに移動(強制移動)させることもある.本報告では,まず,各サーバの移動の公平性の観点から,特定のサーバが過度に頻繁に自発移動できない,逆に過度に頻繁に強制移動させられることを防ぐことを目的として,各サーバの過去の自発移動回数と強制移動回数にもとづいて,追い出しサーバを絞り込む方法を3つ提案する.また,絞り込まれたサーバの中から,移動に伴うネットワークへのインパクトを極力小さくできるようなサーバの決定法を2つ提案する.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

近年,既存の光デバイスにより構築可能であり,かつ,波長単位よりも細かい粒度での帯域割当が可能なネットワークアーキテクチャとして光トレイルネットワークが注目されている.また,このアーキテクチャとトークンパッシング型メディアアクセス制御を前提として,トークン保持ノードにおいてトレイルを上流トレイルと下流トレイルへと分割し,それぞれのトレイルで独立してデータ転送を行うことにより最大スループットの向上を目指した分割型光トレイルネットワークも提案されている.分割トレイルにおける最大スループットの向上度合は,上流/下流トレイルでの各送信ノードのトークン保持時間をいかに適切に設定するかに依存する.本稿では,最大スループットの最大化を目的としたトークン保持時間設定問題を非線形計画問題として定式化し,それを解くことにより分割トレイルの最大スループットを導出し,非分割の通常トレイルに対する最大スループットの向上の程度を検討している.数値例によると,分割トレイルに最も都合の悪い(全ノードが終点ノードのみに送信を行う)トラヒックパターンでは,最大スループットの向上は見られないが,一様な(各ノード間の送信レートが互いに等しい)トラヒックパターンでは最大で約1.6倍,分割トレイルに最も都合の良い(隣接ノード間のみで送信が行われる)トラヒックパターンでは,最大で約1.9倍の最大スループットを得ることができる.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

近年,既存の光デバイスを用いて低コストで構築可能な光ネットワークアーキテクチャとして光トレイルネットワークが注目されている.トレイルは送受信ノード間で特定の波長を予約することにより設定される通信チャネルであり,トレイル上の任意のノードはその下流ノードへのパケット送信が可能である.トレイルは共有メディアであるため,パケットの衝突回避を目的としたメディアアクセス制御(MAC)が必要となる.従来,MAC方式としてトークンパッシング型のMAC方式が提案されているが,これらの方式ではトークン保持ノードの下流の帯域のみが使用され,その上流の帯域は遊休状態になるという問題が発生する.そこで,本研究では,トークンパッシング型MAC方式において,遊休状態にある帯域を有効に利用することによるスループットの向上を試みる.そのために,トークン保持ノードにおいて,トレイルを上流トレイルと下流トレイルの二つのトレイルへと分割し,それぞれのトレイルにおいてパケットを送信する動的トレイル分割方式を提案する.数値計算およびシミュレーションにより通常のトレイルと分割トレイルの最大スループットを比較評価した結果,提案法により最大スループットが最大で80%以上向上することが明らかとなった.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

近年,インターネットアクセス回線の高速化に伴い,動画配信のような大容量データを扱うサービスが続々と登場している.このようなサービスの急激な流行は,そのサービスに起因するトラヒック(増加トラヒック)が流れ込むバックボーン回線の容量不足といった問題を引き起こすため,その早期検出が重要となる.本報告では,サブネット単位での分散的なトラヒック計測を前提として,各サブネットでのトラヒック量と増加トラヒックの発生するサブネットの数の将来の増加傾向とを考慮して,将来のトラヒックの量を予測する部分集約法を提案し,前者のみを考慮する従来法(集約法)との検出性能をシミュレーションにより比較する.数値例によれば,部分集約法の方が最大で90日程度早期に増加トラヒックを検出できる.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

本稿では,P2Pライブストリーミングにおいて,リアルタイム性を維持するために,ストリーミング配信サーバから各ピアまでに経由するピアの数(論理ホップ)をあらかじめ決められた値以下するという前提の下で,参加可能ピア数の増加とAS間トラヒック量の減少を目標として二つのピア選択方式を提案している.一つ目の方法は,参加可能ピア数を増加させることを優先する方法であり,新規参加ピアは,既に参加しているピアのうちで,論理ホップの極力小さいピアを供給元として選択する.また,そのようなピアの数が多すぎるときには,AS間トラヒック量を減少させるために,自分が属するASと候補となるピアが属するASとの間のAS数(物理ホップ数)が極力小さくなるように候補を選ぶ.二つ目の方法は,AS間トラヒック量を減少させることを優先させる方法であって,一つ目の方法とは逆に,まず物理ホップの極力小さなピアを選択し,候補となるピア数が多すぎるときには,論理ホップ数の極力小さいピアを選択する方法である.数値例によれば,参加可能ピア数の面からは,前者が優れている.また,AS間トラヒック量の面からは,参加ピア数が少ない時点では,後者が優れているが,参加ピア数が増加してくると,前者が優れている.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

近年,安価で軽量なノートPCなどの携帯可能な端末の普及,かつ無線LANスポット等の外出先でのインターネット接続環境の充実により,場所を選ばないインターネット利用が可能となりつつある.しかし,TCPコネクションの確立中に端末が別のネットワークに移動した場合,通信遮断の継続による再送タイムアウトや端末のIPアドレスの変化に伴い,コネクション切断が発生するという問題点がある.そこで本稿では,VPNを用いて移動端末のIPアドレスの変化をTCP層に隠蔽し,さらにプロクシ方式によりTCPの再送を抑制することにより,端末の広域移動に対してTCPコネクショツを維持する方法を提案する.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

未知のウィルス・ワームの攻撃・感染の検知方法として、攻撃行動・感染行動に起因する変化点検出を利用する方法が有望視されている。しかし、この方法においては、ハードウェア障害等に起因する変化点をも検出してしまうため、攻撃行動・感染行動のみを検出するような絞り込み方法が求められている。このような方法として、ネットワーク全体で変化点が同時多発的に検出されるかどうかに着目する方法が考えられる。なぜなら、未知のウィルス・ワームの攻撃行動・感染行動は同時多発的に発生する傾向があるためである。そこで本稿では、分散配置したIDSからの変化点検出情報を集約し、ネットワーク全体で統合的に同時多発イベントの発生を検出する階層型検出機構を提案する。シミュレーションの結果、0.99の検出率を達成しようとする場合に、提案法はIDS単体での検出法よりも常に低い誤検出率を実現し、最大で約98%低減できることがわかった。

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

光バースト交換ネットワークにおいて,パケット棄却率を低減するための一つの方策として,オフセット時間(バースト生成終了時点から実際にそのバーストを転送する時点までの時間)を長めにとることが考えられる.しかし,オフセット時間中に到着するパケットは次のバーストに含まれることになるので,このままでは,パケットの伝送遅延が大きくなり過ぎることも考えられる.そこで,オフセット時間中に到着するであろうパケットの総量を予測し,これらのパケットも現バーストに含めることができるように各光コアルータで波長を予約する方法を提案する(提案法A).また,この方法の場合,予測した総量が大きすぎる場合,予約した期間の一部が無駄になることも考えられる.そこで,ここでは,オフセット時間中に再度,到着するであろうパケットの総量を予測し,最初の予約時間が大きすぎた場合には,余分な予約時間を解放するという方法を提案する(提案法B).数値例によるパケット棄却率の比較結果によれば,パケットの到着間隔がパレート分布の場合は,提案法AとBは同等の性能であり,ともに従来法より優れている.また,ON/OFFトラヒックに対しては,提案法Bが優れている.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

EDD(Earliest Due Date)コネクション受付制御方式において, 最悪リンク遅延を計算するためには, 有限個の離散的な時点(チェックポイント)について, ポイントスケジューラビリティ判定と呼ばれる処理が必要となる. 従来, この処理が必要なチェックポイントの上限値(チェックポイント上限値)を小さくすることにより, 導出時間を低減しようとする方法が提案されている. しかし, この方法では, 許容遅延の大きいコネクションが多数設定される場合には, 効果がほとんどないという問題点があった. 本報告の目的は, 従来法を改良して高速化することにある. ここでは, 従来法でのチェックポイント上限値より小さいチェックポイントのうち, ある一定範囲のチェックポイントについては, ポイントスケジューラビリティ判定が不要であることを明らかにし, これにより高速化を図っている. 数値例によれば, 許容遅延の大きいコネクションが多数設定される場合, 提案法の方が格段に高速である.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

EDD(Earliest Due Date)コネクション受付制御方式において, 最悪リンク遅延を計算するためには, 有限個の離散的な時点(チェックポイント)について, ポイントスケジューラビリティ判定と呼ばれる処理が必要となる.従来, この処理が必要なチェックポイントの上限値(チェックポイント上限値)を小さくすることにより, 導出時間を低減しようとする方法が提案されている.しかし, この方法では, ネットワーク負荷が一時的に高くなるような場合には, 効果がほとんどないという問題点があった.本報告の目的は, 従来法を改良することにある.従来法でのチェックポイント上限値は, 被覆関数と呼ばれる一次関数の傾きと切片に依存するが, 傾きの少しの増加に対して急激に増加し, 切片の多少の増加に対してはあまり増加しない.従来法では, 被覆関数を導出する際, 切片を最小値に固定した上で, 傾きを可変にして導出していたが, 負荷が一時的に高くなる場合, この傾きがかなり大きくなり, 結果的に導出時間が低減できない状況に陥っていた.本報告では, 従来法とは逆に, 傾きを最小値に固定した上で, 切片を可変とすることで高速化を図っている.数値例によれば, 負荷が一時的に高くなる場合, 改良法の方が格段に高速である.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

シャッフル網を対象とした従来のノード配置アルゴリズムDIST法およびSPARSE法では,経路はあらかじめ与えられると仮定していたために,与えられた経路によっては特定のリンクにトラヒックが集中してしまう可能性が高かった.これを緩和するために,本報告では,経路も設計変数とするノード配置アルゴリズムNew_DIST法およびNew_SPARSE法を提案している.New_DIST法(New_SPARSE法)は,ノード配置についてはDIST法(SPARSE法)と同様であり,経路ついては,ノード間最短経路の中でトラヒックが極力分散されるような経路を選ぶ.4種類の典型的なトラヒック行列に対する数値結果によれば,いずれかの提案法を用いれば,2つの従来法より,優れた性能を達成することができる.

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記述言語：英語   出版者・発行元：一般社団法人電子情報通信学会  

This paper proposes a new supply current test method for detecting floating gate defects in CMOS ICs. In the method, unusual increase of the supply current caused by defects is promoted by superposing an AC component on the DC power supply. Feasibility of the test is examined by some experiments on four DUTs with an intentionally caused defect. The results showed that our method could detect clearly all the defects, one of which may be detected by neither any functional logic test nor any conventional supply current test.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

EDDコネクション受付制御方式では,コネクションの設定要求が発生した場合,それに属するパケツトの最悪の場合のend-to-end遅延を計算し,この遅延がそのコネクションの許容するend-to-end遅延以下のときのみその設定を許可するが,それらの遅延の差(遅延余裕)を何時の時点でリンクに割り当てるか(リンクで使用するか)ということが問題となる.従来の遅延余裕割当法では,この割当てをコネクションの設定要求が許可された時点で行なっていたため,ネツトヮーク内に同時に設定可能なコネクションの最大数(最大コネクション数)が小さくなる可能性が大きかった.本報告では,設定要求が許可された時点で割当てを行なわないで遅延余裕を保存しておき,将来の時点で必要が生じたときに初めて割り当てる方法を提案する.数値例によれば,提案法での最大コネクション数は従来法に比して小さくなることはなく,最大で約30%増加させることができる.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

本稿では,テーブル参照型FPGAを対象に,任意のconfiguration(プログラム)に対して正しく機能するか否かを検査するための方法を提案する.このタイプのFPGAは,論理ブロック,クロスポイントスイッチブロック,スイッチマトリックスおよびIOブロックの4種類のブロックから構成される.ここでは,まず,各ブロックを単独で検査する場合のconfigurationとテストパターンを導出している.次に,FPGA上での各ブロックの配置の規則性に着目して,上述のconfigurationとテストパターンを,同種の複数のブロックに対して同時に適用できるように拡張している.これによれば,FPGA全体の検査がアレイサイズの行数(列数)のオーダのconfigurationにより実行できる.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

マルチウィンドウ合成方式として広く用いられている描画時合成方式では,管理ソフトウェア(例えば,Xウィンドウシステムにおけるサーバ)において,ウィンドウ間の重なり処理や再描画処理が実行されるが,ウィンドウの移動やリサイズなどのように,短い時間間隔で連続的に要求が発生するウィンドウ操作の場合,各操作に伴うこれら2つの処理がこの時間間隔内に終了せず,連続的操作全体の実行時間がユーザの許容範囲を越えてしまう.一方,この問題点を緩和できる方式として拡張表示時合成方式が知られている.この方式では,各ウィンドウの画像(他のウィンドウに隠されている画像も含む)が別々のメモリに格納され,これらから同時に読み出した画像が,各ウィンドウの表示優先順位にもとづいてハードウェア的に合成される.従って,管理ソフトウェアにおける重なり処理は不要となるが,多数のウィンドウの表示を可能とするためには,メモリ量が膨大になる.本報告では,拡張表示時合成方式の利点を極力生かしつつ,メモリ量を大幅に低減することのできる合成方式を提案する.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

Shuffle-Like Switching Network(SL-Net)は,n×2^k個のノードを構成される.本報告では,SL-Netを対象に,ネットワーク形態がエンドユーザ間のパケット伝送遅延に与える影響について検討している.まず,各ノードにおいて,パケット発生確率が一様,かつ,迂回ルーチングを採用するという前提のもとで,SL-Netのnとkの種々の組合せに対してシミュレーションを行い,伝送遅延を解析している.次に,この結果から,ノード数が与えられたときのSL-Netの最適な形態は,平均最短パス長が最小となる付近の(n,k)の組合せの中で,nの値が最大となるような形態であることを明らからしている.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

本報告では,Manhattan Street Network(MS-Net)およびShuffleNet(SH-Net)を対象にして,迂回ルーチングを行ったときのエンドユーザ間での性能をシミュレーションによって評価・比較している.まず,ユーザからネットワーク内へのパケットの送出モードとして,従来のFIFOモードの他に新たにSCANモードを考え,正常であるときの各ネットワークの性能評価結果から,SCANモードの方が優れていることを明らかにしている.次に,SCANモードを用いるという前提下で,2つのネットワークの性能を相互に比較し,正常時,故障時(1本のリンク故障時)のいずれにおいても,SH-Netの方が優れていることを明らかにしている.

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記述言語：英語   出版者・発行元：IEICE-INST ELECTRON INFO COMMUN ENG  

The locally exhaustive testing of multiple output combinational circuits is the test which provides exhaustive test patterns for each set of inputs on which each output depends. First, this paper presents a sufficient condition under which a minimum test set (MLTS) for the locally exhaustive testing has 2w test patterns, where w is the maximum number of inputs on which any output depends. Next, we clarify that any CUT with up to four outputs satisfies the condition, independently of w and n, where n is the number of inputs of the CUT. Finally, we clarify that any CUT with five outputs also satisfies the condition for 1 less-than-or-equal-to w less-than-or-equal-to 2 or n-2 less-than-or-equal-to w less-than-or-equal-to n.

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記述言語：日本語  

プロセッサ制御回路は機能的にシーケンサ（），タイミングコントローラ（）およびコンディション・ディサイダに分割されるが，このうちのSCについては，命令実行により検査するという前提の下で，既に最小テスト集合(の入力と状態との組のすべてを生起する最小数の命令実行）導出法が与えられている．しかし，この方法では，並行して実施できるTCの検査が全く考慮されていない．本論文では，TCの検査も考慮したときのSCの最小テスト集合導出法を示している．この方法によれば，TCの入力と状態との組を最大限に検査できるような，SCの最小テスト集合が得られる．また，その最小テスト集合で生起できない(の）入力と状態との組に対するテスト集合の要素数は最小となる．A processor contlor circuit can be separated into three blocks, Sequencer (SC), Timing Controller (TC), Condition Decider. There are some studies on testing each block with instruction executions, where a test set is a set of instruction executions which generate all of the input-state vectors in the block. This paper describes a method of generating the minimum test set for SC which covers the input-state vectors in TC as many as possible. In case of Intel 8080 processors, about 95 percent of the input-state vectors in TC are covered by the test set for SC obtained by the use of the method.

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