Publishing type：Research paper (scientific journal)  

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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DOI： 10.1109/ICTC55196.2022.9952427

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Other Link： https://dblp.uni-trier.de/db/conf/ictc/ictc2022.html#DeguchiKTFY22

Publishing type：Research paper (international conference proceedings)   Publisher：IEEE  

DOI： 10.1109/ictc52510.2021.9621088

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Publishing type：Research paper (international conference proceedings)   Publisher：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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Other Link： https://dblp.uni-trier.de/db/conf/icoin/icoin2021.html#UrimotoFTMY21

Publishing type：Research paper (international conference proceedings)   Publisher：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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Other Link： https://dblp.uni-trier.de/db/conf/icoin/icoin2021.html#OishiTFY21

Publishing type：Research paper (international conference proceedings)   Publisher：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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Other Link： https://dblp.uni-trier.de/db/conf/ictc/ictc2020.html#NumataITFY20

Publishing type：Research paper (international conference proceedings)  

© 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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Publishing type：Research paper (international conference proceedings)   Publisher：IEEE  

DOI： 10.1109/ICTC46691.2019.8939877

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Other Link： https://dblp.uni-trier.de/db/conf/ictc/ictc2019.html#KishimotoOTFY19

Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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DOI： 10.1109/NTMS.2018.8328725

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Publishing type：Research paper (international conference proceedings)  

© 2018 IEEE. Server migration service (SMS) has been proposed as a new class of service to augment the existing IaaS (Infras-tructure as a Service) cloud service. SMS allows servers (server-side processes of a network application) to dynamically and automatically migrate close to their clients (client-side processes of the network application) in order to reduce the penalty that the SMS provider pays to is SMS subscribers when failing to provide them with the guaranteed level of QoS. In this paper, we tackle a link capacity provisioning and server location decision problem where we consider the sum of link capacity provisioning cost and the penalty as the total expenditure of SMS business, and aim at minimizing the total expenditure. In the problem, we determine how much capacity to add to links, and when and to which location to migrate servers. We formulate the problem as an integer programming model, solve the model, and obtain the optimal link capacities, server locations at each time, and the global optimum value of the total expenditure of SMS business. Numerical examples show that the proposed method decreases the total expenditure of SMS business by up to 49% compared to a conventional method that determines the server locations while fixing the link capacities.

DOI： 10.1109/CloudNet.2018.8549545

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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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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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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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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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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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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (scientific journal)  

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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)  

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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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Language：English   Publishing type：Research paper (scientific journal)  

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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

The programming circuit of SRAM-based FP-GAs consists of the 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 all application of the test procedures to test Xilinx XC4025.

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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：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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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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Language：English   Publishing type：Research paper (scientific journal)  

Reliability and performance for telecommunication networks have been traditionally investigated separately in spite of their close relation. A design method integrating them for a reliable packet switched network, called a proofing method, was discussed in previous papers. This paper first presents the proofing method in detail. Then two heuristic design approaches (max-average, max-delay-link) for optimizing network cost in the proofing method are described. In order to verify their effectiveness and applicability, they are compared numerically for three example network topologies. Finally, the sensitivity of these two methods is examined with respect to changes in traffic demand and in link reliability. Numerical results show that the max-delay-link method provides a lower minimum network-cost than does the max-average method, for both a small and a large example network. The answers obtained by these two methods are not highly sensitive to changes either in traffic demand or in link reliability. Thus a network designed by these two methods is robust to a system change which is not considered at a design stage. The max-average method is superior to the max-delay-link method in terms of the sensitivity. Many other sources of failures must be considered in our failure model, eg, node and software faults. Only statistically independent failures are considered. Statistical dependence effects must be included to make the model more realistic. The max-average and max-delay-link methods cannot prevent a state where there are no routes between a particular source and destination node pair. To cope with this, a topological design method must be added to these methods. The computational complexity of these methods needs to be clarified to identify how large a practical problem can be solved using them. © 1991 IEEE.

DOI： 10.1109/24.93767

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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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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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Language：English   Publishing type：Research paper (scientific journal)  

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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Language：English   Publishing type：Research paper (international conference proceedings)   Publisher：I E E E, COMPUTER SOC PRESS  

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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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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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Language：English   Publishing type：Research paper (scientific journal)   Publisher：ELSEVIER SCIENCE BV  

DOI： 10.1016/0169-7552(86)90059-0

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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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As a new optical network architecture that can be built with currently available devices and can achieve bandwidth allocation with granularity finer than a wavelength, the light-trail architecture attracts attention. For the light-trail architecture using token passing medium access control, the split trail architecture is proposed for improving the throughput. In the architecture, a trail is split into the upstream trail and the downstream trail at the token holding node and independent data transmissions on the two trails are permitted. However, the split trail architecture leaves the bandwidth of links upstream of the upstream-token holding node idle because the split of the trail is driven by the reception of only the downstream-token. In this paper, we propose the multi-split trail architecture for further improving the throughput. Our architecture utilizes the idle bandwidth by splitting the trail driven by the reception of both upstream- and downstream-tokens. We formulate the decision problem of the traffic accommodation where we optimize the token holding times so that the input traffic is successfully accommodated, as a linear programming problem. Then, we derive the maximum effective throughput of the multi-split trail architecture by solving the problem, and investigate the degree of improvement over the split trail architecture. According to numerical examples, the multi-split trail architecture achieves 1) the same maximum effective throughput as the split trail architecture for an uniform traffic pattern where every node pair has the same sending rate, and 2) about 1.93 times as high maximum effective throughput for its favorite traffic pattern where every transmission node sends data to its adjacent downstream node only and their rates are identical.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

Server migration service provides network applications (NW Apps) with server running environment on a computer (WP) and service of server migration to another WP where the NW App can attain better QoS when it is degraded. Because the file size of a server is large, the traffic for server migration causes QoS degradation of its background traffic. Thus, it is important to decrease the degree (impact) of the QoS degradation. In this paper, we propose three destination selection algorithms MIA, MRA and MCA that try to decrease the impact while increasing the number of NW App's clients whose SLA are satisfied. When a server migration is triggered, MIA, MRA and MCA move the server to the WP with the minimum impact, the WP with the maximum remaining space for servers and the WP with the maximum communication range where the SLA is satisfied, respectively. Numerical examples show that when there is no full-cover WP (on which a server can communicate with an arbitrary client while satisfying the SLA) or its space is small, MIA outperforms the others, otherwise MCA outperforms the others.

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Server migration service is considered as one of IaaS-typed cloud services. The target of the service is such a application that consists of servers and their clients, and, each server works on a computer called a working place (WP) and can voluntarily migrate to another WP if necessary. However, if the destination WP is full of servers, a server in the WP may be forced to be pushed out of the current WP and migrate to another WP to accept the voluntary immigration server. In this paper, in order to protect some particular servers from experiencing voluntary migration or forced migration much more than the other servers, in the viewpoint of migration fairness, three methods to determine the candidates to be pushed out based on the numbers of voluntary migrations and forced migrations. Next, we propose two methods to determine such one pushed-out server from the candidates that the impact caused by forced migration against the network becomes as small as possible.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

As a new optical network architecture that can be built with currently available devices and can achieve bandwidth allocation with granularity finer than a wavelength, the light trail architecture attracts attention. For the light trail architecture using token passing medium access control, the split trail architecture is proposed. In the architecture, a trail is split 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 architecture achieves higher maximum throughput than the original unsplit trail architecture. The degree of throughput improvement by the split trail one depends on how appropriately we set upstream and downstream token holding times of each transmission node. In this paper, we formulate decision problem of the token holding times as a non-linear 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 one. According to numerical examples, the split trail architecture achieves 1) 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 an uniform traffic pattern where every node pair has the same sending rate, 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.

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As a new optical network architecture that can be built with currently available devices, light trail networks attract attention. A trail is an unidirectional communication channel which is set up by reserving a specific wavelength along a route between a source node and a destination node, and packet communication from an arbitrary node to its downstream nodes is possible. Because a trail is a shared medium, we need medium access control (MAC) to avoid collisions of packets. Token passing type MAC methods such as light-trail fair access MAC (LT-FA MAC) and adaptive round time MAC (ART MAC) have been proposed. In those methods, however, downstream bandwidth of the token holding node is only used, and upstream bandwidth becomes idle. In this paper, we try to improve throughput of a trail by using those idle bandwidth effectively in a token passing type MAC method. In order to achieve this, we propose a dynamic trail partitioning method that dynamically partitions a trail into upstream trail and downstream one at the token holding node, and permits parallel transfer of data using two trails. Numerical calculations and simulations show that the dynamic trail partitioning method can improve throughput by a maximum of more than 80%.

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Language：Japanese   Publisher：The Institute of Electronics, Information and Communication Engineers  

With the spread of broadband access lines, many services making efficient use of them such as video streaming services 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 which cause these problems early. In this paper, we propose an early detection method (partial aggregation method) of increasing traffic under per-subnet based distributed traffic monitoring. The method estimates future overall traffic volume by taking account of estimated traffic volume in all subnets and the estimated number of subnets generating increasing traffic. We compare the proposed method to a previous method (aggregation method) which takes account of estimated traffic volume in all subnets only. The results show that the partial aggregation method can detect increasing traffic earlier than the aggregation method by a maximum of 90 days.

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In this paper, we propose two peer selection algorithms to increase the number of joining peers and decrease the traffic amount between ASs in P2P live streaming, where we assume that the number (logical hop count) of relaying peers is limited by a predetermined value in order to keep real-time property of live streaming. In the first method, 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 traffic amount between ASs. In the second peer selection method, a newly joining peer selects providing peers in the reverse order of the first selection method. Simulation results show that the former increases the number of joining peers compared to the latter. Moreover, when the number of joining peers is small (large), the traffic amount between ASs in the latter (the former) is smaller.

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In recent years, inexpensive and lightweight portable terminals such as laptop PCs are spreading, and outdoor Internet services such as wireless hot spots are increasing. Therefore, users can connect to the Internet at various locations by their own terminal. However, if a terminal establishing TCP connections starts to move to another network, there is a problem that TCP connections are aborted due to retransmission timeout and the change of terminal's IP address. In this paper, we propose a method to preserve TCP connections for macro mobility of terminals by hiding the change of IP address from TCP layer using VPN and by preventing retransmission using proxies.

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Change-point detection schemes are a promising approach for detecting network anomalies such as attacks and infections by unknown viruses and worms. They detect those behaviors as change-points. In general, however, because they also detect false-positive change-points, those caused by other factors such as hardware troubles, we need a scheme that only detects true-positive change-points caused by attacks and infections. True-positive change-points tend to occur simultaneously, and the number of true-positive change-points is very large, while false-positive change-points tend to occur sporadically. We exclude false-positive change-points by neglecting change-points that occur sporadically, based on information gathered from the whole network. In this paper, we propose a multi-stage network anomaly detection scheme that aggregates change-point information from distributed IDSs (Intrusion Detection Systems) and detects the true-positive change-points. Simulation results illustrate that, compared to a scheme using only one IDS, our method always yields a smaller false-positive rate, a reduction of up to 98%, under a constraint that the detection rate of the true-positive change-points must exceed 0.99.

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In optical burst switching networks, a larger value of the offset time (the time duration between the burst assembly finishing time and the transmission time of the burst) is desirable to decrease packet loss rate. However, a larger offset value introduces a longer delay at edge routers because packet arriving during the offset time are not assembled into the current burst in conventional methods. In this paper, we consider a strategy where a larger offset time is used and packets arriving during the offset time are assembled into the current burst. In order to attain the policy, we propose two methods A and B. In the method A, we estimate the amount of packets arriving duration the offset time and we reserve wavelength channels so that the packets can be transmitted by the current burst. The method A however has a weak point that the reserved time is partly useless when the estimated amount is too large. In order to resolve it, the method B again estimates the amount of arrived packets halfway through the offset time and releases the redundant reserved time when the second estimated amount is less that the first one. Numerical examples shows the effectiveness of the proposed methods, and the method B is superior to the method A when the packet arrival process is the Pareto ON-OFF process.

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In order to calculate the worst-case link delays in the EDD (Earliest Due Date) connection admission control scheme, a check called the point schedulability check is necessary for each of finite and discrete time instants (checkpoints). In our previous calculation method, high speed calculation is attained by decreasing an upper bound value of checkpoints. However, the method can not calculate at high speed when many connections with large allowable end-to-end delays are established. The purpose of the paper is to improve the previous method. In this paper, we clarify that we do not have to execute the point schedulability check for an interval of checkpoints which are less than the upper bound value in the previous method, and based on the fact, high speed calculation can be attained. Numerical examples show that the proposed method drastically decreases the calculation time when many connections with large allowable end-to-end delays are established.

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In order to calculate the worst-case link delays in the EDD (Earliest Due Date) connection admission control scheme, a check called the point schedulability check is necessary for each of finite and discrete time in stants (checkpoints). In our previous calculation method, high speed calculation is attained by decreasing a upper bound value of checkpoints. However, the method can not calculate at high speed when networks are temporarily in congested state. The purpose of the paper is to improve the previous method. The upper bound value of the previous method depends on the slope and the intercept of a linear function called a covering function, and its value drastically increases for the small increase of the slope and slowly increases for the small increase of the intercept. Because the previous method uses a covering function whose slope is large in the temporal congested state, the calculation time becomes large. The improved method uses a covering function whose slope is minimum and whose intercept somewhat larger than the previous method, and consequently the upper bound value of checkpoints become smaller. The improved method therefore can attain high speed calculation in the temporal congested state. Numerical examples show that the improved method drastically decrease the calculation time in the temporal congested state.

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Two heuristic node placement algorithms, DIST and SPARSE, have been proposed for Shuffle-Like Networks (SL-Nets). Since a route between two nodes is assumed to be given, the algorithms have the problem that traffic amounts over some links may be too large. In this report, we propose two new node placement algorithms, New_DIST and New_SPARSE, in which routes are assumed to be design variables. New_DIST (New_SPARSE) places nodes in the same way as DIST (SPARSE), and selects such a shortest path between two nodes that traffic amounts of links are balanced. Numerical examples for four typical traffic matrices show that the better performance can be obtained by using one of the proposed algorithms for each matrix.

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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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In the EDD connection admission control scheme, a connection is established if its allowable end-to-end delay is larger than or equal to the worst-case end-to-end delay, and the timing of assignment of the difference (called a delay margin) between the two delays is an important issue. In conventional assignment methods, the assignment is performed when the connection establishment request is accepted. In this paper, we propose a new assignment method in which the assignment is not performed when the connection establishment request is accepted, and is performed when it is necessary to establish a connection in the future. Simulation results show that the maximum number of established connections by the proposed method is roughly equal to or up to 1.30 times as large as those by conventional ones.

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This paper presents a universal testing method which can check whether unprogrammed look-up table FPGAs(Field Programmable Gate Arrays) are fault free or not. Such an FPGA is constructed with four kinds of blocks, configurable logic block, cross-point switch block, switch matrix and IO block. First, configurations and test patterns for testing each block are derived. Second, they are extended so that several blocks of the same kind can be simultaneously tested. The FPGA can be tested with O(N) configurations, where N is the row size (column size) of the FPGA.

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This paper describes the influence of network topologies on end- to-end packet transmission delays in Shaffie-Like Switching Networks.The network consists of n×2^k nodes.The influence is anal yzed by means of simulation in which n and k are varied in various combinations.And it is carified that,if the number of nodes is held constant,the best topology is obtained at the maximum value of n under the condition that the average minimum end-to-end path length is held near the minimum value.

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This paper evaluates performances of two distributed type switching networks,the Manhattan Street Network(MS-Net)and ShuffleNet(SH-Net),by simulation.The performance measure is average end-to-end transmission delay.Packets enter the networks in the order of generation(FIFO mode)or an order determined by their destinations(SCAN mode),and traverse the networks along the routes decided by deflection routing.SCAN mode is superior to FIFO one in either network if it is,fault-free.In both of fault-free case and a single link failure case,SH-Net is superior to MS-Net if SCAN mode is adopted.

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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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