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掲載種別：研究論文（学術雑誌）   出版者・発行元：Institute of Electrical and Electronics Engineers (IEEE)  

DOI： 10.1109/access.2021.3133851

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DOI： 10.1587/transcom.2018EIP0001

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

DOI： 10.1109/LANMAN.2019.8846996

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DOI： 10.1587/transcom.2018EBP3170

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DOI： 10.1587/transcom.2018EBP3154

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DOI： 10.1587/transcom.2018EBP3045

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

DOI： 10.1109/GLOCOM.2018.8647130

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DOI： 10.1587/transcom.2018EBP3043

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）  

DOI： 10.23919/ICMU.2018.8653625

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

In recent years, software-defined networking (SDN), which performs centralized network management with software, has attracted much attention. Although packets are transmitted based on flow entries in SDN switches, the number of flow entries that the SDN switches can handle is limited. To overcome this difficulty, this paper proposes a flow-based routing method that performs flexible routing control with a small number of flow entries. The proposed method provides mixed integer programming. It assigns common paths to flows that can be aggregated at intermediate switches, while considering the utilization of network links. Because it is difficult for mixed integer programming to compute largescale problems, the proposed method also provides a heuristic algorithm for them. Through numerical experiments, this paper shows that the proposed method efficiently reduces both the number of flow entries and the loads of congested links.

DOI： 10.1587/transcom.2017ITP0007

Scopus

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

This paper first investigates how a network operates when multiple receivers download content simultaneously in content-centric networking (CCN) when the receivers’ downloading speeds differ. The results indicate that the performance of the download completion time of a faster user degrades excessively due to a decrease in the cache-hit rate in the router. Based on the investigation, this paper proposes a novel in-network caching method for simultaneous download from multiple receivers in CCNs. The proposed method keeps cached data packets in a router until slower receivers download the data, in order to prevent slower users from directly downloading data from the content provider. We conduct computer simulations and confirm the effectiveness of the proposed method. We show that the proposed method can improve the download completion time performance in the situation where multiple receivers download content at different speeds in CCN.

DOI： 10.2197/ipsjjip.25.164

Scopus

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

This paper proposes a data center traffic engineering scheme with Markov approximation. Markov approximation is a distributed optimization framework. It optimizes networks by independent behaviors of users that construct a time-reversible continuous-time Markov chain based on minimum necessary information in the networks. The proposed scheme aims at minimizing the maximum link utilization in data center networks by means of Markov approximation. The proposed scheme provides two strategies with different search space. Through simulation experiments, we show that the proposed scheme efficiently reduces the maximum link utilization under not only static situations but also dynamic situations where traffic demands dynamically change in data center networks.

Web of Science

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

In recent years, software-defined networking (SDN), which performs centralized network management with software, has attracted much attention. In SDN, the data plane is separated from the control plane. Network administrators can dynamically change the setting of the data plane and control routing on the control plane through a control protocol such as OpenFlow. Although packets are transmitted based on flow entries in the data plane, the number of flow entries that data plane can have is limited. It is difficult to perform flexible routing control with a small number of flow entries. To overcome this difficulty, this paper proposes a routing method that reduces the number of flow entries, while reducing the maximum link utilization. The proposed method allocates common paths to flows that can be aggregated, considering the utilization of the links. Through numerical experiments, this paper shows the effectiveness of the proposed method.

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

Communication infrastructures under the influence of the disaster strike, e.g., earthquake, will be partitioned due to the significant damage of network components such as base stations. The communication model of the Internet bases on a location-oriented ID, i.e., IP address, and depends on the DNS (Domain Name System) for name resolution. Therefore such damage remarkably deprives the reachability to the information. To achieve robustness of information retrieval in disaster situation, we try to apply CCN/NDN (Content-Centric Networking/Named-Data Networking) to information networks fragmented by the disaster strike. However, existing retransmission control in CCN is not suitable for the fragmented networks with intermittent links due to the timer-based end-to-end behavior. Also, the intermittent links cause a problem for cache behavior. In order to resolve these technical issues, we propose a new packet forwarding scheme with the dynamic routing protocol which resolves retransmission control problem and cache control scheme suitable for the fragmented networks. Our simulation results reveal that the proposed caching scheme can stably store popular contents into cache storages of routers and improve cache hit ratio. And they also reveal that our proposed packet forwarding method significantly improves traffic load, energy consumption and content retrieval delay in fragmented networks.

DOI： 10.1587/transcom.2016CNP0003

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

Quantized congestion notification (QCN), discussed in IEEE 802.1Qau, is one of the most promising Layer 2 congestion control methods for data center networks. Because data center networks have fundamentally symmetric structures and links are designed to have high link utilization, data center flows often pass through multiple bottleneck links. QCN reduces its transmission rate in a probabilistic manner with each congestion notification feedback reception, which might cause excessive regulation of the transmission rate in a multiple-bottleneck case because each bottleneck causes congestion feedbacks. We have already proposed QCN with bottleneck selection (QCN/BS) for multicast communications in data center networks. Although QCN/BS was originally proposed for multicast communications, it can also be applied to unicast communications with multiple bottleneck points. QCN/BS calculates the congestion level for each switch based on feedback from the switch and adjusts its transmission rate to the worst congestion level. In this paper, we preliminarily evaluate QCN/BS in unicast communications with multiple tandem bottleneck points. Our preliminary evaluation reveals that QCN/BS can resolve the excessive rate regulation problem of QCN but has new fairness problems for long-hop flows. To resolve this, we propose a new algorithm that integrates QCN/BS and our already proposed Adaptive BC_LIMIT. In Adaptive BC_LIMIT, the opportunities for rate increase are almost the same for all flows even if their transmission rates differ, enabling an accelerated convergence of fair rate allocation among flows sharing a bottleneck link. The integrated algorithm is the first congestion control mechanism that takes into account unicast flows passing through multiple tandem bottleneck points based on QCN. Furthermore, it does not require any modifications of switches used in QCN. Our simulation results show that our proposed integration of QCN/BS and Adaptive BC_LIMIT significantly mitigates the fairness problem for unicast communications with multiple bottleneck points in data center networks.

DOI： 10.1587/transcom.2015EBP3404

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

Content oriented network is expected to be one of the most promising approaches for resolving design concept difference between content oriented network services and location oriented architecture of current network infrastructure. There have been proposed several content oriented network architectures, but research efforts for content oriented networks have just started and technical issues to be resolved are still remained. Because of content oriented feature, content data transmitted in a network can be reused by content requests from other users. Pervasive cache is one of the most important benefits brought by the content oriented network architecture, which forms interconnected caching networks. Caching network is the hottest research area and lots of research activities have been published. This paper surveys recent research activities for caching networks in content oriented networks, with focusing on important factors which affect caching network performance, i.e. content request routing, caching decision, and replacement policy of cache. And this paper also discusses future direction of caching network researches.

DOI： 10.1587/transcom.2015AMI0001

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

In recent years, much work has been devoted to developing protocols and architectures for supporting the growing trend of data-oriented services. One drawback of many of these proposals is the need to upgrade or replace all the routers in order for the new systems to work. Among the few systems that allow for gradual deployment is the recently-proposed Breadcrumbs technique for distributed coordination among caches in a cache network. Breadcrumbs uses information collected locally at each cache during past downloads to support in-network guiding of current requests to desired content. Specifically, during content download a series of short-term pointers, called breadcrumbs, is set up along the download path. Future requests for this content are initially routed towards the server which holds (a copy of) this content. However, if this route leads the request to a Breadcrumbs-supporting router, this router re-directs the request in the direction of the latest downloaded, using the aforementioned pointers. Thus, content requests are initially forwarded by a location ID (e.g., IP address), but encountering a breadcrumb entry can cause a shift over to content-based routing. This property enables the Breadcrumbs system to be deployed gradually, since it only enhances the existing location-based routing mechanism (i.e. IP-based routing). In this paper we evaluate the performance of a network where Breadcrumbs is only partially deployed. Our simulation results show Breadcrumbs performs poorly when sparsely deployed. However, if an overlay of Breadcrumbs-supporting routers is set-up, system performance is greatly improved. We believe that the reduced load on servers achieved with even a limited deployment of Breadcrumbs-supporting routers, combined with the flexibility of being able to deploy the system gradually, should motivate further investigation and eventual deployment of Breadcrumbs. In the paper, we also evaluate more coarse level than router level, i.e. ISP-level Breadcrumbs deployment issues. Our evaluation results show that Higher-layer first deployment approach obtains great improvement caused by Breadcrumbs redirections because of traffic aggregation in higher layer ISP.

DOI： 10.1587/transcom.2014EBP3471

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

Content Centric Networking (CCN) is one of the most promising content-oriented network architectures. In CCN, forwarding information base (FIB) is allowed to have multiple entries for a same content name prefix, which means CCN potentially supports multi-source download. When a content is obtained from multiple sources, the technical knowledge obtained for congestion control in the current Internet cannot be simply applied. This is because in the current Internet, FIB is restricted to have only one entry for each IP address prefix, which causes quite different path feature from CCN. This paper proposes a new congestion control for CCN with multi-source content retrieval. The proposed congestion control is composed of end-to-end window flow control and router assisted Interest forwarding control and enables transmission rate regulation only on a congested branch.

DOI： 10.1145/2984356.2985235

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

In content oriented networks (CON), routers in a network are generally equipped with local cache memories and store incoming contents temporally. Efficient utilization of total cache storage in networks is one of the most important technical issues in CON, which brings reduction of content server load, content download latency and network traffic. Performance of networked cache is reported to strongly depend on combination of caching decision and content request routing. In this paper, we evaluate several combinations of these two strategies. Especially for routing, we take up off-path cache routing, Breadcrumbs, as one of evaluated content request routings. Our performance evaluation results show that off-path cache routing, Breadcrumbs, obtains poor performance with caching decisions which has good performance with shortest path routing (SPR), and obtains good performance with TERC (Transparent En-Route Cache) which is well-known to have poor performance with widely used SPR.

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

In-network guidance for off-path caching, called Breadcrumbs, has been proposed for cache networks. In Breadcrumbs, during content download, a router holds a pointer, breadcrumb, which denotes the latest download direction. When a content request occasionally encounters a breadcrumb along its default-path to the repository, location-based routing is switched to hop-by-hop breadcrumb-based routing and it is forwarded to off-path cache. In our previous work, we evaluated Breadcrumbs in fixed wireless multi-hop networks and showed that total network throughput is significantly improved. In this paper, we focus on mobile wireless multi-hop networks where Breadcrumbs trail might be fragmented due to node mobility. To resolve this technical problem in mobile environment, we propose a new in-network guidance method, overhearing-aided Breadcrumbs. In overhearing-aided Breadcrumbs, a router which overhears content download by other nodes makes one-hop breadcrumb. Our evaluation results show that proposed overhearing-aided Breadcrumbs outperforms conventional Breadcrumbs in mobile wireless multi-hop networks.

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

In this paper, we first investigate how a network works when multiple receivers download content simultaneously in content-centric networking (CCN). We are especially interested in the case that the receivers' downloading speeds differ. From our investigation, we show that the download completion time of a faster user degrades excessively due to the problem that there being different downloading speeds decreases the cache-hit rate. To solve this problem, we propose a novel cache management method for simultaneous downloading from multiple receivers for CCN. The concept of the proposed method is to keep cached data packets in a router until they are transferred to the receiver. The proposed method increases the cache-hit rate for a slower receiver, and prevents a slower user from directly downloading data from the content provider. Through computer simulations, we confirm that the proposed method can improve the download completion time performance in the situation that multiple receivers download content at different speeds in CCN.

DOI： 10.1109/WAINA.2016.133

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

Large-scale content transfer, especially video transfer, is now a dominant traffic component in the Internet. Originally, content transfer had a content-oriented feature, i.e., "Users do not care where content is retrieved. Users only take care of what content they obtain." Conventional traffic engineering (TE) aims to obtain optimal routes for traffic between ingress and egress router pairs, i.e., TE has focused on a location-oriented approach that takes care of where to connect. With increased demand for content-oriented features for content traffic, TE needs to focus on content-oriented routing design. In this study, we therefore propose a novel approach to content-oriented TE, called content aware routing (CAR). In CAR, routes are designed for content and egress router pairs, i.e., content traffic toward a receiver-side router. Content demand can be flexibly distributed to multiple servers (i.e., repositories) providing the same content, meaning that content can be obtained from anywhere. CAR solves the optimization problem of minimizing maximum link utilization. If there are multiple optimal solutions, CAR selects a solution in which resource usage is minimized. Using numerical examples formulated by the linear programming problem, we evaluated CAR by comparing it with combinations of conventional content delivery networks and TE, i.e., location-oriented designs. Our numerical results showed that CAR improved maximum link utilization by up to 15%, with only a 5% increase of network resource usage.

DOI： 10.1587/transcom.E98.B.575

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

Data center network is composed of high-speed Ethernet extended in a limited area of a data center building, so its RTT is extremely small of mu sec order. In order to regulate data center network delay large part of which is queuing delay, QCN is proposed for layer 2 congestion control in IEEE 802.1Qau. QCN controls transmission rate of the sender by congestion feedback from a congested switch. QCN adopts probabilistic feedback transmission to reduce the control overhead. When the number of flows through a bottleneck link increases, some flows might receive no feedback even in congestion phase due to probabilistic feedback transmission. In this situation, queue length might be significantly fluctuated. In this paper, we propose a new delay-based congestion detection and control method. Our proposed delay-based congestion control is cooperated with the conventional QCN so as to detect and react congestion not detected by QCN.

DOI： 10.1587/transcom.E98.B.585

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

In content oriented application, such as P2P, a user does not care about where content is obtained and a content file can be downloaded from anywhere it is obtained. In some P2P systems, e.g. BitTorrent, a content file is divided into small parts, chunks, and each chunk is downloaded from anywhere. In this many-to-one communication style, one TCP session is set up for each pair of a sender and a receiver. If these sessions go through the same bottleneck link, many-to-one communication might obtain unfairly large throughput. In this paper, we propose a new receiver-driven congestion control for this many-to-one communications. Our proposed congestion control detects bottleneck-sharing flows and clusters them. Window coupling technique is applied for these clustered flows, which enables these flows to obtain fair aggregated throughput. Our simulation results show that our proposed receiver-driven congestion control fairly shares a bottleneck link with regular TCP.

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

Network traffic for sharing contents is significantly increasing. The cache network is a desirable architecture to reduce traffic and to improve reliability of contents retrieval. "Breadcrumbs" is one of the promising techniques to find cached contents in a distributed manner. Thanks to Breadcrumbs, users can retrieve a content without managing where the cache exists. This demo shows a contents sharing application among mobile users in the Breadcrumbs-enabled cache network.

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

Communication infrastructures under the influence of the disaster strike e.g., earthquake, will be partitioned due to the damage of components such as base stations. The communication model of the Internet bases on a location-oriented ID, i.e., IP address, and depends on the DNS (Domain Name System) for name resolution. Therefore such damage remarkably deprives the reachability to the information. To achieve robustness of information retrieval in disaster situation, we try to apply CCN (Content-Centric Networking) to fragmented networks. Since CCN does not require name resolution, it can work without the global system such as DNS. However, existing retransmission control in CCN is not suitable for the fragmented networks with intermittent links due to the timer-based end-to-end scheme. Though the requirement to save energy and bandwidth are critical in disaster case, overhead caused by redundant retransmissions might waste bandwidth and energy. Accordingly, we propose the energy efficient retransmission scheme with the dynamic routing protocol suitable for fragmented networks. With the proposed method, messages are forwarded hop-by-hop with recognizing whether the state of links is connected or not. Simulation results reveal that the proposed method can reduce traffic load, energy consumption and delay of contents retrieval.

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

Because of the popularity of rich content, such as video files, the amount of traffic on the Internet continues to grow every year. Not only is the overall traffic increasing, but also the temporal fluctuations in traffic are increasing, and differences in the amounts of traffic between peak and off-peak periods are becoming very large. Consequently, efficient use of link bandwidth is becoming more challenging. In this paper, we propose a new system for content distribution: storage aware routing (SAR). With SAR, routers having large storage capacities can exploit those links that are underutilized. Our performance evaluations show that SAR can smooth the fluctuations in link utilization.

DOI： 10.1587/transcom.E97.B.2641

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

In data center networks, group communication is currently playing an important role and multicast communications is an effective way to support group communication for large numbers of virtual machines. Layer-2 congestion control named QCN (Quantized Congestion Notification) has been proposed to realize the high reliability required by LAN/SAN integration in data center networking. Our preliminary evaluation in this paper shows that a multicast flow suffers lower throughput than unicast flows when conventional QCN is applied in a naive manner. This is because a sending device receives congestion feedback from multiple locations on a multicast tree and decreases transmission rate accordingly. To counter this throughput degradation of multicast flows, we propose a new Layer 2 congestion control algorithm in multicast environment, Quantized Congestion Notification with Bottleneck Selection (QCN/BS). In QCN/BS, the switch in the worst congestion level is selected and the transmission rate of the sending device is calculated exclusively according to feedback from the selected switch. Simulation results show that when conventional QCN is used, a multicast flow experiences lower and more severely unfair throughput than a unicast flow. The proposed QCN/BS resolves this problem.

DOI： 10.1587/transcom.E97.B.1121

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

In wireless multi-hop networks, content servers are generally located outside a wireless multi-hop network and a user accesses these servers through a gateway node. So, content traffic has a tendency to be concentrated at a gateway node, which might cause throughput degradation in a whole wireless multi-hop network. Caching network is one promising way to resolve this technical problem. However, caching network has a limitation of cache availability on a default-path. For wired networks, Breadcrumbs has been proposed as in-network guide for cached contents. In this paper, we evaluate Breadcrumbs in wireless multi-hop networks. Our simulation results show that Breadcrumbs improves total throughput performance of wireless multi-hop networks. And also they surprisingly show that not only popular content throughput but also unpopular content throughput is improved.

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記述言語：英語   出版者・発行元：The Institute of Electronics, Information and Communication Engineers  

ID/Locator separation architecture is proposed in order to reduce routing table size in the current Internet. In this architecture, an end-host can be identified by unique ID which is independent of its network protocol. Network protocol independency of ID provides possibility of communication with end hosts operating different network-layer protocol. This inter-layer 3 networking brings several benefits in addition to routing table size improvement, e.g. improvement of shortest path and robustness. In our previous paper, performance of inter-layer 3 networking from the viewpoint of shortest path improvement has been evaluated. In this paper, we focus on improvement of robustness brought by inter-layer 3 networking in the case of node failure. Node failure generally suffers serious degradation of connectivity because all links connected to a node are failed. Our simulation results show that significant improvement of robustness can be obtained only with small number of vertical links going through multiple network planes.

DOI： 10.1587/comex.2.372

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記述言語：英語   出版者・発行元：一般社団法人電子情報通信学会  

Network topology significantly affects network cost, path length, link load distribution, and reliability, so we need to consider multiple criteria with different units simultaneously when designing a network′s topology. The analytic hierarchy process (AHP) is a technique of balancing multiple criteria in order to reach a rational decision. Using AHP, we can reflect the relative importance of each criterion on the evaluation result; therefore, we have applied it to network topology evaluation in past research. When evaluating network topologies using AHP, we need to construct the set of topology candidates prior to the evaluation. However, the time required to construct this set greatly increases as the network size grows. In this paper, we propose applying a binary partition approach for constructing a topology candidate set with dramatically reduced calculation time. To reduce the calculation time, we introduce an upper limit for the total link length. Although the results of AHP are affected by introducing the upper limit of the total link length, we show that desirable topologies are still selected in AHP.

DOI： 10.1587/transcom.E96.B.163

CiNii Books

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

In large-scale data centers, two types of network are implemented: local area networks (LANs) and storage area networks (SANs). To achieve simple network management, integration of these two networks by Ethernet technology is of great interest. A SAN requires a significantly low frame loss rate. To integrate LANs and SANs, a multi-hop Ethernet configuration is generally used, and congestion may occur in traffic hot spots. Therefore, layer-2 congestion control that prevents frame loss in multi-hop Ethernet, Quantized Congestion Notification (QCN), is now discussed in IEEE 802.1Qau. In this paper, we evaluate QCN's throughput performance and reveal a technical problem with fairness among active flows. We also propose Adaptive BC_LIMIT for QCN where BC_LIMIT is adaptively decided according to current transmission rate of flows. Simulation results show that our proposed method significantly improves fairness among QCN flows.

DOI： 10.1587/transcom.E96.B.99

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

In recent years, much work has been devoted to developing protocols and architectures for supporting the growing trend of data-oriented services. One drawback of many of these proposals is the need to upgrade or replace all the routers in order for the new systems to work. Among the few systems that allow for gradual deployment is the recently-proposed Breadcrumbs technique for distributed coordination among caches in a cache network. Breadcrumbs uses information collected locally at each cache during past downloads to support in-network guiding of current requests to desired content. Specifically, during content download a series of short-term pointers, called breadcrumbs, is set up along the download path. Future requests for this content are initially routed towards the server which holds (a copy of) this content. However, if this route leads the request to a Breadcrumbs-supporting router, this router re-directs the request in the direction of the latest downloaded, using the aforementioned pointers. Thus, content requests are initially forwarded by a location ID (e. g., IP address), but encountering a breadcrumb entry can cause a shift over to content-based routing. This property enables the Breadcrumbs system to be deployed gradually, since it only enhances the existing location-based routing mechanism (i.e. IP-based routing).
In this paper we evaluate the performance of a network where Breadcrumbs is only partially deployed. Our simulation results show Breadcrumbs performs poorly when sparsely deployed. However, if an overlay of Breadcrumbs-supporting routers is set-up, system performance is greatly improved. We believe that the reduced load on servers achieved with even a limited deployment of Breadcrumbs-supporting routers, combined with the flexibility of being able to deploy the system gradually, should motivate further investigation and eventual deployment of Breadcrumbs.

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

In research community of content distribution, a new communication paradigm of rendezvous-based abstraction which easily enables location-free content-oriented services, attracts great attention. ID/Locator separation architecture is originally proposed for resolving increase of routing table size in the current Internet but it has possibility of supporting this abstraction because host ID is clearly separated from its location. This feature of ID/Locator separation means an end host can be identified by ID independent of its network layer protocol. So, ID/Locator separation architecture enables communication with end host(s) operating different network-layer protocol, which might increase candidate paths between host pairs and improve content distribution path. In this paper, we evaluate this improvement of shortest path brought by inter-layer 3 networking. In inter-layer 3 networking, a shared node connecting different network layer plane plays an important role. We evaluate shortest path improvement with various shared node locations and show that strategic assignment of shared nodes brings large improvement. When multiple layer 3 networks are available for users, shortest path might be improved even only with multi-homing (without inter-layer 3 networking). We also evaluate shortest path improvement brought by multi-homing and inter-layer 3 networking for incremental deployment scenario of multi-homing. Our simulation results show that inter-layer 3 networking brings great improvement even with small number of users deploying multi-homing while only multi-homing itself brings small improvement.

DOI： 10.1587/transcom.E94.B.2741

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

In ad hoc wireless networks, end-to-end TCP is well-known that it can obtain very low throughput with increase of the number of hops. One promising way to resolve this technical problem of TCP throughput degradation is proxy-based TCP. In proxy-based TCP, an end-to-end TCP session is divided by (an) intermediate proxy host(s), which leads to reduction of hop-count of each divided session and improvement of TCP throughput. However, we show that proxy-based TCP surprisingly does not bring performance improvement and obtain lower throughput than conventional end-to-end TCP in a realistic situation of limited buffer space in proxy host(s). The reason for this degradation is interruption of segment transmission caused by wireless channel capture around the proxy host. In this paper, for resolving this problem, we propose intentional window How control which prevents wireless channel capture around the proxy host. In our proposed intentional window flow control method, the proxy host and the destination host return advertised window size which is intentionally set to small value almost independently from its vacant buffer space. Performance evaluation results show that our proposed intentional flow control with small advertised window size prevents wireless channel capture around the proxy host and significantly improves proxy-based TCP performance.

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

In wireless multi-hop networks, when an end host would like to communicate with external wired hosts, data packets are to be transmitted at first to a gateway node which is connected to both a wireless multi-hop network and a wired network. When there are multiple candidate gateways to be connected, a wireless host should select one gateway as its own default gateway. One possible and simple way is minimum hop gateway selection policy. This minimum hop gateway selection may cause performance degradation in the case of imbalance of traffic distribution. In this paper, we propose a new gateway selection method where an end host selects the default gateway from the viewpoint of gateway load balancing. Wireless communication channel around the gateway is very important resource because all the external traffic goes through the gateway. So, gateway load in our proposed method takes account of not only traffic intensity at a gateway but also interference of wireless channel around a gateway. Our simulation results show that our proposed method brings improvement of gateway throughput performance.

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）  

ALM (Application Level Multicast) is one of the promising ways for resolving multicast incentive problem of non-multicast hosts not implementing multicast routing protocol. ALM protocol is implemented in application layer, which means it is easier to be altered by end users than lower layer protocols. Malicious users would like to get higher position in ALM tree because it can obtain higher throughput and lower delay. They also would not like other hosts to be connected because of forwarding burden will bring them wireless throughput degradation. This paper evaluates performance improvement perceived by a malicious user and shows that surprisingly it can obtain performance improvement only with almost 50% probability. We also investigate the reason why this performance effect and show that link sharing with other ALM hosts in long hops is one of the reasons of performance degradation of malicious users. ©2009 IEEE.

DOI： 10.1109/APCC.2009.5375478

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）  

Multicast communication is an e.cient packet dissemination method in ad hoc network. However, in ad hoc multicast communication, packet collisions due to the hidden terminal problem happen frequently because each host does not use the RTS/CTS frames that has been used in conventional unicast communication. So, wireless channel is not reserved for any multicast packet transmissions and hidden terminal problem degrades multicast communication performance. One promising way for improving hidden terminal problem is directional antenna. Because interference can be decreased by directional antenna, the hiding terminal problem can be improved. In this paper, we evaluate basic performance of multicast communication with directional antenna in ad hoc networks. Performance evaluation by computer simulation shows that throughput performance can be improved by using directional antenna because the radio wave interference decreases and the hidden terminal problem that is brought by some independent multicast sessions can be controlled. ©2009 IEEE.

DOI： 10.1109/APCC.2009.5375481

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

Application-level multicast has drawn a lot of attention as an alternative to IP multicast. In application-level multicast, multicast related features, such as group membership management, packet replication and packet forwarding are implemented at end-hosts instead of routers. The host perceived transmission quality and multicast forwarding responsibility depend on its position in the multicast distribution tree. This nature of application-level multicast motivates selfish members to alter their position by unrightful means to maximize their private benefits. Uncooperative behaviors of these selfish members, i.e. cheating, increase unfairness between selfish members and faithful members. In the context of bulk data distribution, this unfairness between members significantly impacts the receiver throughput. In this paper, to alleviate the negative impact of cheating members, we propose a new tree building protocol which builds dual multicast trees. Our proposed protocol constructs a shortest-widest path tree as the 1st tree. The members having lower position in the 1st tree are located at higher position in the 2nd tree in exchange for their Unfairness. To investigate performance of our proposed protocol, it is compared with the existing application-level multicast protocol. Our simulation results show that our protocol outperforms the existing protocol from the view point of throughput and resource utilization against member cheating.

DOI： 10.1093/ietcom/e91-b.12.3872

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

In Multicast is an effective way of dissemination of the same information to potentially large number of users. In IP multicast, a multicast IP address is used for identifying each multicast session. When large number of various contents should be disseminated, e.g. software distribution to various types of information home appliances, lots of multicast addresses are necessary. Source Port Specific Multicast (SPSM) is an effective way of saving IP multicast addresses. In SPSM, both of a source port of transport layer and an IP address in network layer is used for identification of multicast session. When SPSM routers are partially deployed, i.e. SPSM routers and non-SPSM routers (IP multicast router) co-exist in a network, unnecessary traffic may be generated because non-SPSM routers cannot identify multicast sessions having the same multicast address. In this paper, we analyze this unnecessary traffic volume with various deployment scenes of SPSM routers.

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

In ad hoc networks, a TCP session can obtain very low throughput with increase of the number of hops. One promising way to resolve this technical problem of TCP throughput degradation is proxy-based TCP. In proxy-based TCP, an end-to- end TCP session is divided by (an) intermediate proxy host(s), which leads to reduction of RTT of each divided session and improvement of TCP throughput. We evaluate proxy-based TCP in ad hoc networks with a realistic assumption of finite buffer in proxy hosts. Our simulation results show that proxy-based TCP surprisingly does not bring performance improvement and obtain lower throughput than conventional end-to-end TCP. In the paper, we show that wireless channel capture around the proxy host is the reason for this degradation. And we propose adaptive rate control method and intentional flow control method at the proxy host which improves throughput performance of proxy-based TCP. Performance evaluation results show that proxy-based TCP with our proposed methods outperforms conventional end-to-end TCP.

DOI： 10.1109/GLOCOM.2008.ECP.163

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記述言語：英語   出版者・発行元：Information and Media Technologies 編集運営会議  

Ad hoc network which requires no infrastructure equipment such as access points is paid great attention in recent years. In ad hoc network, each host controls its wireless access in a distributed fashion. Generally, carrier sensing is used for detecting channel availability. By using RTS and CTS control frames, MAC sub-layer predicts the channel use and resolves hidden-terminal problem. Broadcast nature of RTS and CTS control frames leads to inefficient spatial reuse. One promising solution for this inefficient spatial reuse is usage of directional antenna. In ad hoc networks, heterogeneous network configuration is general, which means each host is equipped with different network facilities. For antennas, it will be general situation that not all hosts are equipped with directional antenna. This paper evaluates throughput performance of the ad hoc network in this realistic situation and shows how much performance improvement deployment of directional antenna will bring to the network.

DOI： 10.11185/imt.2.907

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

In this paper, we discuss TCP performance in a wireless overlay network where wireless LANs and cellular networks are integrated. In the overlay network, vertical handover, where a mobile node changes its access link during a session, is one of the most important technologies. When a vertical handover occurs, throughput performance of a TCP flow is degraded due to not only packet losses during the handover, but drastic change of its bandwidth-delay product. In this paper, we propose an ACK-pacing mechanism for TCP congestion control to improve the performance degradation. The proposed system is receiver-driven, so no modification is required to the mechanism of TCP sender. In the proposed system, a TCP receiver adjusts a transmission rate of ACKs according to the relationship between bandwidth-delay products before and after a handover. Since the ACK-clocking mechanism of TCP adjusts the transmission rate of TCP segments, the TCP receiver can seamlessly adjust its congestion window size to the new bandwidth-delay product. Computer simulation results show that the proposed system can improve the TCP performance during the vertical handover.

DOI： 10.1093/ietcom/e90-b.4.885

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

Recently the content distribution networks (CDNs) are highlighted as the new network paradigm which can improve latency for Web access. In CDNs, the content location strategy and request routing techniques are important technical issues. Both of them should be used in an integrated manner in general, but CDN performance applying both these technologies has not been evaluated in detail. In this paper, we investigate effect of integration of these techniques. For request routing, we focus on a request routing technique applied active network technology, Active Anycast, which improves both network delay and server processing delay. For content distribution technology, we propose a new strategy, Popularity-Probability, whose aim corresponds with that of Active Anycast. Performance evaluation results show that integration of Active Anycast and Popularity-Probability provides robust CDNs.

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

Recently the content distribution networks (CDNs) are highlighted as the new network paradigm which can improve latency for Web access. In CDNs, the content location strategy and request routing techniques are important technical issues. Both of them should be used in an integrated manner in general, but CDN performance applying both these technologies has not been evaluated in detail. In this paper, we investigate effect of integration of these techniques. For request routing, we focus on a request routing technique applied active network technology, Active Anycast, which improves both network delay and server processing delay. For content distribution technology, we propose a new strategy, Popularity-Probability, whose aim corresponds with that of Active Anycast. Performance evaluation results show that integration of Active Anycast and Popularity-Probability provides robust CDNs.

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記述言語：英語   出版者・発行元：Information and Media Technologies 編集運営会議  

Self-similar traffic patterns have been observed in many measurements of Internet traffic. Self-similarity is very detrimental to the performance of packet networks and recent research has focused on understanding and reducing its amount in Internet traffic. TCP Reno has been identified as being one of the primary sources of self-similarity. We explore the potential of another version of TCP in this paper to reduce the degree of self-similarity in aggregated TCP traffic. We decompose both TCP Reno and TCP Vegas to demonstrate and explain their underlying mechanisms, and separately measure what effects congestion-avoidance and timeouts/exponential backoff mechanisms have on the self-similarity in aggregated TCP flows. We reveal how TCP Vegas reduces the degree of self-similarity and eventually completely eliminates it from aggregated TCP flows at low levels of packet loss. However, at high levels of packet loss we show that TCP Vegas is detrimental, because it increases the degree of aggregated TCP-flow self-similarity.

DOI： 10.11185/imt.2.347

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

IP multicast is an effective technology to distribute identical data simultaneously to multiple users. However, for technical and administrative reasons, IP multicast has not been globally deployed on the Internet. Another approach to multicast is application-level multicast (ALM). In ALM, multicast related features, such as group membership management, multicast routing and packet replication, are implemented at end-hosts instead of routers. Multicast delivery tree is constructed in the application layer, so all nodes in this tree are end-hosts. Packet transmission between end-hosts uses conventional IP unicast service. Therefore, ALM is a promising alternative to IP multicast. However, application-level multicast relying on end-hosts is more fragile than IP multicast relying on routers. Especially, when a mobile host forwards packets to downstream hosts, a handover causes performance degradations on downstream hosts. In this paper, to alleviate the impact of a handover, we propose a new tree building protocol which locates mobile hosts on leaves of multicast tree. A handover of the mobile host on a leaf does not affect other end-hosts. To investigate performance of our protocol, it is compared with existing application-level multicast protocols. Our simulation results show that our protocol outperforms existing protocols from the viewpoints of loss probability, throughput and delay performance.

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記述言語：英語   出版者・発行元：Information Processing Society of Japan  

There have been no studies evaluating the dominant factor responsible for generating self-similar network traffic, although many researchers have discussed the causes of self-similarity. In this paper, we have isolated and compared the effects of the application, transport and network layers on self-similarity. We first focus on the interaction between the transport and network layers only. With a simple computer simulation scenario we show that in realistic bottleneck scenarios with only long greedy TCP flows present, the TCP flows fully utilize the bottleneck link, which make the aggregated TCP flow short-range dependent while each flow that share the bottleneck link itself is long range dependent. We also show that in special situations, when the TCP flows become synchronized, the bottleneck link may become underutilized giving space to long-range dependence to surface. We show the connection between the network layer parameters necessary for synchronization to occur, and verify our results with simulation. To examine the effect of the application layer, we introduce web traffic besides the long greedy flows into the bottleneck link. We show that link utilization is again a key factor determining the self-similar properties of the aggregated flow, but has much less impact on the self-similar properties of the aggregated web sessions and long greedy flows. Finally, we point out the most influential effects and the conditions when they dominate the self-similar properties of aggregated TCP flows.

DOI： 10.2197/ipsjdc.1.576

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科研費基盤研究 20040401-20060331

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

In multicast congestion control, the receiver of the worst congestion level is selected as the representative and transmission rate of the sender is adjusted to TCP throughput of the representative. This approach has high scalability and TCP friendliness. However, when this approach is applied in wireless communications, wireless-caused packet loss will cause to frequent change of the representative. This is because degradation of wireless channel quality causes bursty packet loss at a corresponding receiver. Fading, one of main reasons of wireless channel degradation, is expected to be recovered after short time period, which leads to frequent change of the representative. This frequent change of representative makes the sender adjust its transmission rate to the tentative worst receiver, which brings severe performance degradation to wireless multicast. We call this technical problem, the wireless-caused representative selection fluctuation problem. Wireless-caused representative selection fluctuation problem is one of scalability problems in the wireless multicast, because this problem remarkably happens for large scale multicast. We propose two possible solutions for this problem, an end-to-end approach and a network support approach. Performance evaluation in various situation show that an end-to-end approach is sensitive for its inferring error but a network support approach shows good performance improvement.

DOI： 10.1093/ietcom/e88-b.7.2819

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

科研費基盤研究 20040401-20060331

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

In the mobile Internet, a handover brings significant performance degradation of TCP due to bursty packet losses during handover processing. In this paper, we propose a new bandwidth control for improving the TCP performance. In the proposed system, when a mobile node changes its accessing base station, an intermediate router suppresses an available bandwidth to the corresponding TCP flow. Because suppressing the bandwidth results in reducing mis-forwarded packets to the old base station, the bursty packet losses can be avoided. In the hierarchical mobile network structure, which is recently developed in order to realize micromobility protocol, all packets transferred to mobile nodes are converged to several gateways such as mobility anchor points (MAP) in hierarchical Mobile IPv6 (HMIPv6). Therefore, the proposed system is suited to the hierarchical structure because it can be easily implemented at such gateways. Computer simulation results show that the proposed system can improve the TCP performance degradation especially in a situation where handovers frequently occur.

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

Application-level multicast (ALM) is a feasible alternative to IP multicast. In ALM, multicast related features, such as group membership management, multicast routing and packet replication, are implemented at end-hosts instead of routers. A multicast distribution tree is constructed in the application layer, so all nodes in this tree are end-hosts. Packet transmission between end-hosts uses conventional IP unicast service. Therefore, all end-hosts can enjoy multicast communications without IP multicast service. However, ALM has a serious problem that the multi-cast distribution tree is intrinsically fragile and an end-host failure causes tree partitions. In this paper, to deal with this problem, we propose a new tree construction protocol which makes outdegrees of intermediate nodes be balanced. The degree-balanced distribution tree can reduce the average number of nodes decoupled by tree partitions. To investigate performance of our protocol, it is compared with an existing ALM protocol. Our simulation results show that our protocol outperforms the existing protocol from the viewpoints of robustness, loss probability and receiver-perceived delay. Copyright © 2005 The Institute of Electronics, Information and Communication Engineers.

DOI： 10.1093/ietcom/e88-b.12.4427

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

In content distribution networks(CDNs), the request routing which directs user requests to an adequate server from the viewpoint of improvement of latency for obtaining contents is one of the most important technical issues. In the request routing, network support technology enables a router to select an adequate server by using information of server load and network delay. In this system, a network support router selects a server based on information between router and server, so the placement of network support router is important to improve performance of request routing. In this paper, we investigate an efficient placement method of network support router making use of statistical characteristics of Internet topology.

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

Application-level multicast (ALM) is a feasible alternative to IP multicast. In ALM, multicast related features, such as group membership management, multicast routing and packet replication, are implemented at end-hosts instead of routers. A multicast distribution tree is constructed in the application layer, so all nodes in this tree are end-hosts. Packet transmission between end-hosts uses conventional IP unicast service. Therefore, all end-liosts can enjoy multicast communications without IP multicast service. However, ALM has a serious problem that the multicast distribution tree is intrinsically fragile and an end-host failure causes tree partitions. In this paper, to deal with this problem, we propose a new tree construction protocol which makes outdegrees of intermediate nodes be balanced. The degree-balanced distribution tree can reduce the average number of nodes decoupled by tree partitions. To investigate performance of our protocol, it is compared with an existing ALM protocol. Our simulation results show that our protocol outperforms the existing protocol from the viewpoints of robustness, loss probability and receiver-perceived delay.

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

In this paper, we discuss TCP congestion control in the future heterogeneous mobile Internet where various access links such as a wireless LAN and a cellular network are integrated. In the heterogeneous network, vertical handover, which is a handover technique between different types of access links, is one of the most important technologies. In the heterogeneous network, since mobile terminals can simultaneously receive packets from several wireless interfaces during a handover processing, a seamless vertical handover for layer 2 and layer 3 can be realized and as a result, no packet loss occurs due to the vertical handover. However, the vertical handover significantly degrades throughput performance of TCP because it causes drastic change in a bandwidth delay product of each TCP flow. In this paper, we propose an ACK-pacing mechanism for TCP congestion control to realize a seamless layer 4 handover. The proposed system is receiver-driven, so no modification is required to the mechanism of TCP sender. In the proposed system, a TCP receiver adjusts a transmission rate of ACKs according to the relationship between delay-bandwidth products before and after a handover. Since the ACK-clocking mechanism adjusts the transmission rate of TCP, TCP receiver can seamlessly adjust its congestion window size to the new delay-bandwidth product by transmission rate control of ACK. Computer simulation results show that the proposed system can improve the TCP performance during the vertical handover.

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科研費基盤研究 20040401-2006-0331

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

The window flow control based end-to-end TCP congestion control may cause unfair resource allocation among multiple TCP connections with different RTTs (round trip times) at a bottleneck link. In this paper, in order to improve this unfairness, we propose the active ECN which is an ECN based active queue mechanism (AQM). A bottleneck router with the proposed mechanism marks TCP segments with a probability which depends on the RTT of each connection. By enabling the TCP senders to reduce their transmission rate when their packets are marked, the proposed mechanism can realize the same transmission rate among TCP connections with different RTTs. Further-more, the active ECN can directly mark ACKs from TCP receivers. while the conventional ECN marks TCP segments coming from the TCP senders. As a result, the queue length distribution at the bottleneck link gets stabilized, because the sender can quickly react to the marking according to variation of the queue length.

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

In wireless environment, TCP suffers from significant performance degradation due to bit errors on wireless link and handovers because it responds to all packet losses by invoking congestion control even though packet losses are not related to congestion. Several schemes have been proposed to improve the performance degradation due to each cause. They have been evaluated in a specific network environment where either bit errors or handover occurs, i.e. they do not occur at the same time. In this paper, we reveal the packet recovery mismatch problem in an environment where both of bit errors and handover can cause the performance degradation. We pick up one scenario that TCP traffic is transmitted in the situation that ARQ (Automatic Repeat reQuest) and packet forwarding are implemented together. They are proposed to reduce the influence of bit errors and handover respectively and are natural approaches from the viewpoint of protocol layering. Computer simulation shows that in that scenario both techniques cannot perform efficiently due to interaction of each other. We also propose two buffer control approaches to resolve the packet recovery mismatch problem in our scenario according to applicability of cross-layer operation between layer 2 and layer 3.

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

Large parts of the Internet are still incapable of native multicast, and ubiquitous deployment of multicast will take a long time. There are two approaches to provide wide-area multicast service in today's Internet. One is tunneling approach and the other is application-level multicast approach. In this paper, we focus on application-level multicast approach and propose a new scheme which improves the performance penalties of application-level multicast by making use of network support. Because in application-level multicast, endhosts provide multicast functionality instead of routers, application-level multicast delivery tree is inherently less efficient than IP multicast tree. Therefore, in our scheme, the router on the application-level multicast delivery tree alters the tree based on network-level delivery path. We evaluate our scheme with simulation experiment. Our simulation results indicate that our scheme improves the performance of application-level multicast. Further we compare our scheme to the tunneling approach from the viewpoint of transmission performances. The results reveal applicable domains of both approaches.

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

TCP traffic has been reported to have not only self-similarity but also a multi-fractal nature. The multi-fractal nature of traffic volume in general is well known to have a significant influence on queueing behavior in short time scales. In our previous study, we showed that the Retransmission Time Out (RTO) mechanism plays an important role in the multi-fractal nature of TCP traffic.
In this paper we investigate how to control the multifiractal nature of TCP traffic in a real network Here we focus on the Random Early Detection (RED) mechanism in order to reduce the occurrence of RTO. We evaluate the RED mechanism from the viewpoint of multi-fractal nature and show that the RED mechanism improves not only throughput and delay performance, but also the multifractal nature of TCP traffic.

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

In mobile Internet, a handover may result in significant degradation of transmission performance. Especially in a shared medium wireless network, handover traffic which moves into a new wireless access network competes with existing traffic for available bandwidth of shared medium wireless channel. This interaction between handover traffic and existing traffic causes significant performance degradation of both of traffic. In this paper, we propose a new concept of Transport Layer Graceful Degradation. We think handover traffic should increase its available bandwidth gradually and available bandwidth of existing traffic is to be decreased slowly to bandwidth which achieves fair share of shared medium with handover traffic. In the paper, we propose a new bandwidth control method for TCP traffic which achieves our proposed concept, transport layer graceful degradation. Performance evaluation results show that our proposed scheme can improve TCP throughput performance of both handover traffic and existing traffic.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

従来,マルチキャストとユニキャストの公平性を扱う場合には,TCP親和性のように,リンクの帯域を公平利用するという,微視的な観点からの研究が主流であった.本論文では,マルチキャストがネットワーク帯域を効率的に使用するという点を考慮し,巨視的な観点から,マルチキャストをユニキャストに対し優先的に扱うアプローチについて検討を行う.具体的には,廃棄率の点でマルチキャストフローを優遇するパケット廃棄法を提案し,その有効性を性能評価によって明らかにする.

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記述言語：英語   掲載種別：研究論文（学術雑誌）  

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

In content distribution networks (CDNs), the content routing which directs user requests to an adequate server from the viewpoint of improvement of latency for obtaining contents is one of the most important technical issues. Several information, e.g. server load or network delay, can be used for content routing. Network support, e.g. active network, enables a router to select an adequate server by using these information. In the paper, we investigate a server selection policy of a network support approach from the viewpoint of which information to be used for effective server selection. We propose a server selection policy using RTT information measured at a router. Simulation results show that our proposed server selection policy in content routing selects a good server under both conditions where server latency and network delay is a dominant element of user response time. Furthermore, we also investigate about location of routers with network support bringing good performance for our proposed scheme.

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

Multicast transmission, which can send the same information simultaneously to multiple users, is a key technology in content delivery networks. In this paper, we discuss a new multicast architecture with network coding proposed by Li et al., which breaks limitation of existing IP multicast in terms of network resource utilization. Network coding based multicast can achieve the max-flow, which is the theoretical upper bound of network resource utilization. However, the max-flow transmission is not always effective and may not be robust against congestion because it maximally uses link capacity of multicast distribution tree. In this paper, we first introduce a load balancing method of network coding as an alternative use to the max-flow transmission. Next, we study the feasibility of network coding based multicast architecture from performance aspect and evaluate the network coding in terms of the max-flow and load balancing with a computer simulation. There has been no evaluation of network coding in practical network environment with packet losses and propagation delay. We also describe required key techniques and technical problems to implement network coding on the current IP networks. Our results will offer valuable insight for designing the future Internet with higher and more effective network utilization.

Web of Science

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

TCP/IP is a key technology in the next generation mobile communication networks. A significant amount of wireless traffic will be carried in the Internet, and wireless connections will have to share network resources with wired connections. However, in a wireless network environment, TCP suffers significant throughput degradation due to the lossy characteristic of a wireless link. Therefore, to design the next generation mobile networks, it is necessary to know how much the wireless connection suffers from the degradation in comparison to the wired connection. In this paper, we discuss the fairness issue between TCP connections over wireless and wired links, and theoretically analyze the fairness of throughput between TCP over wireless link with ARQ (Automatic Repeat reQuest)-based link layer error recovery and TCP over error-free wired link. We validate our analysis by comparing the numerical results obtained from the analysis with the results obtained from computer simulation. The numerical results show that the fairness is sensitive to network propagation delay and variation rapidity of wireless link characteristic. We also obtain the theoretical lower bound of fairness.

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

One of the most important technical problems in reliable multicast protocol is reducing redundant feedback information, e.g. NAKs, to avoid feedback implosion. A number of feedback suppression mechanisms have been proposed to deal with this problem. In the MBONE, which is a virtual multicast network and makes multi-point communication across the Internet feasible, the source link, the links directly connected to or very close to the source, is reported to contribute high percentile packet loss. When a well-known NAK suppression mechanism is applied, in the case of the source link loss, all receivers suffer the same packet loss and NAK suppression mechanism does not work effectively. In this paper we propose a Reliable Multicast Protocol Applying Local FEC, called Local FEC, where the source link loss is recovered with FEC applied locally only to the source link. To investigate performance of our proposed protocol, it is compared with a popular reliable multicast protocol with NAK suppression mechanism. Our performance analysis results with mathematical analysis and Computer simulation show that our proposed protocol outperforms the NAK suppression protocol from the viewpoint of scalability and wasted bandwidth.

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

There have been few studies evaluating the dominant factor responsible for generating self-similar network traffic while many researchers have discussed the causes of self-similarity. In this paper, we have isolated and compared the effects of the application, transport and network layers on the self-similarity of TCP flows and their aggregates. With a simple computer simulation scenario, we show that in networks where congestion frequently occurs the transport layer dominates the self-similarity of single TCP flows over the application layer. Next we focus on the interactions between the transport and network layer. We show that on high packet loss rates the exponential backoff technique of the TCP is responsible for the elevated level of self-similarity while on low packet loss rates, the synchronization between the TCP flows may induce high degree of self-similarity of the aggregated TCP sessions. To support this conjecture on synchronization, we show synchronization is only present when the TCP flows of the the aggregate operate mostly in congestion avoidance mode.

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

Large parts of the Internet are still incapable of native multicast, and ubiquitous deployment of multicast will take a long time. There are two approaches to provide wide-area multicast service in today's Internet. One is tunneling approach and the other is application-level multicast approach. In this paper, we focus on application-level multicast approach and propose a new scheme which improves the performance penalties of application-level multicast by making use of network support. Because in application-level multicast, endhosts provide multicast functionality instead of routers, application-level multicast delivery tree is inherently less efficient than IP multicast tree. Therefore, in our scheme, the router on the application-level multicast delivery tree alters the tree based on network-level delivery path. We evaluate our scheme with simulation experiment. Our simulation results indicate that our scheme improves the performance of application-level multicast. Further we compare our scheme to the tunneling approach from the viewpoint of transmission performances. The results reveal applicable domains of both approaches.

Web of Science

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

TCP traffic has been reported to have not only self-similarity but multi-fractal nature. Multi-fractal nature of traffic volume, in general, is well known to have significant influence to queueing behavior in short time scale. In the paper, we investigate about "what component of TCP congestion control mechanism affects to multifractal nature of generated traffic?". We artificially divide TCP congestion control mechanism to several components and evaluate multi-fractal nature of generated traffic based on multi-fractal scaling analysis. Our simulation results show that Retransmission Time Out mechanism plays an important role for multi-fractal nature of TCP traffic.

Web of Science

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

In the Internet, the explosive increase of the number of users requires one-to-many information distribution tools. Multicast is one of the most promising ways. One of the most important technical problems of multicast communications in the Internet is the increase of forwarding table entries with the increase of multicast groups. CIDR effectively used for unicast communications cannot be applied to multicast communications because a multicast address is a virtual address. In this paper, we propose a new multicast group aggregation technique where a shared tree aggregates several multicast groups with taking account of redundant traffic volumes in a shared tree. Simulation results show that our proposed scheme can effectively reduce multicast forwarding table entries in a backbone router with small overhead of redundant data transmission in a shared tree.

Web of Science

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

特定多数の受信ノードに対し信頼性を維持しながら同一情報を転送する信頼性マルチキャストでは,一般的にエンド-エンド間での制御によってスループットが最も低い受信ノードに合わせて情報の転送が行われる.最も低い受信ノードに合わせた場合,同一マルチキャストグループに属する他の受信ノードにおいて,自らが受信できる限界よりかなり低いスループットしか得られないという状況が生じる.これは,各受信ノードのふくそう状況が異なることが一般的にみられる,マルチキャスト通信特有の問題である.この観点からみた受信ノード間の公平性は,Intra-session Fairnessと呼ばれる.信頼性マルチキャストにおいて,エンド-エンド間での制御だけでIntra-session Fairnessを実現するのは不可能である.本論文では,Intra-session Fairessを実現するため,ネットワーク内にサーバを配置しネットワーク支援を適用することによりサーバが独自にdownlinkに対してふくそう制御を行う方式を提案し,シミュレーションにより評価を行った.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

インターネットのふくそう制御では,特定ユーザがボトルネックリンクの帯域を占有することなく送信ホストのパケット送出量を減らす必要がある.ところが,インターネットトラヒックの大部分を占めるTCP(Transmission Control Protocol)のふくそう制御では,コネクションのRTT(Round-Trip Time)によってスループットが異なるため,ボトルネックリンクの不公平帯域利用が問題となる.そこで本論文では,TCPコネクションのスループットをRTTに関係なく公平に割り当てる方法として,ネットワーク中のルータのレート通知によるTCPふくそう制御方式を提案する.本方式は,ルータを高機能化し,従来エンドーエンド間のみで行われてきたTCPふくそう制御にルータが介在することにより実現される.ルータでは,リンクを共有するコネクションの利用可能帯域を算出し,これをレートとしてエンドホストに通知する機能を実装する.エンドホストでは,ルータからの通知レートに従ってウィンドウサイズを更新する.計算機シミュレーションにより性能評価を行い,本方式の有効性を確認している.

CiNii Books

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Hosted by Prof. Biplab Sikdar

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

近年の研究により,長期依存性はトラヒック量だけではなく,パケットの伝搬遅延時間にも存在することが示されている.パケットの往復伝搬遅延時間(RTT:Round Trip Time)はTCPの性能を決定する重要なパラメータの一つであり,その特性を理解することが可能となればTCPの性能改善に寄与することができる.そこで本論文では,RTTの長期依存性がTCPの性能に与える影響について,TCPのウィンドウ制御におけるふくそう回避及びスロースタートの二つのふくそう制御フェーズに着目して,計算機シミュレーションによって性能評価を行った.まず,ふくそう回避フェーズへの影響に関する評価では,長期依存性が高く転送ファイルサイズが大きいほど,ファイルの転送完了時間に大きなばらつきが生じることを示した.次に,スロースタートフェーズへの影響に関する評価では,RTTの長期依存性がTCPのRTO(Retransmission Time Out)に与える影響について評価し,長期依存性が高いほど,実際にはロスをしていないパケットに対し再送をしてしまう不必要なタイムアウトが発生しやすいことを示した.更に,RTTの長期依存性の影響を緩和するための考察を行い,既存のRTO算出アルゴリズムに若干の変更を加えることにより,TCPの性能改善の可能性を示した.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

マルチキャスト通信をATM網において実現する技術として,IP Multicast over ATMが注目を集めている.IP Multicast over ATMを実現する方法としては,サーバ方式とメッシュ方式が提案されている.サーバ方式はサーバでのAAL 5 SDUへの再構築にある程度の遅延を要するという問題点があり,メッシュ方式は全送信者から受信グループにpoint-to-multipoint VCを設定する必要があるという問題がある.これらの問題を解決する新しい方式として,サーバにおいて複数のpoint-to-multipoint VCを用意し,複数のデータを並行して転送する際に異なるVCを用いることにより,サーバ方式において被る遅延を大幅に短縮する複数VCサーバ方式を提案する.計算機シミュレーションにより,複数VCサーバ方式とサーバ方式,メッシュ方式との性能比較を行い,複数VCサーバ方式が最も小さいデータ転送遅延時間を実現する点で有効な方式であることを検証する.

CiNii Books

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

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

Web of Science

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

In content distribution networks(CDNs), the content routing which directs user requests to an adequate server from the viewpoint of improvement of latency for obtaining contents is one of the most important technical issues. Several information, e.g. server load or network delay, can be used for content routing. Network support, e.g. active network, enables a router to select an adequate server by using these information. In the paper, we investigate a server selection policy of a network support approach from the viewpoint of which information to be used for effective server selection. We propose a server selection policy using RTT information measured at a router. Simulation results show that our proposed server selection policy in content routing selects a good server under both conditions where server latency and network delay is a dominant element of user response time. Furthermore, we also investigate about location of routers with network support bringing good performance for our proposed scheme.

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

A significant amount of wireless traffic will be carried in the Internet, and wireless connections need to share the network resources with wired connections. However, in a wireless network environment, TCP, one of the most important transport protocol of TCP/IP, suffers from significant throughput degradation due to the lossy characteristics of a wireless link. Therefore, in order to design the next generation mobile networks, it is necessary to know how much the wireless connection suffers from the degradation in comparison to the wired connection. In this paper, we discuss the fairness issue between TCP connections over wireless and wired links, and theoretically analyze throughput fairness between TCP over wireless link with ARQ(Automatic Repeat reQuest)-based link layer error recovery and TCP over error-free wired link. We validate our analysis by comparing numerical results obtained from the analysis with computer simulation ones.

Web of Science

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記述言語：英語   掲載種別：研究論文（学術雑誌）  

A content distributing algorithm based on market models for a replication distribution scheme of Web contents is proposed. In general, replication distribution is considered to effectively utilize the time and space local properties of access from a client and has been the topic of numerous studies. However, maintaining QoS (such as shortening the content obtaining time) in replicated servers when servers are in a congested state has not been considered in these studies. Since replicated servers constitute a resource shared by multiple ISPs (Internet Service Providers), the accesses need to be limited to replicated servers in case of a congested state to maintain the QoS enjoyed by each client. Thus, in this paper, a scheme for determining allocation of access to replicated servers for each content during congestion based on market models is proposed. The analytical and simulation results have confirmed that the proposed method controls the QoS in accordance with the access arrival rate to each content.

DOI： 10.1002/ecjb.1068

Scopus

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記述言語：英語   出版者・発行元：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

In this paper, we evaluate the performance of flow control mechanisms for reliable multicast under several retransmission approaches in terms of scalability. The mechanisms examined are a window-based flow control mechanism for ACK-based retransmission approaches and a rate-based How control mechanism for NAK-based retransmission approaches. Our simulation results show that the NAK-based flow control mechanism has better scalability and higher throughput than the ACK-based flow control mechanism, and the delay incurred by a NAK suppression mechanism does not affect the performance of multicast flow control.

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

In server load balancing where replicated servers are dispersed geographically and accesses from clients are distributed to replicated servers, a way of distributing the accesses from clients to an adequate server plays an important role from the viewpoint of load balancing. In the paper, we propose a new network paradigm for server load balancing using active anycast. In active anycast, an end user only sends its request to group of servers using an anycast: address. When this request arrives at an active router, it selects an adequate sur ver from the viewpoint of load balancing and changes the anycast address of a packet to the unicast address of the selected server. The decision which server is the best one from the viewpoint of server load balancing is made by an active router rather than an end user, so active anycast is a network-initiated method. Simulation results show that active anycast can accomplish efficient server load balancing, even when a small Dart of routers are equipped with active network technology.

Web of Science

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

本論文では、マルチキャスト通信を用いたアプリケーションの一つであるストリーミングサービスを取り上げ、IP Multicast over ATM上で提供する場合のネットワーク構成法について考察する。IP Multicast over ATMにおいてはIPマルチキャストアドレスから ATMアドレスへの変換を行う必要がある。これを実現する手法に Multicast Address Resolution Server(MARS)を利用したものがある。この方式ではMARSにてマルチキャストグループのメンバ情報を管理し、ホストの参加・離脱に伴うグループ内メンバの変更が生じれば該当するホストがMARSへ情報更新メッセージを送信する。ストリーミングサービスでは、番組の切換えなどの理由でMARSへの情報更新メッセージが多数発生する時間区間が存在し、この期間ではMARSに多大な負荷がかかる。本論文では、ストリーミンダサービスをMARSを用いて導入する場合を対象に、情報更新メッセージの送信頻度を解析的に求め、送信頻度と番組の特徴、並びにネットワーク規模との依存関係を明らかにする。解析結果と他のトラヒックがネットワークに与える影響を考慮した結果から、MARSへの制御トラヒック量を抑制するための適切なネットワーク設計方針を明らかにする。

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

信頼性マルチキャストにおける再送制御において, Active Network技術を適用しNAKの抑圧や再送キャッシュをネットワーク内で実現する方式の提案がなされている.本論文では, Active Network技術を信頼性マルチキャストに適用した場合に, 従来のエンド-エンドでの再送制御に比べてどの程度の改善効果が得られるかという点について, 特にネットワーク内のActive Network技術を適用したルータの割合を変化させた状況での性能評価を行った.その結果, 20%以上のルータにActive Network技術が適用された状況で, 従来の再送制御方式と比べてより多くの受信ノードを収容可能であることを確認した.また, ルータにおける再送パケットのキャッシュ機能の重要性を明らかにする.更に, Active Network技術を適用したルータの配置について考察を行い, 各MANのゲートウェイにあたるルータにActive Network技術を適用することで, 遅延特性を改善し, 冗長NAKを大幅に削減できることを明らかにした.

CiNii Books

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

One of the most important technical problems in reliable multicast protocol Is reducing redundant NAKs to avoid NAK implosion. A number of NAK suppression mechanisms have been proposed to deal with this problem. In the MBONE, which is a virtual multicast network and makes multi-point communication across the Internet feasible, the source link, the links directly connected to or very close to the source, contributes 5% packet loss. In NAK suppression mechanism, in the case of such a loss, the source link loss, all receivers suffer the same packet loss and NAK suppression mechanism does not work effectively. In this paper we propose a Reliable Multicast Protocol Applied Local FEC, called the local FEC, where the source link loss is recovered with FEC applied locally only to the source link. To investigate performance of the local FEC, it is compared with reliable multicast protocol with conventional NAK suppression mechanism. Our simulation results show that the local FEC outperforms the NAK suppression protocol from the view point of scalability and wasted network bandwidth.

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

Recent measurement showed that self-similar nature is found not only in network traffic volume but also round trip packet delay. In this paper, we discuss three issues of the self-similarity of round trip time (RTT), which is one of the most important parameters to determine TCP throughput performance. First, we discuss the origin of the packet delay self-similarity. A recent study anticipated that the queueing delay of self-similar traffic is the reason for packet delay self-similarity. With computer simulation, we evaluate the correlation between traffic and RTT self-similarity. Next, we investigate the impact of RTT self-similarity on TCP throughput performance. Computer simulation results show that RTT self-similarity gives high variability to file transfer time. Finally, we investigate the impact of RTT self-similarity on RTO (Retransmission Time Out). We discover that the bigger the Hurst parameter of RTT is, the more frequent unnecessary timeouts occurs. Furthermore, we propose a new RTO calculation algorithm to improve these unnecessary timeouts.

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

Recent measurement showed that self-similar nature is found not only in network traffic volume but also round trip packet delay. In this paper, we discuss three issues of the self-similarity of round trip time (RTT), which is one of the most important parameters to determine TCP throughput performance. First, we discuss the origin of the packet delay self-similarity. A recent study anticipated that the queueing delay of self-similar traffic is the reason for packet delay self-similarity. With computer simulation, we evaluate the correlation between traffic and RTT self-similarity. Next, we investigate the impact of RTT self-similarity on TCP throughput performance. Computer simulation results show that RTT self-similarity gives high variability to file transfer time. Finally, we investigate the impact of RTT self-similarity on RTO (Retransmission Time Out). We discover that the bigger the Hurst parameter of RTT is, the more frequent unnecessary timeouts occurs. Furthermore, we propose a new RTO calculation algorithm to improve these unnecessary timeouts.

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

In reliable multicast communications, retransmission control plays an important role from the viewpoint of scalability. Previous works show that the implosion of control packets, e.g. ACKs or NAKs, degrades the total performance of reliable multicast communications. Local recovery which enables receivers receiving a packet successfully to initiate recovering a lost packet may have the possibility to solve this scalability problem. This paper presents the performance evaluation of local recovery caused LS grouping receiving nodes in reliable multicast communication. There seems to be many features dominating the performance of local recovery, the number of nodes in a group, the shared loss occurring simultaneously at multiple receivers and so on. When the number of receivers in a group increases, the geographical expansion of a group will degrade the delay performance of the receivers. In a configuration where most nodes in a local-reca cry group suffer from shared loss, the failure of local recovery degrades the total performance. Our simulation results under a hierarchical network topology like the real Internet show that a local-recovery group configuration with two-adjacent MANs grouping performs well.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

Webにおけるコンテンツの複製分散配置方式において,市場モデルに基づいたコンテンツ分散配置アルゴリズムを提案する.一般にcachingよりも複製分散配置(replication)のほうがクライアントからのアクセスの時間的・空間的局所性を有効に活用できるとされ,数多くの研究がなされているが,サーバがふくそう状態に陥った場合における複製サーバにおけるQoS(コンテンツ取得時間の短縮等)の維持については議論されていない.複製サーバは複数のISP(Internet Service Provider)等によって共有される資源であるため,ふくそうに陥った場合には複製サーバへのアクセスを制限し,各クライアントが享受するQoSを維持する必要がある.よって本論文では,市場モデルを用いてふくそう時における各コンテンツごとの複製サーバへのアクセス割当てを決定する方法を提案する.解析及びシミュレーションの結果,提案方式は,各コンテンツへのアクセス到着率に応じてQoSに差を生じさせる制御を行っていることを確認した.

CiNii Books

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

In this paper, we evaluate the performance of flow control schemes for reliable multicast under several retransmission approaches in terms of scalability. The schemes examined are a window-based flow control scheme for ACK-based retrasnmission approaches and a rate-based flow control sheme for NAK-based retransmission approaches. Our simulation results show that the NAK-bascd flow control scheme has better scalability and higher throughput than the ACK-based flow control scheme, and the delay incurred by a NAK-suppression mechanism does not affect the performance of multicast flow control.

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

One of the most important technical problems in reliable multicast protocol is to reduce redundant control packets and retransmitted packets. An approach for reducing them by making use of newly-invented network architecture, the active network technology, has been proposed. In the paper we compare a network-initiated approach, i.e. reliable multicast protocol with network support, and conventional end-to-end approaches. These are compared under more realistic situation such as part of routers are equipped with the newly-invented active network technology. Our simulation results show that network-initiated approach with 15% active routers outperforms conventional end-to-end approaches from the viewpoint of scalability. We also investigate desirable location of active routers with sophisticated network topology model, Tiers model, which reflects hierarchical structure of the Internet. Our simulation results suggest that at first active routers should be implemented in I VAN because delay performance can be improved with small number of active routers. After active routers are broadly implemented in WAN, policy for location of active router should be switched to MAN with expecting decrease of redundant NAK transmission inside a network.

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

This paper presents a performance evaluation of NAK-based reliable multicast communication protocols operating in an environment where end-to-end delay are heterogeneous. In the case of heterogeneous delay, performance of a timer-based retransmission control scheme may become worse. We show that a counter-based retransmission control scheme works well in the case of heterogeneous transmission delay. We also compare two NAK-based protocols and show that a NAK-multicasting protocol outperforms a NAK-unicasting protocol from the viewpoint of scalability even when delays are heterogeneous.

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

In reliable multicast communications, transmission rate of a sender should be restricted to the node capability of the lowest node in order to support reliable transmission of packets to all receivers in a multicast group. Even a node of high capability should receive packets at lower rate when there are lower capability nodes in a corresponding multicast group, In reliable multicast communications, heterogeneity of node capability degrades total performance of a multicast group. In the paper, we present the layered multicast group construction, which is one of technical solutions for performance degradation caused by heterogeneity. The basic concept of the layered multicast group construction is to divide a multicast group into multiple subgroups and order them based on node capability. This reduces diversity of node capability inside each sugbroup, which improves delay performance of a whole multicast group. We investigate optimal construction of the layered multicast group construction, i.e. optimal dividing points of subgroups. Numerical examples show that average delay performance is remarkably improved by the layered multicast group construction compared with a conventional single multicast group construction and performance improvement obtained by two or three subgroups is sufficient for practical use.

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

In reliable multicast communications, transmission rate of a sender should be restricted to the node capability of the lowest node in order to support reliable transmission of packets to all receivers in a multicast group. Even a node of high capability should receive packets at lower rate when there are lower capability nodes in a corresponding multicast group, In reliable multicast communications, heterogeneity of node capability degrades total performance of a multicast group. In the paper, we present the layered multicast group construction, which is one of technical solutions for performance degradation caused by heterogeneity. The basic concept of the layered multicast group construction is to divide a multicast group into multiple subgroups and order them based on node capability. This reduces diversity of node capability inside each sugbroup, which improves delay performance of a whole multicast group. We investigate optimal construction of the layered multicast group construction, i.e. optimal dividing points of subgroups. Numerical examples show that average delay performance is remarkably improved by the layered multicast group construction compared with a conventional single multicast group construction and performance improvement obtained by two or three subgroups is sufficient for practical use.

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

One of the objectives of ABR traffic control is a fair allocation of transmission rate to every connection. Algorithms which have been proposed can achieve the fair allocation of transmission rate in persistent traffic. even though the connections have different number of hops. In bursty traffic, however, the transmission rate during the burst period, called average burst throughput, may become degraded when the number of hops increases. In this paper, we evaluate the average burst throughput characteristics of ERICA switch algorithm and investigate the effect of the number of hops. We propose a new bandwidth allocation method which can improve the performance degradation by taking the number of hops into account. The effectiveness of our method is evaluated by computer simulation.

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

One of the objectives of ABR traffic control is a fair allocation of transmission rate to every connection. Algorithms which have been proposed can achieve the fair allocation of transmission rate in persistent traffic. even though the connections have different number of hops. In bursty traffic, however, the transmission rate during the burst period, called average burst throughput, may become degraded when the number of hops increases. In this paper, we evaluate the average burst throughput characteristics of ERICA switch algorithm and investigate the effect of the number of hops. We propose a new bandwidth allocation method which can improve the performance degradation by taking the number of hops into account. The effectiveness of our method is evaluated by computer simulation.

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

DQRUMA ( Distributed-Queueing Request Update Multiple Access) protocol has been proposed as an access protocol for the wireless ATM Local Area Networks. DQRUMA protocol is useful to transmit fixed-length pac packets (e.g. ATM cells). However, it cannot be applied to multimedia environment because it does not include any access control policy for multimedia traffic. In the paper, we propose a slot assignment scheme of DQRUMA protocol in wireless ATM LAN which supports integrated multimedia traffic with different service requirements. In this scheme we can allocate network resources according to the service requirements of each medium because the base station assigns Transmit-Permission flexibly according to the features of each medium.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

処理能力の異なるさまざまな計算機をネットワークを介して接続して構成される不均質な分散システムにおいて, 統計的な負荷情報を用いて計算機能力に応じた負荷分散を自律的に行う動的負荷分散制御方式を提案する.本方式では, プロセスが投入された計算機を優先的に選択する従来の方式とは異なり, システム内の全計算機のなかから公平に処理計算機を選択する.このことにより, ユーザが投入する計算機によって性能が依存せず, 従来考慮されることが少なかったユーザ間の公平性を高めている.また, 負荷の瞬時情報と統計的情報を統合して用いることにより, 効率の良い負荷分散制御に不可欠な, 瞬時負荷の均衡と計算機処理能力に対応した負荷の割当ての自律的な制御を可能としている.

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

In reliable multicast communications, lost information due to packet loss should be re-multicasted. NAK-based retransmission scheme is said to be effective for scalable reliable multicast communications because it can avoid implosion of control packet. When the source sends packets faster than receiver's capacity, packet loss due to buffer overflow occurs constantly at corresponding receivers. With this constant packet loss, implosion of control packet degrades total throughput seriously even in the case of NAK-based retransmission scheme. In order to prevent this kind of constant packer loss at receivers, flow control scheme should be implemented for reliable multicast communications. Iii the paper we propose a new flow control scheme, which is suitable for NAK-based retransmission scheme. From the viewpoint of flow control, receiving a NAK intends that transmission rare is too high. However; in multicast communications, multiple NAKs may be generated for a corresponding packet. If simply receiving a NAK decreases the transmission rate, rate may be decreased excessively In the paper, log information of transmission rate stored at the sender is proposed in order to prevent the transmission rate from decreasing excessively. Performance evaluation by computer simulation shows that our proposed scheme notably improves through put performance compared with the case of no flow control.

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

In reliable multicast communications, lost information due to packet loss should be re-multicasted. NAK-based retransmission scheme is said to be effective for scalable reliable multicast communications because it can avoid implosion of control packet. When the source sends packets faster than receiver's capacity, packet loss due to buffer overflow occurs constantly at corresponding receivers. With this constant packet loss, implosion of control packet degrades total throughput seriously even in the case of NAK-based retransmission scheme. In order to prevent this kind of constant packer loss at receivers, flow control scheme should be implemented for reliable multicast communications. Iii the paper we propose a new flow control scheme, which is suitable for NAK-based retransmission scheme. From the viewpoint of flow control, receiving a NAK intends that transmission rare is too high. However; in multicast communications, multiple NAKs may be generated for a corresponding packet. If simply receiving a NAK decreases the transmission rate, rate may be decreased excessively In the paper, log information of transmission rate stored at the sender is proposed in order to prevent the transmission rate from decreasing excessively. Performance evaluation by computer simulation shows that our proposed scheme notably improves through put performance compared with the case of no flow control.

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

A growing number of network applications require the use of a reliable multicast protocol to disseminate data from a source to a potentially large number of receivers. This paper presents an analytic performance analysis of the packet delay incurred under three generic sender-and receiver-initiated approaches towards reliable multicast. We focus on the host processing requirements of these protocols and derive expressions for average time between the initial arrival of a packet at a sender and its correct reception at a randomly chosen receiver Our numerical results indicate that a NAK-based protocol that limits NAK generation by intentionally and randomly delaying NAK packets can achieve substantially higher throughput than the other two protocols examined, and can do so without suffering an appreciable higher delay over a range of system parameters.

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

In this study, we investigate the problem of accommodating multicast traffic in ATM networks, with emphasis on the virtual path (VP) environment. We propose a network structure called ''virtual copy network'' that is suitable for multicast communication and connection setup scheme, taking into account the VP environment. With our technique, we can expect statistical multiplexing gain in accommodating multiple multicast traffic streams over a VP. Also, we propose two multicast routing algorithms (the CNR-LH algorithm and the improved CNR-LH algorithm) for an ATM network environment. In these algorithms, in addition to determining an adequate route, the exact corresponding VPs on the route are also obtained. We examine the efficiency of the algorithms by demonstrating their basic characteristics using computer simulations.

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

The message level performance of error controls in data communication on ATM network is analyzed. Three layers, ''a cell''(a unit of transmission), ''a block''(a unit of error controls) and ''a message''(a unit of transmission of user level) are considered. The error controls treated in this paper are GBN (Go-Back-N) and FEC+GBN. The cell loss process is assumed to be the two stale Markov chain considering the cell loss process in ATM networks. Numerical results show that (1) the improvement of the message forwarding delay is saturated in some environments when the interface rate becomes high, (2) FEC is efficient when the burstiness of the cell loss process is small, the message length is large and the interface rate is high.

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

The optical ATM transport networks are the key technology for B-ISDN which integrates wide variety of communication services. In a photonic ATM switch, electronically operated switching control drastically limits the total throughput. With this bottleneck of the control speed of the switch compared to the cell transmission speed, if the unit of switching operation can be made longer, the system throughput will be improved. This paper proposes the optical backbone network configuration to obtain traffic concentrating effect to construct a large switching unit called a multi-cell. In the backbone network applied the concept of virtual path set (VPS), a multi-cell is constructed by cells from all of ATM switches in a regional network connected with each own's cross connect. The multi-cell format in the case of two different network models is investigated in this paper. The average delay and average idle cells per multi-cell in the multi-cell MUX of an optical cross connect are evaluated by computer simulation as the performance of the multi-cell transfer scheme in the backbone network. Simulation results show that the multi-cell transfer scheme can be operated efficiently with traffic load of more than 0.5. This paper also proposes the configuration of multiplexing and demultiplexing module to assemble and disassemble the multi-cell.

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

This paper considers the shared output buffer-type switching system as an architecture suited to the large-scale ATM switching system. An architecture is proposed, which can solve the fair throughput assignment problem among the ports belonging to the same shared memory, which is a problem in sharing the output buffer. The proposed architecture realizes fair access to the destination port contained in the specified shared buffer, by sort processing and the address attachment, without using the random access contention unit to the buffer in the knockout switch.
For the proposed architecture and the conventional shared output buffer-type knockout switching system, the performances are compared by deriving the cell loss probability due to the output contention for each port. As a result, it is verified that the proposed architecture is better in terms of the realized cell loss probability.
The performance of the switching system is evaluated by a computer stimulation when various buffering schemes are applied to the shared output buffer. Then it is shown that the fair throughput assignment among the considered output ports can be realized by combining the proposed architecture with an adequate buffering scheme.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

本論文ではATMに親和性の高い離散時間環境において,到着過程だけでなく,サービス過程にも相関性のある有限容量単一サーバ待ち行列としてD-BMAP / D-MAP / 1 / Nを解析し,定常状態確率分布,廃棄率,および稼働期間,システム滞留時間,退去間隔の確率母関数を得ている.また本待ち行列モデルの応用例としてバッファドリーキーバケットを考え,数値例を示している.リーキーバケットに代表されるような,ATM網でのトラヒック制御手法の中には,そのセル送出規律に相関性を有するものがあり,これらのシステムを待ち行列モデルにより性能評価する場合は,到着過程だけでなくトラヒック制御によるサービス過程の相関性も考慮する必要がある.本待ち行列モデルでは,到着過程にD-BMAPを,サービス過程にD-MAPを仮定している.Blondiaらによって提案されたD-BMAPは,D-MAPを集団到着に拡張したものであり,その重畳過程もまたD-BMAPで表現できる.またD-MAPは,MMBP等の相関性を有する到着過程だけでなく,リーキーバケットなどのトラヒック制御システムのセル送出規律も表現できる.従って,サービス過程をD-MAPとすることにより,到着過程だけでなくトラヒック制御によるサービス過程の相関性も反映させた性能評価が可能となる.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

本論文では,大規模ATM交換機に適したアーキテクチャとして共有出力バッファ型交換機を取り上げ,出力バッファの共有化の際に問題となる,同一共有メモリに属する端子間の不公平なスループット配分を解消するアーキテクチャを提案する.本アーキテクチャはノックアウトスイッチにおけるランダムなバッファへのアクセス競合部を使用せず,ソーティング処理とアドレス付加処理により特定の共有バッファに収容される目的端子への公平なアクセスを実現する.本論文では,提案アーキテクチャならびに従来のノックアウトスイッチで出力バッファを共有化したタイプに対し,出力競合に関する端子別のセル廃棄率を導出し特性比較を行う.その結果,提案アーキテクチャが優れたセル廃棄率性能を実現することを確認する.更に,計算機シミュレーションにより共有出力バッファに各種バッファリング方策を施した場合の交換機の特性を評価し,本アーキテクチャに適切なバッファリング方策を併用することにより目的出力端子間のスループット配分を公平にすることが可能であることを示す.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

本論文ではATM環境を対象としたVP(Virtual Path)環境における同報型トラヒック収容問題について検討を行い,実現可能と思われるマルチキャスト通信に適した網構成法を提案する.本手法ではVPへ複数の同報型トラヒックを収容するため統計多重効果を期待することができ,かつ同報型トラヒック収容時に必要なマシンコストを低減化することが可能となる.また,ATM網環境における同報型経路選択アルゴリズム(CNR法,改良型CNR法)を提案する.計算機シミュレーションにより本アルゴリズムの基本特性を示すことでその有効性を検証する.

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

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記述言語：英語   出版者・発行元：IEICE-INST ELECTRONICS INFORMATION COMMUNICATIONS ENG  

A new transfer mode and a switching architecture which can support loss free and no delay jitter service class with shorter switching delay compared with ''stop and go queueing scheme'' is proposed. This scheme combines ATM scheme with hierarchical STM framing concept.

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

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

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

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

In this paper, we propose a new network organizing method for packet radio networks, a layered self-organizing method. In the layered self-organizing network, whole service area is divided into multiple sub-areas and one base station is settled in each sub-area. Communication links are settled in shorter time than the conventional self-organizing method. We evaluate the network organizing performance of the method by using simulations.

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

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記述言語：英語   掲載種別：研究論文（学術雑誌）   出版者・発行元：IEEE-INST ELECTRICAL ELECTRONICS ENGINEERS INC  

In this paper, we propose a new ATM switch architecture which uses the broadcasting transmission medium for transmission of cells from input ports to output ports. Cell transmission and its control are separated completely and cell transmission control, i.e., header operation, is executed before cell transmission (control ahead). By this operation, cell transmission and its control can be executed in pipeline style. This enables high-speed cell exchange and makes transmission control easier. One of the essential problems for ATM switches which use the broadcasting transmission medium is high-speed operation of the transmission medium. We analyze the switch fabric performance according to its switching speed. Numerical results show that the ATM switch proposed here shows good cell loss performance even when its switching speed is restricted, provided that switch utilization is below 1. Furthermore, we show some extension for the switch leads to robustness against bursty traffic.

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

Multicast routing problem is one of the essential problems for supporting multicast and broadcast communication service which is the most important service of the multimedia information networks. Multicast routing is the problem of finding out an adequate path which connects one source node and more than one destination node, i.e. a tree shaped path. In packet type networks, a packet for multicast communication should go through the tree shaped path as making its copies at a branching node for efficient use of network resources. However, concentration of packet copy operations at a particular node leads to performance degradation of other calls which go through this node. In this paper we propose two multicast routing algorithms which distribute packet copy operations through whole nodes in the multicast path; a link added type algorithm and a loop constructed type algorithm. Both algorithms, at first, find out an approximate solution for minimum cost path, and avoid concentration of packet copy operation at a little sacrifice of total cost in the path. Computer simulation results show that these algorithms can decrease the burden of packet copy operation per a node at the sacrifice of increase in average distance(cost) of a source-destination pair but the sacrifice of total cost is very small.

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記述言語：日本語   出版者・発行元：電子情報通信学会通信ソサイエティ  

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記述言語：日本語   出版者・発行元：電子情報通信学会  

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記述言語：日本語   出版者・発行元：電子情報通信学会  

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記述言語：日本語   出版者・発行元：電子情報通信学会  

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記述言語：日本語   出版者・発行元：電子情報通信学会  

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記述言語：英語  

This paper first investigates how a network operates when multiple receivers download content simultaneously in content-centric networking (CCN) when the receivers' downloading speeds differ. The results indicate that the performance of the download completion time of a faster user degrades excessively due to a decrease in the cache-hit rate in the router. Based on the investigation, this paper proposes a novel in-network caching method for simultaneous download from multiple receivers in CCNs. The proposed method keeps cached data packets in a router until slower receivers download the data, in order to prevent slower users from directly downloading data from the content provider. We conduct computer simulations and confirm the effectiveness of the proposed method. We show that the proposed method can improve the download completion time performance in the situation where multiple receivers download content at different speeds in CCN.------------------------------This is a preprint of an article intended for publication Journal ofInformation Processing(JIP). This preprint should not be cited. Thisarticle should be cited as: Journal of Information Processing Vol.25(2017) (online)DOI http://dx.doi.org/10.2197/ipsjjip.25.164------------------------------This paper first investigates how a network operates when multiple receivers download content simultaneously in content-centric networking (CCN) when the receivers' downloading speeds differ. The results indicate that the performance of the download completion time of a faster user degrades excessively due to a decrease in the cache-hit rate in the router. Based on the investigation, this paper proposes a novel in-network caching method for simultaneous download from multiple receivers in CCNs. The proposed method keeps cached data packets in a router until slower receivers download the data, in order to prevent slower users from directly downloading data from the content provider. We conduct computer simulations and confirm the effectiveness of the proposed method. We show that the proposed method can improve the download completion time performance in the situation where multiple receivers download content at different speeds in CCN.------------------------------This is a preprint of an article intended for publication Journal ofInformation Processing(JIP). This preprint should not be cited. Thisarticle should be cited as: Journal of Information Processing Vol.25(2017) (online)DOI http://dx.doi.org/10.2197/ipsjjip.25.164------------------------------

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記述言語：日本語   出版者・発行元：電子情報通信学会  

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記述言語：日本語   出版者・発行元：電子情報通信学会  

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

CiNii Books

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記述言語：日本語   出版者・発行元：電子情報通信学会  

CiNii Books

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記述言語：日本語   出版者・発行元：電子情報通信学会  

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

主要ネットワークサービスがコンテンツ取得となっている現状を踏まえ,コンテンツ名を指定することでネットワークが主体的にコンテンツ発見,コンテンツ転送機能を提供するコンテンツ指向ネットワークが注目されている.ユーザにとってはどこからコンテンツが得られようが関心がないというコンテンツ指向の概念に基づき,同一コンテンツを提供するコンテンツサーバが複数ある場合,ルータにおけるルーチングテーブルは同一コンテンツ名に対し複数のエントリをもつことが可能である.本稿では,ルータにおいてInterestパケットとそれに対応して返送されるデータパケットのRTTを計測し,これをもとに複数エントリの中から確率的に転送先インタフェースを選択する,確率的コンテンツ指向ルーチングを提案する.提案方式では,計測RTTに基づきインタフェース選択確率を適応的に変化させ,サーバ負荷分散及びネットワーク輻輳の回避を目指す.簡易モデルを用いたシミュレーション評価により,提案手法によりサーバ負荷分散が実現でき,コンテンツ取得遅延が改善されていることを示す.

CiNii Books

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記述言語：日本語   出版者・発行元：国立研究開発法人 情報通信研究機構  

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

近年,東日本大震災の影響を受け,災害時における通信技術への関心が高まっている.既存の通信インフラ設備が地震等の大規模な災害によって破損した場合,DNS等の機能が利用できなくなり,サーバへのアクセスや安否情報等のコンテンツ取得が困難となる.本研究では,サーバのIPアドレスが不明でもコンテンツ取得が可能であるCCN(Content-Centric Network)に着目し,災害時に分断されたネットワークへの適用を試みる.有線網を前提としたCCNアーキテクチャでは分断ネットワークを想定していないため,冗長なコンテンツ要求の再送が発生する.再送の頻発は,限られた無線リソースの浪費,衝突によるパケット喪失,電力消費などの問題を引き起こす.本稿では,災害時の分断されたネットワークにおけるCCNの再送問題を解決するコンテンツ要求再送抑制方式を提案する.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

キャッシュネットワークにおいて,ユーザからサーバに至る経路上のキャッシュ以外のキャッシュコンテンツを有効に活用する手法として,キャッシュへの誘導情報を用いるBreadcrumbsが提案されている.Breadcrumbsにおいては,キャッシュコンテンツが時間経緯とともに置き換わる可能性を考慮し,最新の誘導情報のみを保持している.人気コンテンツに対しては,ある程度古い誘導情報でもキャッシュコンテンツが残っている可能性が高いことに着目し,筆者らは複数Breadcrumbsを提案している.本稿では,この複数Breadcrumbsのコンテンツ発見機能を活用し,発見した複数コンテンツから最も高いスループットを与えるものを選択するThroughput Sensitive選択予法を提案する.性能評価により,提案方式は人気コンテンツのコンテンツ流通を効率化することで,ネットワーク全体の資源の効率利用が可能となり,不人気コンテンツについてもスループット改善が得られる優れた方式であることを明らかにする.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

インターネット上を流れるトラヒックのかなりの部分を占めているコンテンツ配信は,CDNなどの技術により,コンテンツがどこから転送されているのかには全く関心がない形でのコンテンツオリエンテッドなサービスとして展開されている.コンテンツ流通を支えるインフラであるインターネットは,ホスト間通信を基本としたロケーションオリエンテッドなアーキテクチャを用いている.このように,ユーザに展開されているコンテンツ配信サービスはコンテンツオリエンテッドなものであり,その下の情報転送基盤としてのインターネットはロケーションオリエンテッドな設計となっており,両者に乖離がみられている.ネットワークアーキテクチャをコンテンツオリエンテッドなものとし,この乖離を解決しようとする取り組みとして,コンテンツオリエンテッドネットワークの研究が近年活発となっている.本稿では,このコンテンツオリエンテッドネットワークの最新研究動向について,そのアーキテクチャとして代用的なNDN(Named Data Networking)を紹介したのちに,ルータやキャッシュに関する興味深いアプローチを紹介する.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

インターネット上を流れるトラヒックのかなりの部分を占めているコンテンツ配信は,CDNなどの技術により,コンテンツがどこから転送されているのかには全く関心がない形でのコンテンツオリエンテッドなサービスとして展開されている.コンテンツ流通を支えるインフラであるインターネットは,ホスト間通信を基本としたロケーションオリエンテッドなアーキテクチャを用いている.このように,ユーザに展開されているコンテンツ配信サービスはコンテンツオリエンテッドなものであり,その下の情報転送基盤としてのインターネットはロケーションオリエンテッドな設計となっており,両者に乖離がみられている.ネットワークアーキテクチャをコンテンツオリエンテッドなものとし,この乖離を解決しようとする取り組みとして,コンテンツオリエンテッドネットワークの研究が近年活発となっている.本稿では,このコンテンツオリエンテッドネットワークの最新研究動向について,そのアーキテクチャとして代用的なNDN (Named Data Networking)を紹介したのちに,ルータやキャッシュに関する興味深いアブローチを紹介する.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

データセンタにおける2層輻輳制御として,各スイッチからのフィードバックをもとに送信レートを決定するQCN(Quantized Congestion Notification)が提案されている.データセンタネットワークでマルチキャスト通信を行う際には,木状経路の複数地点で輻輳が発生する可能性があり,これら複数ボトルネックからのフィードバックによる過剰なレート抑制が発生する.この問題に対処する方式として,筆者らはQCN/BS(QCN with Bottleneck Selection)を提案している.本稿では,マルチキャスト通信を対象に提案したQCN/BSをユニキャスト通信で複数ボトルネックリンクが存在する状況に適用した場合,予想に反して複数ボトルネックを通るフローの送信レートが,他のフローに比べ過度に抑制されることを明らかにする.また詳細な評価結果を通して,その原因が複数ボトルネックリンクを通るフローの見る待ち行列長が,他のフローに比べ若干大きいことにあることを示す.また,筆者らがQCNの公平性改善手法として提案したAdaptive BC_LIMITを適用することで,この不公平性がある程度改善できることを示す.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

無線マルチホップネットワークにおいて,ゲートウェイノード(GW)を介して外部サーバからコンテンツをダウンロードする際には,クエリのGWへの集中,ならびにコンテンツダウンロードトラヒックがGWを起点とすることから,スループットが大きく劣化することが懸念される.この問題に対する一つの解決策として,ネットワーク内の中継ノードにキャッシュを設ける,キャッシュネットワークの適用がある.キャッシュネットワークは,クエリがコンテンツサーバに向かう途中のdefault-path上のキャッシュのみが活用できるという制限がある.この制限を解消する手法として,インネットワーク誘導方式であるBreadcrumbsが提案されている.本稿では,無線マルチホップネットワークにこのBreadcrumbs方式を導入した際の性能評価を行い,トータルとしてのネットワーク内コンテンツトラヒック量を大きく改善すること,高人気コンテンツのみならず低人気コンテンツに対してもスループットが改善する効果があること,を詳細な検討を通して明らかにする.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

近年のコンテンツ流通トラヒックの増大に対応する手法として,ネットワーク内のルータにキャッシュを配備する,キャッシュネットワークがある.キャッシュネットワークにおいて,ユーザからサーバに至る経路上のキャッシュ以外のキャッシュコンテンツを有効に活用する手法として,キャッシュへの誘導情報を用いるBreadcrumbs方式が提案されている.Breadcrumbs方式の普及にはルータのリプレースが必要であり,その普及過程においては一部のルータにのみ本方式が普及する部分普及状況が発生する.本稿では,階層モデルを用いた評価により,Breadcrumbsの普及過程において誘導効果をもたらす要因を分析し,その分析に基づく戦略的普及方針として,上位優先普及と下位優先普及の二つの方策を提案する.詳細な性能評価により,上位優先普及がトラヒック集約効果によるコンテンツ要求誘導数の増加,さらに下位Trailの効率的運用をもたらす点で,優れた戦略的普及方策であることを明らかにする.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

データセンタネットワークにおける2層輻輳制御であるQCN (Quantized Congestion Notification)においては,ネットワーク内のスイッチが確率的にフィードバック情報を返送しレート制御を行う.筆者らは,QCNにおいてフロー数が増加した際に,輻輳を通知するフィードバックを受信できない送信ノードが発生し,キュー長が大きく変動することを明らかにした.また,これを解決する手法として,QCNに遅延ベースのエンドエンド輻輳制御を組み合わせるQCN/DCをすでに提案している.本稿では,筆者らの提案方式を実用に供するために必要である,1)フロー参入時にみられたリンク利用率の低下問題の解決,2)QCNと遅延ベース輻輳制御を切り替える閾値を適応的に設定する手法の確立,について検討を行う.具体的には,前者に対してはフロー参入時にみられた急激なレート減少を回避するために,遅延ベース輻輳制御フェーズにおいて適切な送信レートを選択する方式を提案する.後者に対しては,各送信ノードにおいてACKにより観測したRTTに基づき適応的に閾値を計算する手法を提案する.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

データセンタネットワークは,高速イーサネットを用い距離が短い環境で用いられることから,RTTはμsecオーダーである.データセンタに適した輻輳制御として,ネットワーク内のスイッチがフィードバック情報を確率的に返送しレート制御を行うQCN(Quantized Congestion Notification)が提案されている.QCNは,RTTが極めて小さいデータセンタにおいてはスイッチにおけるキューイング遅延の影響が大きいことから,キュー長を低く抑えることを主眼に開発されている.さらに,高スループットを実現するためにキュー長をある一定値に安定させ極力0としない方向にチューニングされている.ただし,ボトルネックを通るフロー数が増加した場合には,輻輳を通知するフィードバックを受信しない送信ノードが発生し,キュー長に大きな変動が生じる.本稿では,フロー数増加時のこのような問題に対処する方法として,QCNに遅延ベースのエンドエンド輻輳制御を組み合わせフロー数増加時にフィードバックが返送されない状況でも適切に輻輳制御を働かせる全く新しい手法を提案する.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

近年,高品質の動画や音楽といったリッチコンテンツの流通により,コンテンツサーバヘの負荷増大が問題となっている.その解決策の一つに,ネットワーク内にコンテンツを格納するキャッシュを配置したキャッシュネットワークがある.サーバヘのdefault path 上のキャッシュ以外のキャッシュを有効利用する手法として,コンテンツの存在方向を足跡として残すBreadcrumbs方式が提案されている.Breadcrumbsにおいては,足跡情報として最新のコンテンツダウンロード方向を保持するものとしている.ただし,ある程度新しい情報であれば最新でない古い情報も積極的に利用することで,コンテンツ発見の幅が広がる可能性がある.本稿では,Breadcrumbs情報として,最新以外のものも一定時間保持する手法である複数Breadcrumbsに対し,具体的方式動作の検討と基礎評価を行う.その結果,複数Breadcrumbsによる複数方向へのクエリ誘導を許容することで,複数箇所からのダウンロードに伴うサーバダウンロード負荷の低減効果が期待できることを明らかにする.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

近年,動画などのリッチコンテンツに牽引され,インターネットのトラヒック量は増大の一途をたどっている.全体の量的な増大も然ることながら,時間変動も大きくなっており,ピークとオフピーク時のトラヒック量の差が非常に大きく,全体としてのリンク帯域の有効利用が図れない状況にある.本稿では,ネットワーク内のルータに大規模なストレージを配することで,要求発生からコンテンツ視聴までの時間差を許容するトラヒックが,リンク利用率の低い時間帯およびリンクを有効に利用する新しい枠組みとして,Storage Aware Routing(SAR)を提案する.また,SARの定式化を行ったうえで線形計画法を用いた基礎評価を行い,SARではリンク利用率を空間的,ならびに時間的に平滑化することが可能であることを示す.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

コンテンツを取得することを目的とするコンテンツオリエンテッドサービスにおいては,どこからコンテンツを取得するかという点は重要ではない.さらには,P2Pにおいてチャンクごとに送信ホストが異なることからも分かるように,同一ファイルを取得する先が複数であってもよいという特徴をもつ.この点に着目し,本稿では複数送信ホストから単一受信ホストへ情報を並行転送する際に適した輻較制御を検討する.このような状況では,複数フローの情報を統一的に管理できるのは受信ホストのみであることから,受信駆動型輻較制御を開発する.複数送信ホストから並行転送する際,同一ボトルネックリンクを通るフローに対して統合的に制御することで他のTCPフローに悪影響を及ぼさないこと,異なるボトルネックリンクを通るフローに対しては独立に制御することで効率的にネットワーク資源を利用すること,の実現を目指す.基礎評価により,本稿で提案する受信駆動型輻較制御がこれらの要件を満たすことを示す.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

ユーザがコンテンツを取得するには,まずコンテンツがどこにあるのかを発見するコンテンツ発見フェーズを経たうえで,さらにコンテンツをその存在箇所からユーザまで転送するコンテンツ転送フェーズを経る必要がある.本研究では,これらコンテンツ発見フェーズとコンテンツ転送フェーズの双方に対し,ユーザにとってはコンテンツがどこから得られるのかには関心がなく,どのように効率的にコンテンツを取得できるのかという観点が重要であるという考えに基づき,負荷分散やトラヒック分散の観点で効率的なコンテンツ発見と,ロバスト性と効率を実現するコンテンツ転送を実現する.これにより,新世代コンテンツ配信アプリケーションが具備すべきSpace decoupling(コンテンツ提供者とコンテンツ要求者間のロケーションオリエンテッドな強い関係を緩和し,誰からコンテンツを得るのか,どこからコンテンツを得るのかという点には制約を設けない機能)に対し,ネットワーク資源の効率的運用という観点を取り入れた形での実現を目指す.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

無線マルチホップネットワークにおいて,外部の有線ネットワークへのゲートウェイが複数存在する場合,各ホストは一つをデフォルトゲートウェイとして選択する必要がある.最短ホップ数となるゲートウェイを選択する方法などが提案されているが,トラヒックの地理的分布が不均一である場合には,ゲートウェイ間の負荷不均衡による性能劣化が懸念される.著者らは,先にゲートウェイにおける負荷均衡と周辺無線干渉を考慮した,ゲートウェイ選択方式を提案した.本稿では,さらに一般的な評価モデルを用い,負荷集中を回避することで制御パケットを抑制できること,負荷集中を回避できるエリアだけでなく回避によりかえって負荷が増えるエリアにおいても制御パケットの抑制効果によって正規化スループットが改善すること,を新たに明らかにする.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

従来のトラヒックエンジニアリングは,送受信ルータ間のトラヒックをもとに経路設計を行うという,ロケーションオリエンテッドな設計手法である.本稿では,ビデオ転送などリッチコンテンツの流通トラヒックがネットワーク内の多くを占める状況で,コンテンツはどこから取得しても問題ないというコンテンツオリエンテッドな特徴を利用した,全く新しいコンテンツ指向型トラヒックエンジニアリングとしてCAR(Content Aware Routing)を提案する.CARにおいては,コンテンツ要求ルータが要求するコンテンツ転送トラヒックを把握したうえで,同一コンテンツを提供する複数サーバからの転送経路設計を行う.CARの定式化を行ったうえで線形計画法による性能評価を行い,CARはリソース使用量の若干の増加はみられるものの最大リンク利用率を非常に低く抑え,余剰帯域を確保するという点で有効なトラヒックエンジニアリング手法であることを示す.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

近年,動画配信において,サーバの負荷軽減を図りつつ大規模配信を可能とする技術としてP2Pを用いたストリーミング配信が注目されている.PPLive,PPstreamなどのP2PストリーミングはBitTorrentをベースとしており,各ピアはストリーミング情報の円滑な流通を促すため単位時間内により多くアップロードしてくれたピアを接続ピアとして選択する.ライブストリーミングにおいてはピア間で同一場面を再生する時刻に差が生じており,再生が遅れているピアは再生が先行しているピアに比べアップロード可能なチャンクを保持している確率が低いため,各ピアの再生タイミングによりアップロード量に差異が生じる可能性がある.本稿では,P2Pシステムを用いたライブストリーミングにおいてアップロード公平性を改善する手法として,オフセット時間を考慮したピア選択手法を提案し,その評価を行う.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

近年,大容量コンテンツ配信に起因するサーバ負荷の増大が問題となっている.その解決策として,ネットワーク内に配置されたコンテンツキャッシュを用いてサーバ負荷軽減を実現するため,コンテンツの存在方向を示す足跡情報(インネットワーク誘導)をルータに残すBreaderumbs方式が提案されている.Breaderumbs方式においてコンテンツ要求は当初,ロケーションIDにより転送されるが,所望コンテンツの足跡にヒットするとその足跡情報によりコンテンツキャッシュ方向へ誘導される.この誘導によりネットワーク内からコンテンツ取得を行うことで,サーバ負荷軽減を実現する.本稿では,このBreaderumbs方式が部分普及した状況でのサーバダウンロード低減効果を,ユーザとネットワークの普及率に着目して評価する.その結果,Breaderumbs方式単体では,部分普及時においてユーザ普及率もしくはルータ普及率が低い段階では改善効果が小さいことを示す.この問題点の解決手法として,Breaderumbs対応ルータでオーバーレイネットワークを構成する方法を取り上げ,これにBreaderumbsの一部の機能を下位ネットワークへ開示する方法を組み合わせることで,ユーザ普及率,ルータ普及率の低い段階でも大きな改善効果が得られることを明らかにする.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

近年,広帯域・高遅延のネットワークにおいても高いリンク利用率の得ることができる輻輳制御として様々な高速輻輳制御が提案されている.これらの高速輻輳制御は既にMicrosoftのOSであるWindowsやLinuxOSに実装されており,現在のインターネット上でも利用されている.その中でも,WindowsではCompound TCP(CTCP)が,LinuxではCUBICがそれぞれデフォルトの輻輳制御として実装されている.CTXCPはロスベースと遅延ベースの複合型輻輳制御であり,CUBICはロスベースの輻輳制御である.そのため,これら2つの方式が有線環境下で混在した場合,CUBICがCTCPの遅延ベースの輻輳制御を圧迫し,公平性が著しく低下する.また近年,無線通信技術の発達により無線環境においても帯域遅延積の大きなネットワークが出現し,CUBICやCTCPのような高速輻輳制御を用いることが有効であることが予想される.そこで本研究では,無線環境下におけるCTCPとCUBICのような高速輻輳制御間の公平性に関する評価を行い,その特性について検討を行う.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

近年,データセンタにおける分散処理システムやソーシャルアプリケーション等によるPublish/Subscribe型通信の増加からデータセンタにおけるマルチキャスト通信が注目されている.マルチキャスト通信を行うことで,同一コンテンツを特定多数に送る際にデータセンタ内のリソースを効率的に利用できる.一方,大規模なデータセンタにおけるネットワーク管理を容易にするため,LAN/SAN統合の標準化が進められている.その標準化の一つとして,LAN/SAN統合に向けた2層輻輳制御方式QCN(Quantized Congestion Notification)が提案されている・QCNはユニキャスト通信を対象に設計されており,一般的に輻輳ポイントが複数箇所で発生するマルチキャスト通信に適用した場合,輻輳フィードバック情報を多く受信するマルチキャストフローの方がユニキャストフローに比べ送信レートが不当に抑圧される可能性がある.そこで,本稿ではマルチキャスト通信を考慮したQCN拡張方式として,ボトルネック選択機構を導入したQCN/BSの提案とその評価を行う.

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記述言語：日本語  

現在インターネットを流れるトラヒックのほとんどは，WebトラヒックやP2Pトラヒックなどコンテンツ流通に関連するトラヒックである．インターネットの基本通信モデルは，IPアドレスにより指定するホスト間通信，すなわちロケーションオリエンテッド通信である．コンテンツ流通においては，ユーザはコンテンツそのものに関心があり，どこからコンテンツが得られるかという点は問題ではない．このようなコンテンツオリエンテッドな通信をネットワーク自体が提供する枠組みとして，コンテンツオリエンテッドネットワークの研究が活発に進められている．コンテンツオリエンテッドネットワークにおいては，コンテンツ要求とデータとの間のランデブーポイントをネットワークが提供する，ランデブー型通信モデルが一般的に用いられる．本稿では，コンテンツオリエンテッドネットワークについて，モバイル環境の観点からその基本モデルであるランデブー型通信モデルを論じたうえで，最新研究例を紹介する．The primary use of the current Internet is content distribution and it generates large portion of traffic going through the Internet. Communication model has been point-to-point and location-oriented since creation of the Internet. In content distribution, user's concern is content itself and is not a location where it is obtained. From this viewpoint, content-oriented networks which change communication model to rendezvous-based model becomes recently one of hot research topics concerning new generation network. This paper overviews historical background and basic concept of content-oriented networks from the viewpoint of mobile communications. It also surveys content oriented-network researches in content discovery, content transport and content caching.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

ネットワーク内のルータで発生する輻輳に対し送信ホストの送信レートを抑制する輻輳制御については,これまで多くの研究例がある.特にトランスポート層でのTCPについては,対象とするネットワークの変遷に伴い,多くの改良型が提案されてきた.例えば10年ほど前までは,無線LANの普及,高速化というネットワークの変化に伴い,無線ネットワークを対象としたTCPが数多く提案されている.近年は,ネットワークの高速大容量化に伴い帯域遅延積が大きくなっており,この環境に適したTCPが数多く提案されている.帯域遅延積が大きいネットワークでのTCPの問題点として,AIMDにおけるAI区間が長くなり帯域を十分に使い切れないというScalabilityの問題がある.Scalabilityを向上させるためには,単純にレート増加を積極的なものに変更すればよいが,これだけでは既存TCPとの親和性やRTTの差異に伴う不公平性が悪化するという問題がある.これらの問題を解決する手法として興味深い方法が提案され,実際にWindowsやLinuxなどの広く普及しているOSにおいて採用されている.本稿では,これらの実用的な改良型TCPのいくつかを紹介し,その設計理念について説明する.また,これらのTCPが実際に混在して使用されている状況で発生する,全く新しい観点での公平性の問題について実際の評価例を用いて説明する.さらに,輻輳制御の新しい方向性として,データセンタネットワークを対象とした2層輻輳制御を紹介する.データセンタネットワークでは,帯域は非常に広いものの,伝送距離が非常に短いため帯域遅延積はそれほど大きくないがフレームロスに対する要求条件が従来のネットワークに比べ非常に厳しいという特徴がある.この特徴に応じて,現在標準化が進められている2層輻輳制御を紹介する.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

現在インターネットを流れるトラヒックのほとんどは,WebトラヒックやP2Pトラヒックなどコンテンツ流通に関連するトラヒックである.インターネットの基本通信モデルは,IPアドレスにより指定するホスト間通信,すなわちロケーションオリエンテッド通信である.コンテンツ流通においては,ユーザはコンテンツそのものに関心があり,どこからコンテンツが得られるかという点は問題ではない.このようなコンテンツオリエンテッドな通信サービスをサポートするために,これまでアプリケーション層でCDNやP2Pなど様々な取り組みがなされてきた.ただし,それを支える情報転送基盤であるネットワークは依然としてロケーションオリエンテッドなアーキテクチャに基づいており,上下階層の乖離がみられる.近年,この乖離がもたらす問題点を解決する手法として,コンテンツオリエンテッドネットワークの研究が活発に進められている.本稿では,このようなコンテンツオリエンテッドネットワークに関し,その歴史的背景,原理,そして最新研究動向について述べる.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

ネットワーク内に配置されたキャッシュを自律的に協調させる方法として,コンテンツの存在方向を示す足跡情報(インネットワーク誘導)をルータに残すBreadcrumbs方式が提案されている.コンテンツ要求は当初ロケーションIDにより転送されるが,所望コンテンツの足跡にヒットすると,その足跡方向に誘導される.この誘導により,サーバへの経路上にキャッシュがなくても適切な方向のキャッシュからコンテンツを取得することが可能である.本稿では,このBreadcrumbs方式が部分普及した状況での,サーバダウンロード低減効果を評価した.その結果,単なる部分普及時には非対応ルータにおける誘導経路の分断により,足跡情報の無効化が多く発生し,サーバダウンロードがそれほど低減されないことを明らかにする.また,その解決手法の一つであるオーバレイによる手法の評価も行い,その有効性を明らかにする.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

近年の大規模なデータセンタではLANとSANの二種類のネットワークが存在する.ネットワーク管理を容易にするため,LAN/SAN統合の実現が期待されている.パケットロスなどに対する要求条件の厳しいSANをイーサネット上で統合すること,イーサネットを多段に構成したネットワークがデータセンタ内で用いられることから,LAN/SAN統合に向けた2層輻輳制御方式QCN(Quantized Congestion Notifcation)が提案されている.しかし,QCNは各フロー間の送信レート公平性を保証するアルゴリズムとなっていないため,各フローの送信レートが不公平になるという問題が存在する.本研究ではQCNのレート増加アルゴリズムに着目し,各フローの送信レートに不公平な状況が発生する要因を明らかにする.また,その解決法としてAdaptive BC_LIMITを提案し,その評価を行う.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

近年,広帯域・高遅延のネットワークにおいても高いリンク利用率の得ることができる輻輳制御として様々な高速輻輳制御が提案されている.これらの高速輻輳制御は既にMicrosoftのOSであるWindowsやLinuxOSに実装されており,現在のインターネット上でも利用されている.その中でも,WindowsではCompound TCP(CTCP)が,LinuxではCUBICがそれぞれデフォルトの輻輳制御として実装されている.CTCPは従来の輻輳制御であるNewRenoと遅延ベースの高速輻輳制御との複合型輻輳制御である.また,CUBICはNewRenoとは全く異なったロスベースのみの高速輻輳制御である.その為,これら2つの方式が混在した場合,CUBICがCTCPの遅延ベースの輻輳制御を圧迫し,公平性が著しく低下することが考えられる.そこで本研究では,CTCPとCUBICの公平性に対する評価を行い,公平性向上を目的としたルータにおけるバッファ管理手法の提案を行う.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

無線マルチホップネットワークの技術的課題の一つに,伝送段数の増加に伴いスループットが大きく劣化する点が指摘され,特にTCPを用いた場合にはその劣化が甚大であることが知られている.この問題を解決する一つの方法に,TCPセッションのエンドエンド経路上にTCPプロキシを配置し,セッションをいくつかの短いサブセッションに分割しTCPクローズドループのRTTを小さくするプロキシ型TCPがある.筆者らは文献[7]においてタンデムモデルを用いその評価を行い,予想に反しプロキシ型TCPの性能改善が見られないことを明らかにした.この問題点の解決手法として,適応型レート制御,強制フロー制御という方式を提案し,大幅にスループット性能を改善できることを示している.本稿では,提案方式に対し複数のTCPセッションが存在するモデルを用いて評価を行う.その結果,トータルスループットは大幅に改善されるものの,公平性の点で問題があることを明らかにする.さらに,その原因を考察し,無線チャネルを有効に利用できていることから,かえって一方のフローが独占的に資源を利用できることが問題であることを示す.また,今後の公平性改善の方向性について考察する.

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

近年の大規模なデータセンタにはデータ通信用のLANとデータファイルの保存・管理用のSANの二種類のネットワークが存在する.ネットワーク管理を容易にするため,SANとLANをイーサネットで統合する方向が検討されている.パケットロスなどに対する要求条件の厳しいSANをイーサネットで統合すること,イーサネットを多段に構成したネットワークがデータセンタ内で用いられることから,LAN/SAN統合に向けた輻輳制御方式QCN(Quantized Congestion Notification)の標準化が進められている.しかし,QCNは各フロー間の送信レート公平性を保証するアルゴリズムとなっていないため,各フローの送信レートが不公平になるという問題が存在する.本研究ではQCNの基本特性を公平性の観点で評価し,フロー間公平性における問題提示とその解決法である公平性改善手法の提案と評価を行う.

CiNii Books

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J-GLOBAL

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チュートリアル講演

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

1対多通信であるマルチキャストにおいて,受信ノードごとの1対1通信での最大フローのうち最小値のものを,マルチキャスト通信における最大フローとすることは,一般に実現が困難であった.ところが,ネットワーク内のノードにおいてコーディング処理を行うことで,いかなるネットワークトポロジーにおいても上記のものをマルチキャスト通信時の最大フローとして実現可能であることが最近明らかになった.本稿では,このネットワークコーディングについて分かりやすく解説するとともに,その技術課題や今後の方向性について論じる.

CiNii Books

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J-GLOBAL

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J-GLOBAL

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

特定多数のホストに同一情報を配信する技術として注目されているマルチキャスト通信に対し, インターネット上でこれを実現するプロトコルとしてIPマルチキャストがある. IPマルチキャストの実現には, 全ルータがIPマルチキャスト対応ルータとなることが必要であり, IPマルチキャストを用いたマルチキャストサービスの普及は進んでいない. マルチキャストルータが普及しなければマルチキャスト通信を実現できないという制約からユーザを解放する方法として, ホスト(受信ノード)間の通信は従来のポイント・ツー・ポイント通信を用いた上で, ホストを構成要素とするツリーをオーバーレイネットワーク上に構築するアプリケーションレベルマルチキャストが提案された. アプリケーションレベルマルチキャストにおいては, パケット複製機能やマルチキャストルーチングの機能など, IPマルチキャストではルータが提供していた機能を, エンドホストが提供する. 本稿では, アプリケーションレベルマルチキャストの基本的概念について説明した上で, その技術・研究動向を紹介する. さらに, 今後の研究の方向性などについても論じる.

CiNii Books

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招待講演

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記述言語：日本語   出版者・発行元：関西大学  

CiNii Books

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奨励講演

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奨励講演

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

本稿では、様々な無線アクセスサービスが統合されたWireless Overlay Networkにおいて問題となるVertical Handoverに対するTCPの性能劣化について検討している.無線端末が異種の無線アクセスリンクに接続先を変更するVertical Handoverが発生した場合、TCPフローでBandwidth-Delay Productが急激に変更するためTCPのスループット特性が著しく劣化する.著者らは、この問題に対し、既にACK-Pacingを用いた改善方法を提案し単一TCPフローがネットワーク中に存在する場合には有効に動作することを確認している.本稿では、複数TCPフローが存在するより現実的な環境において、提案方式の実現性を検証する.この環境では、TCPフロー間の公平性を意識したACK-PacingとACKのオーバヘッド削減が新たな技術的課題となる.これらの技術課題を実現するための方法について議論し、シミュレーションにより複数フロー存在時の提案方式の有効性を評価する.

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

アプリケーションに対し高信頼のデータ伝送を提供するTCPは,有線ネットワークでの使用を想定し,ネットワーク内でのパケットロスはルータにおけるふくそうにより発生するという前提のもとで開発されてきた.有線リンクと比較して伝送環境の悪い無線リンクを含む無線ネットワーク上でTCPトラヒックを伝送する場合,ふくそう以外の原因によりパケットロスが発生するため,伝送特性が著しく劣化する.本稿ではこの問題点を解決するための無線ネットワークにおけるTCPの研究動向と今後の課題について解説する.

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奨励講演

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

WDMネットワークにおいて、使用帯域の小さい複数のトラヒックを同一波長で伝送するグルーミング(Traffic Grooming)技術は、高い波長利用効率を実現する上で効果的な技術である。ところが、使用帯域の異なるトラヒックが混在する環境でグルーミングを適用した場合、使用帯域の小さいトラヒックの方がコネクションを確立しやすいという不公平が生ずる。本稿では、この不公平問題を改善するための動的波長ルーチング割り当てを提案する。提案方式では、トラヒック要求を要求帯域が大きいトラヒックと小さいトラヒックの2つに分類し、それぞれに異なるルーチング波長割当ポリシーを適用する。計算機シミュレーションにより提案方式を用いることで不公平性が改善されることを明らかにした。

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記述言語：英語   出版者・発行元：SCRIPTA TECHNICA-JOHN WILEY & SONS  

Recent research has shown that long-range dependence is found not only in network traffic volume, but also in the propagation delay time of packets. The round-trip propagation time of packets, which is known as the round-trip time (RTT), is one of the most important parameters for determining TCP performance. The ability to understand its characteristics can contribute to an improvement in TCP performance. Therefore, in this paper, to determine the impact of the long-range dependence of the RTT on TCP performance, the authors looked at the two congestion control phases known as congestion avoidance and slow start in TCP window control and used computer simulations to evaluate performance. First, the evaluation related to the impact on the congestion avoidance phase showed that higher long-range dependence and larger transfer files caused greater variability in the file transfer completion time. Next, the evaluation related to the impact on the slow start phase, which evaluated the impact of the long-range dependence of the RTT on the TCP retransmission timeout (RTO), showed that higher long-range dependence causes unnecessary timeouts, which end up causing the resending of packets that have not actually been lost, to occur more frequently. The authors also performed investigations for mitigating the impact of the long-range dependence of the RTT and made a number of modifications to the existing RTO estimation algorithm to show that TCP performance can be improved. (C) 2004 Wiley Periodicals, Inc.

DOI： 10.1002/ecja.10073

Web of Science

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

マルチキャスト通信は、1対多型もしくは多対多型通信サービスをネットワーク層において効率よく提供する方法である。マルチキャスト通信を実現するTCP/IPプロトコルとして、IPマルチキャストがある。IPマルチキャストが提案されてから現在までかなりの年月を経ているが、未だ広く普及するに至っていない。本稿では、IPマルチキャストの現状を概説した後に、IPマルチキャストが普及していない理由としての技術的問題点について述べる。そして、これらの問題点を解決するために全世界の研究機関で進められている研究・開発について、特にサービスモデル、信頼性マルチキャスト、マルチキャスト輻輳制御、アプリケーションレベルマルチキャストの観点から詳しく紹介する。

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

CiNii Books

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記述言語：日本語   出版者・発行元：一般社団法人電子情報通信学会  

従来、マルチキャストとユニキャストの公平性を扱う場合には、TCP-friendlinessのように、リンクの帯域を公平利用するという、微視的な観点からの研究が主流であった。本稿では、マルチキャストがネットワーク帯域を効率的に使用するという点を考慮し、巨視的な観点から、マルチキャストをユニキャストに対し優先的に扱うアプローチについて検討を行う。具体的には、廃棄率の点でマルチキャストフローを優遇する、パケット廃棄法を提案し、その有効性を性能評価によって明らかにする。

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