With the advancement of the information society, the importance of acquiring and utilizing various types of information from specific users with designated attributes has been increasing. This paper proposes an information acquisition method that leverages the data forwarding capability of delay tolerant network (DTN) to collect such information with mobile devices in a shorter time, without causing data transfer loads and costs on the fixed communication infrastructure. Its effectiveness is shown through computer simulation.

Practical optical buffers have not yet been implemented in optical packet switching (OPS) networks, where packet-by-packet switching is performed using optical signals at optical switches. As a result, packet collisions occur when multiple optical packets simultaneously reach the same output port of optical switches. This leads to significant communication performance degradation, particularly when Transmission Control Protocol (TCP) is used. In this paper, we propose a method that the receiver host adaptively delays the transmission of TCP acknowledgment (ACK) to achieve a predefined target round-trip time (RTT) for the sender host. This target RTT is determined based on real-time pure RTT measurements obtained during data transmission and reception. By dynamically adjusting the RTT of data packets traversing optical paths of various lengths in an OPS network to one of a limited set of fixed values, we can reduce the likelihood of such collisions.

In an Optical Packet Switching (OPS) network environment, the absence of practical optical buffers results in frequent packet collisions at intermediate nodes, as multiple packets may arrive simultaneously at the same output port of a certain intermediate node. This leads to substantial packet loss even under non-congested conditions. Consequently, the conventional Transmission Control Protocol (TCP) does not fully utilize the bandwidth of the bottleneck links. Moreover, highly bursty data transmission can impose the above impact excessively on intermediate nodes in OPS networks, making it undesirable. To address this issue, we have presented a control mechanism consisting of two disciplines: microburst transmission and congestion window updating. However, the performance of microburst transmission control itself, which corresponds to TCP pacing, has not been quantitatively clarified. In this paper, we quantitatively evaluate the performance of microburst transmission control. Specifically, we investigate the impact of different microburst transmission intervals on throughput and clarify the potential fairness of throughput among optical paths with different route lengths.

Networks of honeypots, similar to network telescopes, enable the monitoring of Internet threat activity. Honeypots can collect service-specific information about threat behavior and capabilities, depending on their configuration. This demonstration paper presents SweetsPot, a distributed honeypot data collection system. SweetsPot aims to facilitate threat research to model attacker behavior and characterize emerging threats. Additionally, SweetsPot supports live-streaming of event data, enabling rapid analysis, visualization, and timely threat response.

Nowadays, the number of wireless stations (STAs) connected to a wireless local area network (WLAN) is increasing, and the quality of service (QoS) requirements of transferred data are diverse. Even in such congested situations, it is necessary to assign sufficient frame transmission opportunities to each STA and base station (BS) according to the QoS requirements of transferred data. IEEE 802.11ax introduces simultaneous transmission of multiple frames by Orthogonal Frequency Division Multiple Access (OFDMA) for efficiently accommodating many STAs in a WLAN. For backward compatibility, OFDMA transmission in IEEE 802.11ax is used in cooperation with Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA) that is conventionally used in IEEE 802.11. In this paper, we propose a frame transmission method that selectively uses OFDMA transmission and CSMA/CA-based transmission, taking into account the QoS requirements of transferred data and the characteristics of the two types of transmissions.

An authentication technique based on the domain name system (DNS), called DNS-based authentication of named entities (DANE), has been used to support the simple mail transfer protocol. However, DANE has not been extensively used for hypertext transfer protocol secure. This is possibly attributed to the increase in latency caused by DNS security extensions, which are essential for DANE. Unlike email services, web-service access is negatively impacted by latency which considerably affects the quality of experience for end users. This study assessed the extent to which DANE affects webpage load time. To investigate the load time, we developed DANEWebPerf, a tool—comprising a web browser, DNS cache server, and proxy that enables DANE use—to measure the webpage load time. Our experiments showed that using DANE increased the average webpage load time by at most 1190.9 ms. Therefore, DANE for the web does not significantly impact the quality of experience of end users.

Edge computing over Named Data Networking (NDN) is expected to be a promising approach for the deployment of applications such as connected cars and virtual or augmented reality. In conventional edge computing frameworks over NDN, computing request packets are not forwarded based on the resource availability status of the edge execution nodes (e.g., CPU and memory utilization). This may result in requests reaching fully loaded nodes that cannot handle these requests. To address this issue, this paper proposes a resource discovery scheme for edge computing over NDN. In this scheme, each execution node distributes its own resource availability status within its vicinity via scoped-flooding. Using this information, requests are guided toward available nodes that can satisfy the requirements of the requests. Moreover, the nodes attach their status to the computed response packets and distribute the status to each node along the response paths. Our extensive simulations show that the proposed scheme helps request packets to effectively discover available edge execution nodes.

In lossy networks, including optical packet switching (OPS) networks and wireless networks, data recovery from frequent packet loss is essential. Erasure-coding, a type of forward error correction (FEC), is one of effective solutions, and furthermore, it could be combined with a technique that feeds back decoding progress information from the receiver to the sender for efficient data transmission. However, conventional feedback methods cannot inform the sender of the decoding progress information at the receiver when acknowledgment (ACK) packets are lost. Therefore, we propose a new data transmission method that is robust to high packet loss by feeding back accumulated information of decoded symbols at the receiver, and furthermore that smartly reduces the calculation overhead of encoding/decoding process. The proposed method can reduce the redundancy by up to 5.84% compared to conventional methods in environments where random packet loss occurs.

Technologies such as volunteer computing, which use computing resources of nodes in the neighborhood of people, have been attracting attention. When using a large number of end nodes with computing power in the neighborhood of people as an “End Cloud,” main problems in selecting an appropriate group of computing nodes that communicate with each other from among the End Cloud include the long physical distances between the computing nodes and the large amount of messages required to precisely grasp the amount of computing resources. In this paper, we propose an algorithm for selecting an appropriate set of computing nodes from the End Cloud taking these problems into account.

Wireless sensor networks and their applications are becoming more diverse, and as an important future challenge, urgent high-priority data must be transferred with minimum delay and loss in a prioritized manner. This paper proposes strictly prioritized multihop transmission of high-priority data, composed of not only single frame but also multiple frames, by reserving the channel and exploiting medium access control coordination among sensor nodes on the route. Its effectiveness is shown through computer simulation.

In 2016, Dyn Inc., a managed Domain Name System (DNS) service provider, experienced a DNS water torture attack. The attackers created several unique and unresolvable fully qualified domain names (FQDNs) with random labels and sent malicious DNS queries to the authoritative DNS server via DNS cache servers. This attack eventually caused the authoritative DNS server to become unserviceable. We propose a collaborative defense framework that minimizes the damage by quickly detecting the attack on the victim side and effectively defending against it on the attack source side. In this framework, the DNS cache servers (attack source) create FQDN-based allowlist filters to eliminate malicious DNS queries; the attacked authoritative DNS server (victim) sends a signal to activate filters on cache servers upon detection. Trace-driven simulations show that the proposed framework effectively detects and protects against stealthy attacks circumventing conventional countermeasures. Further, we find that disposable domains, which are designed for one-time use to send signals from DNS clients to authoritative DNS servers, have similar characteristics to FQDNs created for the attack. Moreover, the operation of disposable domains is found to be a key vulnerability to such attacks.

With the evolution of optical networks, it is expected that each server and user terminal will be equipped with an optical interface and directly connected to all-optical networks. In particular, Optical Packet Switching (OPS) network is one of the candidate future networks, where it can realize ultra-high-speed transmission with a highly statistical multiplexing effect. Thus, in addition to simple one-to-one high-speed connections, Service Function Chaining (SFC), including the selection of servers, optical route, and signal resources, needs to be provided on an optical network operation framework. In this paper, we demonstrate the possibility of selecting appropriate servers and optical signal-level communication resources with an optical network testbed. As a realistic solution to realize SFC on an OPS network operation, we implement a centrally controlled OPS network with software-defined network technology. We verify the dynamic selection of servers and optical communication resources based on aggregated load information of servers and confirm the feasibility of our SFC management.

The domain name system (DNS) has been an important base of the Internet; however, it faces various issues, including DNS server attacks. Blockchain-based DNS services have recently emerged to solve these DNS issues. Among them, Handshake provides a decentralized root zone and supports state-of-the-art features. However, few studies have investigated the performance evaluations of Handshake. We measure the name resolution latency on Handshake to assess its suitability as an alternative to the current DNS root servers and reveal that it has a practical drawback compared to current root servers at the current early stage.

Edge computing over Named Data Networking (NDN) is expected to be a promising approach for the deployment of applications such as connected cars and virtual/augmented reality. In conventional edge computing frameworks, computing request packets are not forwarded based on the resource availability statuses of edge execution nodes (e.g., CPU and memory utilization). This makes it possible for the requests to reach fully loaded nodes that cannot deal with these requests. To address this issue, this paper proposes a resource discovery scheme for edge computing over NDN. In this scheme, each execution node distributes its own resource availability status in its vicinity by scoped-flooding. This information guides the requests toward available nodes that meet the requirements of the requests. Our extensive simulations show that the proposed scheme helps request packets to discover available execution nodes effectively.

In recent years, wireless local area networks (WLANs) have become widespread, and the number of mobile stations (MSs) connected to WLANs and the number of WLANs installed have been increasing. In addition, owing to the diversification of applications using WLANs, transmission opportunities have been proportionally greater in the uplink direction, from MSs to base stations (BSs), than ever. Furthermore, most of this data is transmitted using Transmission Control Protocol (TCP). Therefore, WLANs need to provide opportunities for frame transmission in each direction according to the ratio of TCP traffic intensity. In this study, we propose a TCP-aware frame transmission method that makes effective use of Orthogonal Frequency Division Multiple Access (OFDMA) transmission as introduced in the IEEE 802.11ax standard to improve the TCP throughput of the entire WLAN while realizing the allocation of such frame transmission opportunities. In an environment where TCP flows are transmitted in both uplink and downlink directions, the proposed method transmits each frame by using OFDMA transmission and Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA) adaptively and selectively according to the buffering status of TCP data segments and ACKs in each direction.

Many enterprises are under threat of targeted attacks aiming at data exfiltration. To launch such attacks, in recent years, attackers with their malware have exploited a covert channel that abuses the domain name system (DNS) named DNS tunneling. Although several research efforts have been made to detect DNS tunneling, the existing methods rely on features that advanced tunneling techniques can easily obfuscate by mimicking legitimate DNS clients. Such obfuscation would result in data leakage. To tackle this problem, we focused on a “trace” left by DNS tunneling that cannot be easily hidden. In the context of data exfiltration by DNS tunneling, the malware connects directly to the DNS cache server and the generated DNS tunneling queries produce cache misses with absolute certainty. In this study, we propose a DNS tunneling detection method based on the cache-property-aware features. Our experiments show that one of the proposed features can efficiently characterize the DNS tunneling traffic. Furthermore, we introduce a rule-based filter and a long short-term memory (LSTM)-based filter using this proposed feature. The rule-based filter achieves a higher rate of DNS tunneling attack detection than the LSTM one, which instead detects the attack more quickly, while both maintain a low misdetection rate.

Currently, many kinds of wireless sensor networks need to communicate aperiodically generated packets like event detection as well as periodically generated ones for environmental, vital monitoring, and so on. Thus, collisions should be avoided among both aperiodic and periodic packets. In this paper, we propose a method to schedule transmission timings of aperiodic and periodic packets at the application layer from the source nodes. This has advantages of no modification to the MAC layer standardized by IEEE 802.11, 802.15.4, etc. and low cost of sensor nodes, compared with existing approaches based on the MAC layer modifications. To the best of our knowledge, this is the first proposal that controls transmission timings of both aperiodically and periodically generated packets at the application layer. Performance evaluation verifies that the proposed method improves packet loss rate compared with existing application layer approaches that control transmission timings of only periodic packets.

Currently, in addition to wireless LAN (WLAN), equipments for longer-distance wireless communication like IEEE 802.11af and 802.11ah are being developed. Thus, in the near future, access points (APs) and wireless stations (STAs) equipped with functions for both WLAN communication and longer-distance communication are expected. Although longer-distance communication enables STAs to connect to their APs even outside the WLAN coverages, the transmission rates are smaller. Multihop WLAN communication via relay STAs provides larger transmission rates. However, frame transmission from relay STAs increases collisions, and additional routing is required. This study proposes collision-protected multihop transmission without preliminary routing in the environment in which both WLAN communication and longer-distance communication are available. Collision protection and routing are processed by a two-way handshake between data frame transmitter and receiver stations with the longer-distance communication. After the handshake, the data frame is transmitted via relay STAs by exclusively using the WLAN channel.

Mobile crowdsensing is gaining wide popularity, and its main aim is to collect significant amounts of data by leveraging mobile terminals such as smartphones that provide a ubiquitous communication environment. On the other hand, named data networking (NDN) has become one of the most popular information-centric networks. In this paper, as a new paradigm, we propose an advanced crowdsensing system using NDN. Unlike previous studies, data, including uncertain information such as people's notions, opinions, and knowledge, are gathered from the outer locations of NDN, thanks to the delay-tolerant feature, which opens up new possibilities for data collection. The request for the data is forwarded in an area-focused manner. In addition, with this approach, we expect to significantly reduce the redundant data responses.

With the rapid increase in IoT (Internet of Things) applications, more sensor devices, generating periodic data flows whose packets are transmitted at regular intervals, are being incorporated into WSNs (Wireless Sensor Networks). However, packet collision caused by the hidden node problem becomes serious particularly in large-scale multi-hop WSNs. Moreover, focusing on periodic data flows, continuous packet collisions among periodic data flows are caused once periodic packet transmission phases are synchronized. In this paper, we tackle the compounded negative effect of the hidden node problem and the continuous collision problem among periodic data flows. To realize this objective, we propose a new MAC layer mechanism. The proposed method predicts risky durations during which collision can be caused by hidden nodes by taking into account periodic characteristics of data packet generation. In the risky duration, each sensor node stops the transmission of packets in order to avoid collisions. To the best of our knowledge, this is the first paper that considers the compounded effect of hidden nodes and continuous collisions among periodic data flows. Other advantages of the proposed method include that any new control packets are not required and it can be implemented in widely diffused IEEE 802.11 and IEEE 802.15.4 devices.

We pursue herein the establishment of an advanced and feasible Content Delivery Network (CDN) system by exploiting a simple Information-Centric Network (ICN) capability on the IP network, introducing a Breadcrumbs-based in-network guidance framework in addition to user-side caches. Through caching property analyses, we first present an interesting and beneficial feature by cooperation as follows: each popular content is automatically (without explicit control) cached on a designated surrogate server without duplication. Moreover, we propose a new cooperation method between CDN and Breadcrumbs, which enhances the abovementioned feature and consequently reduces the workload of origin servers and cache miss ratio by effectively using the cache capacity with a smaller processing overhead.

Recently, wireless LANs are closely located in various places, and radio interference occurs among them. In such interfering environments, especially, it is necessary to maximally increase transmission opportunities by smartly and autonomously discovering multiple frames that can be transferred in parallel. One typical approach is to control CST (Carrier Sense Threshold). However, a frame, intended to transfer in parallel with the precedence frame transfer, may interrupt the precedence transfer, which results in the loss of the precedence frame. In addition, the interference with ACKs replied from receiver stations is not considered. Therefore, in this paper, we propose a parallel frame transfer method based on strict judgement for the parallel transfer of multiple frames. In this judgement, all the frames transferred in parallel are intended to result in success, and the possibility of the parallel transfer with the replied ACKs is also checked.

Mobile terminal users prefer WLAN (Wireless LAN) to cellular network because of no connection fee. WLANs are available in some places where many people gather in urban areas, whereas public access network like WLAN should be deployed at a low cost for rural leisure venues and momentary events. In this paper, we propose a wireless access network with larger coverage than WLAN by the cooperation between WLAN and another wireless network with larger coverage. This achieves the wider deployment of such wireless access networks with a smaller number (cost) of base stations. Connection control that decides which wireless network should be used for each wireless station is also proposed to increase the effective network capacity of the wireless access network. Performance evaluations show that the proposal achieves larger effective network capacity compared with greedy connection that all the wireless stations inside the WLAN coverage use the WLAN and the other ones do the other network with larger coverage, respectively.

Recently, the use of big data has attracted attention as a profitable business strategy, and is expected to keep increasing in the future. In contrast to existing ways of big data analysis based on centralized computing environment such on a few huge data centers, we advocate a distributed and parallel computation environment aiming at fine-grained cloud computing, which includes so-called edge computing. In the advocated environment, it is assumed that many users, which own big data to be analyzed, dynamically participate in the network to request computer resources and leave after finishing their analyses. In such a dynamic and realistic environment, this paper improves the proximity of computer resources to big data by applying virtualized network in which nodes with each big data and the corresponding computer resources are mutually connected by proper optical paths. Optical path arrangement is periodically updated for new users, without affecting other users currently using computer resources. Moreover, a resource assignment algorithm suitable for such dynamic changes is also proposed to achieve fairness in terms of the network distance between big data and computer resources, and effective load balancing among resource suppliers. We evaluate its effectiveness by computer simulation.

Recently, wireless LAN has been diffused. Although unicast transmission is mainly used currently, multicast transmission, such as distribution of multimedia information, is expected to increase in near future. Thus, channel utilization of wireless LAN needs to be improved in general environments where multicast flows coexist with unicast. Although many packet transmission methods that achieve high channel utilization have been proposed, they focus on situations that only either multicast or unicast transmission exists. In this paper, focusing on general environments where multicast and unicast flows are communicated in parallel, we propose new packet transfer scheduling so that high network bandwidth and channel utilization can be realized by transmitting packets at as high rate as possible, by reducing packet loss, and by omitting the overhead to communicate control frames. These improvements are realized by effective cooperation between multicast and unicast flows. Evaluation results demonstrate that network throughputs of both unicast and multicast are increased while decreasing packet transfer delay.

In recent years, network function visualization (NFV), which attempts to provide services flexibly by virtualizing network functions on a general-purpose server, has attracted attention. Here, Service Chaining, which dynamically connects each network function (virtual network function: VNF), is critical. However, as it is difficult to select the optimum VNF, in a decentralized manner, especially the distances between the VNFs tend to increase, leading to larger communication and processing delays. Therefore, in this paper, we propose a distributed search method for ordered VNFs to suppress delays while considering the control server load, with an in-network guidance technology for query messages.

Smart meters migrate conventional electricity grid into digitally enabled Smart Grid (SG), which is more reliable and efficient. Fine-grained energy consumption data collected by smart meters helps utility providers accurately predict users' demands and significantly reduce power generation cost, while it imposes severe privacy risks on consumers and may discourage them from using those ``espionage meters". To enjoy the benefits of smart meter measured data without compromising the users' privacy, in this paper, we try to integrate distributed differential privacy (DDP) techniques into data-driven optimization, and propose a novel scheme that not only minimizes the cost for utility providers but also preserves the DDP of users' energy profiles. Briefly, we add differential private noises to the users' energy consumption data before the smart meters send it to the utility provider. Due to the uncertainty of the users' demand distribution, the utility provider aggregates a given set of historical users' differentially private data, estimates the users' demands, and formulates the data- driven cost minimization based on the collected noisy data. We also develop algorithms for feasible solutions, and verify the effectiveness of the proposed scheme through simulations using the simulated energy consumption data generated from the utility company's real data analysis.

With the development of communication services, backbone networks need to deal with various types of traffic by utilizing appropriate switching systems in a flexible manner. Optical packet and circuit integrated (OPCI) networks offer both optical packet-switching (OPS) and optical circuit-switching (OCS) and have been studied for this purpose. A space division multiplexing-based OPCI network provides significant communication capacity and shares spectrum resources for OPS and OCS traffic demands on the same physical infrastructure to efficiently accommodate future services. For appropriate resource utilization, the ratio of spectrum resources between OPS and OCS must be flexibly managed based on the dynamically changing traffic. One of important issues of OPCI networks is sharing spectrum resources under a dynamic traffic model. In this paper, we propose a dynamic spectrum-sharing method for both OPS and OCS data transmissions to enhance resource utilization. The spectrum-sharing method consists of dynamic resource boundary management and load-balancing by exploiting integer linear programming periodically in operation. We evaluate the proposed method with computer simulation and verify its effectiveness in terms of both OPS loss probability and OCS blocking probability.

Recently, by the spread of Internet of Things (IoT), wireless sensor networks and their applications are becoming more diverse. Future tasks in IoT include healthcare and factory networks, in which urgent data, in case of sudden condition change of elderly people or disaster due to machine failure, must be transferred with minimum delay and loss. However, existing methods for such prioritized transmission support only relative prioritization among different data traffic. In this paper, we propose a strictly prioritized transmission method of urgent data for transmission with minimum delay and loss. Urgent data are transmitted with strictly prioritized manner in MAC coordination between neighbor nodes along the route.

Lately, ZigBee has been attracting attention as a short-range wireless communication standard. In a ZigBee network, it is necessary to suppress the packet transfer load because ZigBee needs to operate with low power consumption and within narrow communication bandwidth. On the other hand, in typical environments in which ZigBee is utilized, such as a smart home network, it generally coexists with WLAN (Wireless LAN), which has relatively long idle periods. Thus, to reduce the communication load in ZigBee, this paper proposes a hop counts reduction method, where some packets in the ZigBee network are transferred via WLAN when the WLAN is not overloaded. Existing studies that assume an environment in which ZigBee and WLAN coexist are only concerned with interference avoidance between the two networks. On the other hand, this paper focuses on adaptive cooperation, which enables more efficient communication with low power consumption and higher channel utilization.

In recent years, the use of big data as a profitable business strategy has attracted considerable attention, and is expected to drastically increase in the future. We advocate a distributed and parallel computation environment for big data analysis aiming at fine-grained cloud computing, which includes so-called edge computing. This approach differs from the way that data analysis was performed in the past, namely based on a centralized computing model involving a few huge data centers. In contrast, the realization of a distributed computational environment in which computer resources with relatively low capacity are utilized requires optimal matching between the many owners of big data and the suppliers of computer resources. In this paper, we apply network virtualization with optical paths to improve the proximity of the computer resources to the big data, and present a resource allocation method to achieve fairness in terms of the network distance between the big data and computer resources, and effective load balancing among suppliers. The effectiveness of our approach is evaluated by extensive computer simulation.

Coexistence between ZigBee and WiFi technologies, which operate within the same frequency band, is increasing with the widespread use of the IoT (Internet of Things). ZigBee devices suffer significant decrease in the sink arrival rate of packets in the presence of WiFi interference. To overcome this problem, many channel control methods have been proposed. The existing methods switch only ZigBee channels to avoid interference with WiFi. Here, we propose a method for improving ZigBee packet arrival rate by controlling both the WiFi and ZigBee channels. In this paper, we propose a cooperative channel control method designed to improve ZigBee's packet arrival rate. We propose a method that controls not only ZigBee devices and channels but also requests a temporary pause in the use of specific WiFi channels. Finally, we show the effectiveness of the proposed method using computer simulations.

Recently, usage of wireless LANs (WLANs) by mobile devices has spread rapidly. However, these devices consume large amounts of electricity when they communicate through WLAN. Although IEEE 802.11 specifies power saving mode (PSM), it cannot let wireless stations (STAs) sleep in an optimal manner and increases packet transfer delay. In this paper, we propose a power saving method that uses a different framework. Specifically, STAs reduce power consumption by sleeping for the network allocation vector duration that is extended by bidirectional burst transmission within the acceptable delay of each downlink packet. In addition, we put forth the combination of the proposed method with PSM to improve performance in the situations for which the amount of traffic is small.

An increasing number and variety of devices are connected to the Internet. Thus, it is expected that peer-to-peer (P2P) networks, which have no central servers, will solve the server overload issue in the client-server communication model. Among P2P information search schemes, we focus on Skip Graphs, because they exhibit strong churn resilience and have a range search function, which is desirable for advanced networks such as smart meter networks, smart agriculture, smart cities, vehicular ad hoc networks, and online social networks. However, the overlay of a Skip Graph is constructed regardless of the nodes' locations. As a result, the end-to-end delay between communication nodes becomes much longer than the potential minimum. In a conventional method, this problem is solved by using special landmark nodes. However, this approach sacrifices some advantages of Skip Graphs as a P2P system. Thus, we propose a Skip Graph construction method that does not use any special nodes. The simulation results show that the proposed method provides about 35% improvement in transmission distance without additional query hops.

In recent years, caring for the elderly has emerged as a significant problem, and the number of elderly people living alone has also been on the rise. Against this backdrop, the authors propose a network for watching over the elderly in local communities. In this network, a group to watch over an elderly person is designed in a local community, and the watchers communicate with the elderly through messages over Online Social Networking platforms. Since several people in a group watch the elderly in this network, careful monitoring of the elderly and a reduction in the mental and physical burdens on watchers are achieved. The groups are organized per elderly person by the proposed method inspired by some Meta-Heuristic Algorithms so that the multi-objective requirements to each group, to quickly respond emergent unhealthy states of the elderly, and to notice any kinds of signs and symptoms proactively, are satisfied. Moreover, constraints to alleviate the watchers' burden are also applied as our original contribution.

Recently, the demand for wireless networks to treat periodic traffic is increasing, which includes healthcare wireless sensor networks. This tendency causes the problem of continual packet collisions among different periodic flows. In this paper, we propose a new formula for predicting whether two heterogeneous periodical flows from different source nodes have overlapping packet transfer durations. From this formula, we propose a transfer scheduling method that shifts the packet generation phase (timing) to avoid future collisions. This method adopts naive random access control, like DCF, for the MAC layer process. In addition, source nodes do not require significant computational power, because only the sink intensively schedules the timing and informs them to the corresponding source. Therefore, compared to existing methods in which each source node completely schedules the timing of creating packets based on time division multiple access (TDMA), our method requires less functional expansion, complexity, and computational power. Finally, we demonstrate the effectiveness of our method through computer simulations.

Various video distribution services have emerged, such as Ultra High Definition (UHD) videos, 3D videos and multi-view videos. Despite the fact that the issue of video distribution in best-effort networks is addressed in vast amount of literatures, little work has been done so far to consider video distribution systems in bandwidth guaranteed networks. In this paper, we propose a P2P-based UHD multi-view video distribution system in order to realize high quality play-back and rapid view switch by distributing UHD multi-view video in best-effort and guaranteed networks cooperatively. Through computer simulations, we demonstrate that the proposed system is suitable for future UHD multi-view video distribution systems.

Optical circuit and packet integrated network is a promising network architecture that can deal with various sizes/types of data efficiently. In this integrated network, paths (routes) for circuit and packet switching are assumed to be calculated and selected independently. However, this approach is insufficient in terms of efficiency and rapidity of total path calculation. Calculating appropriate paths with shorter computation time is necessary. In this paper, we propose and evaluate a high-speed multipath computation method that quickly calculates end-to-end paths for optical circuit switching by making use of the calculation results of next hop information for packet switching.

Information leakages are one of the main security threats in today's Internet. As ICN is expected to become the core architecture for Future Internet, it is therefore mandatory to prevent this threat. This paper proves that some ICN configuration prevents information leakages via Data packets and shows that it is an open problem to prevent interest packets from carrying encoded crucial information in their names. Assuming that names in ICN will follow the current URL format commonly used in the Internet, we get the statistics of web URL based on extensive crawling experiments of main internet organizations. Then we propose a simple filtering technique based on these statistics for firewall to detect anomalous names in ICN. The experiment shows that our filtering technique recognizes 15% of names in our dataset as malicious. As the false positive rate is still high for this filter to be used in a real world operation, this work is an important step for detecting anomalous names and preventing information-leakage in ICN.

Lately, network usage has been dominated by content distribution, and hence, content oriented network (CON) has attracted much attention. In CON, however, we encounter a scalability problem regarding content acquisition, particularly in large-scale networks composed of multiple domains, because the total amount of content is much larger than the number of network nodes. CON requires a more scalable routing scheme than the traditional IP routing. We have proposed a breadcrumbs-based architecture to establish highly scalable content routing for the content-oriented paradigm. In this paper, we consider another novel approach for controlling guidance (routing) information. We propose a breadcrumbs-scoping (BC-scoping) framework that manages the distribution scope of guidance information according to some criteria. First, we propose BC-scoping on domain, which limits the distribution scope of guidance information within domains according to a hierarchical network structure. It promotes intradomain content acquisition and prevents the occurrence of cross-domain BC trails, because intradomain content acquisition is preferred, both technically and economically. Second, we consider the use of multiple criteria simultaneously with this framework when the criteria are independent of each other. Specifically, we propose BC-scoping on domain and popularity as a practical example of BC-scoping on multiple-criteria. For both of the proposed methods, we show the effectiveness through extensive computer simulation.

In recent years, wireless networks have been utilized in many situations. However, further increase in network capacity is essential in order to transfer large volume of traffic demanded by many and various clients and applications. By introducing multi-channel environment, network capacity is increased. However, in cases that packet generation rates from wireless stations to a base station are different among wireless stations, all the communication bandwidth over multiple channels is not properly allocated to wireless stations with larger generation rates in particular, even though there is enough available network bandwidth in total. This issue becomes significantly serious in actual wireless access networks where the distance between the base station and each wireless station is different, that means different maximum transmission rate of each station, under the fading condition. Therefore, in this paper, we adaptively allocate proper communication channel, and as a result, adaptive bandwidth to all wireless stations so that they transmit all generated packets by equalizing sojourn time in their sending buffers if there is enough total network bandwidth over all channels. Finally, we show the effectiveness of the proposed method by computer simulation.

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.

Recently, wireless LANs are utilized in various situations. However, inefficient channel utilization is still a serious problem. One of effective solutions is to prevent packet loss and to transmit packets at a higher rate. While the backoff algorithm in IEEE 802.11 avoids only the packet loss caused by collision, other existing works tackle the packet loss caused by channel fading by transmitting packets at a lower rate. The authors have proposed a transmission scheduling algorithm which avoids decreasing transmission rate and improves channel utilization. However, these previous methods do not consider influence to TCP performance. In this paper, we propose a novel packet transmission scheduling which enhances TCP performance by effectively utilizing higher channel transmission rate and by keeping TCP congestion window size large. The proposed scheduling at MAC level avoids retransmission time out (RTO) of TCP ACK and duplicate ACK, without modifying TCP itself and adopting complicated cross-layer approaches. Evaluation results demonstrate that total throughput which aggregates all transferred TCP flows and the minimum throughput among the TCP flows are enhanced by explicitly decreasing the frequencies of duplicate ACK and RTO.

These days, in addition to host-to-host communication, Information-Centric Network (ICN) has emerged to reflect current content-centric network usage that many users are now interested not in where the content is but in acquired contents themselves. However, the authors believe that current IP network still remains, at least from deployment perspective, as one of near future network architectures. This is because ICN has various scalability and feasibility challenges, and host-to-host communication is also diffused like remote login, VoIP, and so on. Therefore, the authors' research aims to establish the feature of information-centric network on conventional IP network to achieve feasible architecture. In this paper, we propose to operate Breadcrumbs (BC) and Content Delivery Network (CDN) frameworks coordinately on IP network to improve the performance on content retrieval and acquisition. Both BC and CDN are important as a content-centric technique. Finally, we compare the proposed method with CDN that we carefully modeled through simulation. Simulation results show that our approach can reduce server load.

To guarantee the quality of service is one of the most important problems in ICT (Information Communication Technology) area. Therefore, we had proposed a service platform which integrates service control and network control by componentization of distributed objects. On the other hand, with advances of wireless communication technologies, users eager to have network services with a mobile terminal. However, this platform does not consider any wireless access networks. In this paper, we extend the platform to support mobile terminals with heterogeneous wireless access networks and also propose a traffic load balancing method on them for bandwidth guaranteed services. Finally, the effectiveness of the proposed method is evaluated by simulation experiments.

Currently, most of cloud computing services based on IaaS are provided by one specific provider for many users. However, this has some problems in terms of initial cost or ISP transit cost. In addition, individual small use is difficult because the services tend to be provided in a large unit to use. To solve these problems, we advocate a “participating cloud computing platform” that personal or small company level resource suppliers participate in, configure and provide cloud computing. Then, to efficiently search nearby resources among many and various ones in the platform, we propose Resource Breadcrumbs (RBC) as a key technology to provide in-network guidance capability for users' requests. This enables to supply users a smaller unit of resources: computing resource, memory resource, content, application etc. in comparison to traditional IaaS service. In the proposal, supplier-nodes distribute, around them, guidance information directed to themselves with dedicated control message. The distributed information enables user to autonomously locate nearby resources. In addition, distributed management reduces computational load on central database and enables to build cloud platform with lower cost.

What is an ultimately evolved form of cloud computing environment? The solution should have two important features; “Fine-Granular” and “Participating”. To realize such an attractive and feasible solution with the two features, we advocate a “participating fine-granular cloud computing platform” that a large number of personal or small company level resource suppliers participates in, configures and provides cloud computing. In particular, to efficiently search nearby resources among many and various ones in the platform, we propose Resource Breadcrumbs(RBC) as a key technology to provide in-network guidance capability autonomously for users' requests. This enables to supply users a smaller unit of resources: computing resource, memory resource, content, application etc. in comparison to traditional IaaS service. In the proposal, supplier-nodes distribute, around them, guidance information directed to themselves with dedicated control message, and additionally, the information may be logged along the trail of ACK message from central database server or supplier to user. The distributed information enables user to autonomously locate nearby resources. In addition, distributed management reduces computational load on central database and enables to build participating fine-granular cloud platform with lower cost.

In a wireless sensor network (WSN), one of the most important challenges is extending network lifetime because it is difficult to replace sensor nodes and recharge their batteries. In a WSN, nodes around a sink tend to die earlier than other nodes because these nodes have to transmit and relay more packets. To overcome this problem, many load-balancing methods were proposed. By the way, in recent years, two or more overlapping WSNs have been constructed in the same geographic area. In this situation, the overlapping WSNs are expected to extend their lifetime by cooperative packet forwarding. In this paper, we introduce shared nodes with multichannel communication capability as interchange points among respective WSN planes. With these shared nodes, we propose a novel cooperative routing method for WSNs to balance each other's load. Finally, we show the effectiveness of the proposed method by simulation experiments.

Recently, the demand of wireless sensor network treating periodical traffic is increasing. This tendency causes the problem of packet collisions among different periodical traffic. Specifically, packet collisions occur continuously in the case that packet generation and forwarding timing of a sensor node overlaps with other nodes. IEEE 802.11 DCF can detect packet collision and retransmit the lost packets to some extent. However, this approach does not solve the collision caused by the overlapping timing, because packets belonging to different traffic collide as long as the transmission timing is similar. In this paper, we propose a transmission scheduling method that shifts packet generation phase of periodical traffic to avoid the overlap of packet generation and forwarding. To realize appropriate phase shift, packet collision event is recorded at each relay node and the information is gathered at the sink. Then, the sink detects the traffic which experiences the most collisions and instructs the corresponding source node to shift the timing of creating packets.

These days, underwater exploration demand is increasing in the field of industry, science, military, and so on. Hence, Underwater Sensor Network to collect data by scattering sensors in water has been attracted attention. However, there are various problems specific to underwater environments including the communication medium. In this paper, we focus on some problems on broken or degrading link between two neighboring nodes moving away from each other along with water current. To recover such links, we propose to introduce and control mobile nodes to relay communication between such two nodes. Existing studies dealing with the controlled mobility focused on the movement from the sea bed towards the water surface in order to avoid broken links caused by a shadow zone. On the other hand, the proposed method recovers broken and degrading links caused by movement of sensors by the water flow. Different from exiting works, limited number of smart mobile nodes move in any directions. In addition, because the cost of node movement in UWSN is higher than that of terrestrial network, this method also aims at reducing the movement distance of mobile nodes. Furthermore, the proposed method can measure and deal with the link disconnection proactively because the movement of mobile nodes is controlled while grasping the network condition. Finally, we show the effectiveness of the proposed method by computer simulation.

Recently, multicast transmission in wireless LAN is increasing. On the other hand, due to wide diffusion of smartphones and tablet computers as well as laptop ones, electric power saving of STAs (wireless stations) becomes more important. However, when a few STAs which can use only a low transmission rate exist in multicast group, the AP (Access Point) needs to transmit packets to all STAs at the low rate. As a result, the power consumption of all the STAs extremely increases by extending operating period of transceiver. In this paper, to reduce total power consumption of STAs in multicast group, we propose multi-rate parallel transmission where packets are transmitted multiple times at high and low rates. This reduces power consumption of STAs on which a high transmission rate is available because packet reception at the high rate decreases operating period of transceiver and sleep period can be inserted. In other words, power consumption required for packet communication processing itself is reduced. This differs from existing power saving method in which STAs sleep only when there are no packets to communicate. Finally, we show the effectiveness of the proposed method by computer simulation.

In future networks, unknown and unintended traffic implosion and hot spot congestion may give serious damage because of heavy traffic implosion and rapid traffic fluctuation. To tackle this issue, the research on virtual network and next-generation network has been advancing. As a result, network control can be much more flexible than before. However, from the perspective of network architecture, the current approaches with fixed IP node/OXC node location still remain larger possibility to enhance the flexibility. Namely, novel solution is eagerly anticipated. For realization of highly advanced next-generation network, this paper suggests novel network architecture that can decrease a network scale, especially, not at the link level but at the entire topology level by adaptively leveraging the underlying OXC networks and by making virtual node function composed of several distantly positioned OXCs, named “Ameba Node”. Main benefits of the proposed Ameba Network include as follows. First, network-level load balancing can be attained by adaptively reconfiguring the Ameba network topology without any other complicated traffic engineering. Second, simple and naive routing algorithm and network controls would be applied thanks to its network level load-balancing capability. Thirdly, IP routing/optical-layer routing processes, and the resultant network resources can be balanced flexibly. Finally, flexibility on node renewal or network scale-up can be realized by not replacing old routers to most advanced brand-new ones but reconfiguring the shape of Ameba nodes in network adaptively. We also evaluate its effectiveness by computer simulation.

Video streaming using hierarchical cache servers is effective to provide high quality services to many users. However, less-accessed videos tend to be sparsely cached on cache servers far from users. Streaming qualities are degraded due to relatively high packet loss rate and long delay. In order to address the issues on heavy and bursty packet loss over such long distance transmission, we have focused on recovery characteristic of a multi-server Forward Error Correction (FEC) function named “Inter-Stream FEC.” In this paper, we propose a high quality streaming system exploiting hierarchical cache servers. The proposed system improves users' experience by adaptively applying Inter-Stream FEC. We also propose a parity data request method combines with parity server selection considering cache status of individual server and disjointness between paths. Simulation results show the proposed system can achieve higher restoration performance compared with a streaming system using traditional FEC, particularly when bursty loss occurs.

In this paper, we focus on an information gathering service like human sensing systems and supervisory system for nursing care, where a server collects information from many clients at the same time, under the current IP network following so-called Best-Effort Discipline. In such applications, especially, communication quality should be enhanced in order to attain application-specific target throughput at maximum. To tackle this technical issue, we propose a method which adaptively tunes TCP socket buffer size and the number of TCP connections according to the policy which preferentially ensures target QoS for high priority clients. This is an unprecedented research about many-to-one transport protocol which has functions to coordinate between multiple sessions. In addition, the proposed method is suitable for the system which transports information to a server considering various network access and priority levels. Through extensive simulation experiments, we demonstrate that proposed methods can ensure the required communication quality within physical limitations according to priorities, regardless of TCP versions.

Inefficient channel utilization is still a serious problem in wireless LANs. One of effective solutions is to decrease the frequency of packet loss. While the backoff algorithm in IEEE 802.11 avoids only the packet loss caused by collision, other previous works that prevent the packet loss caused by channel fading improve channel utilization with transmission at a lower rate. However, utilizing lower transmission rate degrades transmission efficiency because the channel is occupied for a longer time. In this paper, we avoid decreasing transmission rate with a novel transmission scheduling. Specifically, transmission to stations under fading channel condition is interrupted until the condition improves, and other transmission is processed on ahead. Moreover, we propose an extended RTS/CTS handshake which quickly detects the improvement of channel condition and minimizes the wasted time even if fading loss occurs. Evaluation results demonstrate that network throughput is improved without degrading the throughput fairness between stations.

Recently, network services based on distributed components that provide services by using components running on remote nodes have attracted much attention. Advanced services can be provided at a lower cost by using existing components. We have previously proposed a method that ensures a stable QoS for each service by allocating components dynamically according to changes in access frequency. In that method, however, we considered only CPU resources rather than network resources. In this paper, we propose a method for providing bandwidth-guaranteed services stably by considering available wavelength resources in backend servers on a platform with a WDM optical network.

Recently, multicast delivery of multimedia data via wireless LAN is increasing. Though IEEE 802.11 lacks the mechanism to recover packet loss in multicast, network coding (NC) has demonstrated its potential to retransmit lost packets efficiently. However, in existing NC methods, transmission delay becomes worse as the number of receiver stations increases because lost packets are continuously retransmitted until all stations receive them. In transmission of multimedia data, especially, transmission delay should be minimized. Therefore, we decrease transmission delay and improve its scalability for the number of receiver stations, at the expense of small packet loss acceptable for multimedia data.

On real-time video streaming, the distribution with multi-tree Application Level Multicast (ALM) is effective in terms of delay and jitter. In constructing multi-tree networks, a joining node selects counterpart nodes based on their performance or bandwidth capacity and they are connected each other. However, tree-based ALM has the overhead of reconstructing the tree topology when any node leaves. Multi-tree ALM has the large overhead of reconstruction. In addition, ALM does not consider distance between nodes on IP networks. This causes increasing delay and wasting network resources. In this paper, for the purpose of churn resilience, reduction of traffic volume and short delay, we propose a construction method of overlapped cluster-trees considering inter-node distance for resilient live video streaming. The proposed construction method consists of two main functions; clustering neighbor nodes on IP network and adjusting the cluster size. The overlapped cluster-trees are resilient to churn. Clusters autonomously manage member nodes according to the cluster size. The performance of the proposed methods is evaluated through extensive computer simulation. The result shows that the proposed cluster-trees can decrease transmission delay and increase resilience to churn compared with conventional multi-tree ALM.

Lately, access loads on servers and network traffic are increasing due to larger content-size and higher request-frequency in content distribution networks. Breadcrumbs (an architecture with guidance information for locating a content cache) is designed to reduce server-loads and to form content oriented network autonomously in cooperation with cached contents. We also proposed Breadcrumbs+ which solves BC's problem that queries visit the same routers repeatedly with forming a loop of routing. However, Breadcrumbs itself still has a technical issue that some queries potentially traverse longer hops compared to conventional approach. In addition, Breadcrumbs takes only a passive approach; a query is routed by a trail of BCs only when the query happens to encounter a corresponding BC. This indicates that Breadcrumbs has a space to enhance its capability. We expect that active and adaptive control of BC improves user experience. In this paper, we propose Active Breadcrumbs which provides higher cache utilization through active spreading guidance information around cache-nodes. Finally, we show the effectiveness of Active Breadcrumbs in terms of traversing content/query hops and server utilization through the extensive computer simulation.

In this paper, we propose a video data recovery method that exploits multiple streams. The proposed method gives priority to each data of multiple streams in terms of recovery from packet loss, and it is designed for recovering important data with higher probability. The priority of each data is calculated based on both inherent importance in one stream and loss probabilities of multiple streams. Specifically, this method deals with the data of less received streams as the important data. Moreover, the proposed method decides appropriate number of parity packets. This is because the transmission of too many parity packets might affect the other traffic, while that of too few packets might recover insufficient amount of data. The proposed method can improve video quality of total multiple streams. Our simulation results show that the proposed method can recover more important data with higher probability. Furthermore, the proposed method can also achieve almost the same restoration ratio among streams of recovery targets.

Recently, video streaming services using P2P(Peer-to-Peer) have attracted attention to solve the problem of load concentration on servers and reduce a lot of latency. Many P2P streaming systems, like Coolstreaming, however, have a complicated approach to control playback timings severely. And this leads to less churn resiliency and less adaptability to fluctuation of network traffic. Therefore, we focus on a simple and robust approach to realize “pseudo” streaming with high quality, which is based on BitTorrent. In the existing methods with the simple approach, peers download pieces just closer to playback timings to decrease the playback discontinuity. In contrast, in this paper, we propose a new P2P video streaming system considering the cooperation of three metrics; video structure, playback timings, and pieces dispersion on network. In this system, users vary three piece selections to suit the delivery status. Specifically, users preferentially download pieces which affect the video quality, are closer to playback timings, and improve the delivery efficiency. Moreover, we show the effectiveness of the proposed method by computer simulation.

Recently, optical network technologies are actively researched and developed. However, adaptive cooperation of optical network with network applications is not considered deeply. On the other hand, as a content distribution system, P2P(Peer-to-Peer) has attracted attention. Currently, content delivery with P2P is based on IP network, and its application to optical network has not been considered. Therefore, we consider adaptive cooperation between P2P and optical network, especially OCS(Optical Circuit Switching) network for high feasibility to distribute contents more efficiently.

Application level multicast with multiple distribution trees is effective for real-time streaming in terms of low delay and jitter. However, failure of upstream nodes drastically influences on the performance of many downstream nodes in multiple tree-based multicast. This degradation of video data continues until reconstruction of distribution trees finishes. Therefore, it is important to recover from the isolation as fast as possible. In this paper, we propose proactive and reactive approaches in order to resolve this problem. The proactive approach realizes fast recovery from isolation and efficient randomized forwarding by distribution trees cooperation. Each node forwards received data to its child nodes not only in the tree but also in other trees according to a pre-determined probability. In contrast, in the reactive approach, an isolated node requests forwarding from other nodes after detecting the parent node's trouble. This approach achieves quick response by requesting to near nodes. Moreover, we propose the hybrid method combining these two approaches. This hybrid method has advantages of both the proactive and reactive approaches. Finally, we evaluate the performances of the proposed methods through computer simulations, and confirm the effectiveness of the proposed methods.

Recently, network coding (NC) has been popularly applied to wireless networks in order to improve scarce wireless capacity. In wireless LANs, NC can be applied to packet retransmission in which multiple packets can be equivalently transmitted by a single retransmission trial at base station (BS). In this paper, retransmission based on NC cooperates with IEEE 802.11e EDCA. In EDCA, when network load is high, QoS is significantly degraded even in high priority class. To solve this, existing methods improve backoff control, and decrease packet loss caused by collision. However, backoff control cannot prevent packet loss caused by multipath channel fading. In the proposed cooperation between NC and EDCA, QoS of high priority class is improved from the aspect of efficient loss recovery. Unlike NC method with no QoS control, we encounter transmission scheduling problem among NC packet, single lost packet, and new packet. Moreover, in the constitution of packets encoded into NC packet, priority and traffic load of each priority class should be considered. Therefore, we propose how to determine the packet to be transmitted at next transmission opportunity in BS and how to constitute NC packets encoded by adequate set of lost packets. Finally, we show the effectiveness of the proposed method by extensive computer simulation.

Optical network technologies based on WDM (Wavelength Division Multiplexing) are actively researched and developed. However, adaptive cooperation of their property with network applications is not considered deeply. On the other hand, as one of attractive content distribution systems, many researchers focus on P2P(Peer-to-Peer) technologies. Currently, content delivery system with P2P is based on IP network, and its application to optical network has not been directly considered yet, though it is attractive to transfer huge content between data centers or between Origin servers, etc, in addition to forthcoming various large contents. Therefore we consider adaptive cooperation between P2P and optical network to distribute huge contents more efficiently but to supress the Origin server load, and propose the novel and future content distribution platform called BitTorrent over Optical Network (BT-ON). In BT-ON, the OCS (Optical Circuit Switching) is adopted as optical switching scheme due to its feasibility and suitability to large content transfer, while BitTorrent is done as P2P system due to its efficiency and generality. To overcome the technical issues caused by OCS network architecture where the physical resources are actually reserved by specific signaling processes, the BT-ON introduces some novel methods; application-triggered path set-up procedure, wavelength path allocation method, advanced improvement suitable for OCS network like wavelength-path based parallel request and enlargement of data transmission unit, and the peer grouping according to wavelength path reservation status.

In this paper, we propose a data recovery method that exploits multiple video streams. The proposed method gives priority to each data of multiple streams in terms of recovery from packet losses, and it is designed for recovering important data with higher probability. The priority of each data is set based on both inherent importance within a stream and loss probabilities of multiple streams. Specifically, this method deals with the data of less received streams as the important data. Moreover, we focus on the creation method of redundant data to be transferred from extra server for multiple worse conditioned streams, which greatly affects the recovery capability, and propose the creation method which is less influenced from the network condition. Our simulation results show that the proposed method can recover more important data with higher probability. Furthermore, the proposed method can also achieve almost the same restoration ratio among streams of recovery targets which have different loss probabilities.

Network services based on distributed components, which provide advanced network services with lower cost by flexibly assigning and reusing useful components running on remote servers, are attracting increasing attention. In this study, we propose a new platform which provides sustainable QoS and faster service time by allocating components dynamically with fault-tolerance. Specifically, we assume that many kinds of components are running on component servers and each component has its own Service Level Agreement (SLA). The proposed platform achieves to make a load-balancing system to meet SLA for service time. Moreover, the platform can provide faster service time to allocate components dynamically considering inter-component communication delay. Finally, to show the performance of our platform, we perform simulation experiments and prototyping.

To establish seamless and highly robust contents distribution technology in the environment of so-called multiserver contents distribution network, we focus on the Forward Error Correction (FEC) as an error recovery method, and we have proposed the new concept, Inter-Stream FEC, that means data recovery method between several streams by introducing redundant parity streams generated by inter-stream cooperation. In this paper, we extensively improve the proposed basic method. We focus on the recovery failure events in the original InterStream FEC method, and propose its remedy. The proposed method makes existing FEC and Inter-Stream FEC cooperate deeply and this leads to higher loss recovery ratio. Besides, we evaluate the basic method and the extensively proposed method through computer simulation and confirm that proposal can improve recovery capability.

P2P live streaming distribution services which can reduce load of distributor becomes commonly. The contents of these services are provided by not only corporations but also individuals. In existing systems for business services, however, transmission load of distributor still remains in high level to keep quality of service higher. Thus, it is difficult for private content distributors to provide for their own streaming contents. This paper presents and evaluates P2P live streaming system suitable for private distribution. This system constructs network considering transmission bandwidth of each node. In particular, a certain number of broadband nodes are carefully selected and set at the point nearby the distributor's node in P2P network. A set of the selected broadband nodes plays a role of a virtual distributor with large amount of transmission bandwidth and provides contents to peers instead of Origin. The original contribution of our proposal includes double mesh P2P network structure for scalable and efficient private live streaming services, and its detailed design description. In this method, network bandwidth is used effectively. As a result, it is expected to achieve the private distribution with higher scalability. With simulation experiments, we confirmed that the proposed system presents about half load of distributor compared with the existing system.

Multi-Channel MAC protocols increase network throughput because multiple data transfers can take place simultaneously. However, existing Multi-Channel MAC protocols do not take full advantage of the multi-channel environment, because they lack a mechanism allowing wireless stations to acquire vacant channel and time resources. In this paper, to achieve further throughput increase, we newly propose the Multi-Channel MAC protocols that effectively utilize idle channels and potentially available time of stations by employing burst and interrupted frame transfers, under the condition that each station can use only two transceivers. We assume a transceiver can behave as either a transmitter or a receiver at a time. We show the effectiveness of the proposed methods by computer simulation. Furthermore, through the evaluation as to more than two transceivers' case, we confirm two transceivers' case is best solution in terms of both attained throughput and hardware complexity.

Video distributions using TCP (called "TCP streaming") like YouTube have been increased rapidly with the improvement on the speed and capacity of the Internet. Thus, it is an important technical issue to guarantee QoS (quality of service) suitable for TCP streaming. However, it is difficult for the existing IP network to guarantee QoS of the TCP streaming. So far, in order to achieve the QoS enhancement per flow, we have proposed the class-based output buffer control mechanism in QoS-aware router by locally managing only active flows whose backlogged packets remain on the buffer. In this paper, we expand the function of this output buffer control mechanism so as to be suitable for TCP streaming. Specifically, we propose the mechanism which calculates transmission bandwidth and buffer volume preferentially available (called "basic volume") according to active flows' condition estimated by measuring transmitted packets per flow. Furthermore, through computer simulation, the effectiveness of our proposal is verified.

In this paper, we propose a new service platform which provides stable QoS by allocating components dynamically with considering fault-tolerance. We assume that many kinds of components are running on component servers and each component has its own SLA. The proposed platform achieves to make a load-balancing system to meet SLA of these components SLA and considering the number of replications necessary to offer a continuous service in case of failure. Finally, by implementation of its prototyping system, we confirm that the proposed platform is effective and feasible to provide the network services based on distributed components.

Recently, network coding (NC) has been popularly applied to wireless networks in order to improve channel utilization. In wireless LANs, when NC is applied to packet retransmission, a base station can simultaneously retransmit multiple packets destined to different wireless stations for a single retransmission trial. On the other hand, NC creates additional packet delay at both base station and wireless stations, and hence, packet transfer delay may increase seriously. However, existing NC-based retransmission methods do not consider this additional delay explicitly. In addition, when the number of flows is small, NC exhibits less benefits because the chances of NC-based retransmission are highly reduced. Therefore, in this paper, we propose a novel NC-based retransmission method in order to improve packet transfer delay and jitter of received packets. Moreover, to achieve further improvement of delay, jitter and retransmission efficiency, we propose a retransmission method in which NC-based retransmission cooperates with the typical ARQ method. We overcome the disadvantage of NC-based retransmission by combining with ARQ cooperatively. Finally, we show the effectiveness of the proposed methods by extensive computer simulation.

Network services based on distributed components, which provide advanced network services with lower cost by assigning and reusing useful components running on remote nodes, are attracting increasing attention. In this service, when a component fails, a component that has the same function running on another node can be substituted for it. On the other hand, there are three kinds of failures in network services; a software failure at a component, a hardware failure at a node, and a failure of the physical network. We propose a new method to identify a failure in network services, which determines not only a failure at the application level but also in the network layer by collecting a small number of messages via multiple overlay networks.

We propose a programmable and scalable traffic management scheme. Programmable traffic management at high-speed routers is difficult because programmability and high-speed packet processing have involved a serious tradeoff. To attain both, the new scheme combines control programs at a control server and simple packet handling functions, such as sampling packet headers and discarding packets, at routers. Therefore, by installing appropriate control programs into the server, a variety of active queue management schemes, per-flow bandwidth management schemes, DoS mitigation schemes, and so on, are achieved. One of the main contributions of this paper is its proposal of a statistical scheme for handling flows. As only a fraction of complete flow information stored at the control server is loaded into the router's flow table and it is replaced cyclically, the proposed scheme scales more than the router's flow table capacity. Our simulation results indicate that the scheme provides efficient traffic management, per-flow WFQ emulation in our example, even with very small flow tables compared to the number of concurrently active flows. Furthermore, we discuss implementation issues with the proposed scheme and reveal that the processing cost at the server and router is sufficiently small for use with 10 Gbps links.

Recently, sensor network systems are gradually moving into heterogeneous designs, incorporating a mixture of variety kinds of sensor nodes. Many routing protocols have been proposed for sensor networks but most of them did not consider the non-uniform energy drainage caused by many-to-one traffic. In this paper we propose a clustering method for wireless sensor networks with heterogeneous node types which select cluster heads considering not only transmission power and residual energy of each node but also those of it's adjacent nodes. Simulations show that our method significantly increases network lifetime and also maintains high coverage rate in comparison with existing methods.

Prolongation of network lifetime is one of the most important issues of sensor networks. EEHF (Environmental Energy Harvesting Framework) is a framework that includes a data gathering scheme and estimates the energy gained from the environment by using the periodicity of environmental power. Thus, sensor nodes can use environmental energy efficiently and the network lifetime is extended. However, since the accuracy of the estimation depends on the length of the period, and the data gathering scheme used in EEHF does not take environmental energy into consideration, this method has room for improvement. In this paper, we propose a more accurate and adaptive energy estimation method that does not depend on the length of the period, and a data gathering scheme that is optimized for the environmental energy-based sensor network, in order to prolong the lifetime of environmental energy-based sensor networks.

Multicast is efficient transfer scheme for contents distribution to a large number of clients, but it is necessary to meet security issues. On the other hand, source authentication is one of the important techniques for protecting from malicious users that plot eavesdropping, masquerading and so on. Though many authentication schemes have been proposed, most of them are not suitable for practical multicast network. The design of scheme should meet robustness against unreliable network. In this paper, we expand the existent authentication scheme using erasure code, and propose the novel control mechanism that cooperates with data reconstruction process. In addition, we show the effectiveness of our proposal by computer simulation.

In this paper, we propose more effective method for route search in ad hoc networks, that is extended from a most typical routing protocol, AODV. In the proposed method, nodes can search adequate route by means of unicast messages using out-of-date route information if possible, before searching it by means of flooding mechanism. This method introduces two new message formats; Unicast Route Request and Waiting Route Reply. In this proposed method, if the route to the destination indicated by Invalid entries is still available, the source does not have to do more Expanding Ring Search. As a consequence, proposed method can reduce the number of control messages.

In this paper, we aim to establish a seamless and highly robust content distribution technology in the environment of multiserver content distribution networks by focusing on Forward Error Correction (FEC) as an error recovery method. We propose a new concept, Inter-Stream FEC, which is a method for data recovery involving several streams, using redundant parity streams generated via Inter-Stream cooperation. The attraction of the Inter-Stream FEC method is that it can be implemented independently of existing per content FEC methods. As a result, impairment due to heavy packet dropping in a network locality can be repaired by using other healthy original streams and parity streams. After describing the details of our proposed method, we evaluate our proposal by computer simulation to confirm its efficiency.

To guarantee Quality of Service(QoS) for real-time applications strictly, we have proposed an output buffer control mechanism in IP routers, confirmed its effectiveness through simulations and implemented a prototype. This mechanism can guarantee QoS strictly within single router. In this paper, we propose the control scheme of mutual cooperation between IP routers equipped with the above-mentioned mechanism by using one of the signaling protocols. Our proposed scheme aims to stabilize End-to-End (E2E) delay of a flow within target delay. Also, reserved resources are dynamically updated between IP routers with our mechanism to improve E2E packet loss rate. We perform the implemental design of our scheme and empirical evaluation through the implementation. As one of results, we validated quantitatively that our scheme improves the quality of video picture.

The spread of broadband access to the Internet means that real-time applications have become commonplace. Video conferencing, communicating interactively in real time at multiple locations, is one such real-time application. For multipoint real-time communication, each member has to transmit information to all of the other members in the multicast delivery. However, there are some problems in real-time multicasting. Specifically, end-to-end delay and packet loss rate differ among multicast members. In this study, we aim to achieve fairness of time and quality among multicast members in multipoint realtime communication. In this paper, we assume that the multicast is of the application-level shared-tree type. To solve the problems mentioned above, we propose a new control mechanism in the root node of the shared tree. This mechanism consists of a proposed scheduling algorithm and a proposed discarding discipline. To make end-to-end delay among members uniform, the packet transfer order from the buffer is scheduled by considering the packet's generation time and the delay from the root node to the receiver. On the other hand, to make the packet loss rate among members uniform, the discard probability of packets that have arrived at the root node is decided by considering the packet loss rate in the network from the sender to the root node. Experimental performance evaluation demonstrates that our proposed system can achieve fairness in time and quality.

With an enhanced client computer performance and network bandwidth, peer-to-peer (P2P) content distribution networks are receiving much attention. However, along with the popularization of P2P content distribution networks, inter- Internet service provider (ISP) traffic and ISP costs are increasing, and pose an impediment for making a potential P2P content distribution network social infrastructure. This paper presents a method to constitute P2P content distribution networks and a reduction in ISP costs by considering the form of the ISP interconnection in its distribution. Results from our simulations show that our proposal achieves a reduction in ISP costs and distribution time, regardless of the number of peers joining a P2P network and the capacity of its content.

It has been reported that there occurs unfair- ness between downlink and uplink in 802.11 WLAN. This is mainly caused by CSMA/CA employed in DCF. The unfairness specifically has an adverse impact on bi-directional voice calls. Downlink voice connections become a primary factor to limit voice capacity. In this paper, we propose a novel medium access protocol, so called DCFmm, in order to improve QoS of downlink voice traffic as well as fairness between bi-directional voice connections. DCFmm is designed to enhance 802.11 DCF, and is fully compatible to the legacy DCF. In addition, it only requires protocol modifications of the AP. Thus, it can be easily implemented into existing 802.11 WLANs. DCFmm is compared with two conventional techniques through computer simulations. Extensive simulation results show that the proposed DCFmm scheme can improve fairness between downlink and uplink, and consequently, support larger number of voice calls than DCF.

Recently, wireless LAN is achieving remarkable growth, and maturity. On the other hand, by the advance of the Internet, the demand for the multimedia communication services which include video and voice will be expected to grow. Therefore, in the future, the mechanism of QoS guarantee must be realized even in wireless LAN environment. So far, IEEE 802.11e EDCF has been proposed, which is a contention based channel access method to achieve the QoS guarantee in wireless LAN. However, this cannot realize the desired throughput ratio or deterministic target throughput in principle. In this paper, we expand the EDCF to solve such QoS issues and enable more flexible QoS control. Moreover, we show the effectiveness of our proposal by computer simulation.

One of the promising solutions to combat wireless channel fluctuations in IEEE 802.11 networks is to appropriately exploit multi-rate capability of the physical layer. The mechanism to adaptively select actual transmission rates based on channel condition is generally referred to as RA (rate adaptation). Recently, T-RA (throughput-based rate adaptation) has been proposed to maximize system capacity. On the other hand, for multimedia traffic, E-RA (error-based rate adaptation) has been proposed, which guarantees target error rate. In this paper, we newly propose S-RA (service-based rate adaptation) which dynamically selects an RA method suitable for the traffic type of transmission packets. T-RA is employed for best-effort data while E-RA for real-time multimedia. Thus, in S-RA, different RA schemes are adaptively chosen on per-packet basis. The proposed S-RA architecture can be incorporated into the 802.11e QoS MAC where traffics are classified into multiple classes by their QoS requirements. Combined with 802.11e, the S-RA architecture can contribute to better QoS enhancement of real-time traffic, while preserving consistent BE throughput when compared with only T-RA or E-RA.

Various real-time applications are widely used with the development of network infrastructure. To guarantee QoS(Quality of Service) of these applications strictly, the Internet will require a flow-based priority control. Thus, we have proposed an output buffer control mechanism. In this paper, to guarantee End-to-End QoS maximally by exploiting the buffer control mechanism, we propose the method of mutual cooperation between IP routers equipped with our mechanism by using one of the signaling protocols. In addition, we show the effectiveness of our proposal by computer simulation.

With the rapid increase of the traffic demand in the Internet, it is necessary to provide a priority control mechanism in the Internet router in order to realize suitable transmission for various attributes of multimedia traffic. Therefore we have proposed the output buffer control mechanism by flow management, and hardware implementation is performed as a basic design. This implementation has only the basic functions. In this paper, we expand our implementation to realize the advanced QoS guarantee. Concretely, we propose and implement the two ideas, "correspondence to variable length packets," and "correspondence to multiple ports".

In packet networks including the Internet and commercial 3G wireless bearers, network conditions that a streaming application experiences are not known a priori and exhibit time-varying characteristics. For such environments, network-adaptive techniques are important to efficiently stream video. We propose an adaptive packet scheduling algorithm for streaming video over error-prone networks. A MAP (maximum a posteriori) framework is incorporated for adapting to time-varying network loss rate. By MAP-based dynamic network estimation, the proposed scheduler adaptively performs video packet transmissions over time-varying networks. Also, packet transmission schedules are optimally determined in a distortion-optimized way. For low computational complexity, the proposed packet scheduling algorithm is based on a new and simple distortion measure which is referred to the EFS (expected frame-rate reduction). Simulation results for standard video sequence show that the proposed scheduling outperforms conventional ARQ-based schemes from a view point of frame loss rate and reconstructed video quality. In particular, the proposed scheduling algorithm shows significant improvement over highly lossy channel.

We focus on content delivery services using erasure codes. By using erasure codes, reliable content distribution can be performed without multi-server scheduling. Thus, for the efficient use of network resources, we propose dynamic rate control adapting to the change of the network conditions for multi-server content distribution.

Recently, various applications have emerged with improvement of the speed and capacity of the Internet. In the future, various types of traffic will co-exist. Thus, it is required to guarantee QoS suitable for the attribute of each traffic. So far, to realize QoS guarantee per flow, we have proposed a class-based output buffer control mechanism, based on autonomous flow management. However, it guarantees only the bandwidth. We expand the function of this mechanism to guarantee delay, jitter and bandwidth. In addition, we show the effectiveness of our proposal by computer simulation.

As multimedia and high-speed traffic become more popular on the Internet, the various traffic requiring different qualities of service (QoS) must co-exist. In addition, classified services based on Diff-Serv (differentiated service), MPLS (multi-protocol label switching), etc., have come into wide use. Today's Internet environment requires routers to perform control mechanisms in order to guarantee various QoSs. In this paper, we propose a smart buffer management scheme for the Internet router that uses hierarchical priority control with port class and flow level. Furthermore, since the proposed scheme must operate at very high speed, we first propose several design policy for high speed operation and the hardware implementation is performed in VHDL (VHSIC hardware description language) code. Implementation results show that the proposed scheme can scale with high-speed link, achieving the maximum rate of 4.0 Gbps even though the conventional 3.5 /spl mu/m CMOS technology was applied.

In future IP networks, the main technical challenge will be to guarantee QoS for each flow. However, it has been reported that the typical congestion control scheme, random detection (RED), causes unfairness per flow, especially for TCP flows. The differentiated services (Diffserv) mechanism has also been studied, but as it is a class-based control mechanism, the unfairness per flow still remains within the class. Moreover, to guarantee a flexible QoS, routers require a flow-based control mechanism. In this paper, we propose a flow-based congestion control scheme, called RED with dual-fairness metrics (DRED), to dissolve the unfairness per flow and so provide a feasible QoS. DRED enables to consider backlogged packets in the buffer, and a flow succession time by simple flow management per router. In addition, DRED is a suitable scheme for TCP transmission, cooperating with TCP congestion control mechanism.

With the rapid diffusion of mobile terminals in recent years, the number of the Internet access from mobile terminals has been drastically increasing. In this environment, the condition of channels changes greatly due to moving terminal locations, and transmission errors frequently occur as a result. Retransmitting the dropped packet can usually correct the errors. However, when burst errors occur, retransmission repeats and the throughput of the whole base station degrades. Since the throughput per terminal depends on the channel condition, the throughput between terminals differs each other. This paper aims at improving the throughput at base stations and the throughput unfairness per terminal, and proposes a new packet transfer scheduling scheme with throughput compensated considering radio conditions. Our proposed method consists of two mechanisms. One is a throughput control mechanism which compensates the throughput to each terminal. The other is a transmitting control mechanism which inspects fine channels where little error occurs. Cooperating these two mechanisms, if a channel condition is good, packets belonging to this channel is prioritized for the purpose of higher transmission efficiency. Meanwhile, the proposed test protocol data unit (PDU), one of our proposals, is sent to a bad channel to confirm if the channel is restored or not. After confirming the restoration, the degraded throughput is compensated. The use of the test PDU controls the channel properly without wasting radio link bandwidth and can achieve the fairness between each terminal. Computer simulations have evaluated the performance of our proposed method. Our proposed method is verified its efficiency with the results showing the improvement of the throughput at base stations and unfairness per mobile terminal.

There is an increasing demand for the technology of content distribution, by which each user can request desired content through a network. Because of the low efficiency of existing systems, we proposed a new block transfer type video distribution system called burstVoD. The burstVoD system aggressively utilizes multicasting, and divides the content data into a mass of block files, which it periodically transmits to a terminal through a high-speed network, using a higher rate than the playback speed. However, by using the scheduling algorithm of the burstVoD system, when users request the same content from different periods, the VoD server repeatedly transmits the same block files in different periods. In this paper, we propose an advanced scheduling algorithm based on the burstVoD system to improve its multicasting efficiency. In addition, we propose a multi-channel burstVoD in order to reduce the interface bandwidth of client.

Digitalization of broadcast advances rapidly in Japan since the broadcast satellite (BS) digital broadcast services began in December 2000. It is a requirement that the receiver has a built-in HDD, for storing interesting programs, which may be connected with appliances over the home network. However, it is difficult to guarantee QoS on the home network composed of heterogeneous sub-networks. Therefore, for the purpose of an efficient and appropriate transmission of a video stream such as MPEG over the home network, it is required to develop a new scalable MPEG transmission method. For this reason, we propose a scalability technique considering impairments propagation (SIP), which takes into account the impairments propagated to other frames due to the loss of a macroblock in a given frame. We introduce the SIP technique, evaluate its performance and give a system model using SIP.

The block-based fast transmission scheme, which is a typical stored video delivery scheme, is reasonable in terms of its bandwidth efficiency. However, it causes packet loss because of its burst data transmission method. Thus, we suggest a slotted multicast scheme for MPEG video based on the block transmission scheme to maintain a higher quality and to include time constraints. We define two delivery units, the "GoPs group" and the "frame type", on the basis of the MPEG characteristics with periodical NACK feedback from the clients. The former is tolerant to burst packet loss, and the latter gives priority to important frames. Block multicast has two phases: a "transmission phase" and a "retransmission phase". In the former, a server multicasts a block, and in the latter, a server retransmits lost packets using multicast according to the proper delivery unit.

We believe that existing wire solutions such as Home-PNA2.0 and HomePlug and wireless solution such as HomeRF are the most promising solutions, because of its cost-effectiveness. However, MAC protocols of these solutions provide only class of service (CoS) using priority mechanism like HomePNA and HomePlug or consider only voice among real-time traffics like HomeRF For these reasons, we perceive the needs of the new MAC protocol which is no new wire solution and provides guaranteed quality of service (QoS) for not only voice but also video and audio. In light of this, we present the design and software implementation of a new MAC protocol for the home network called the HomeMAC. Our evaluation results of software implementation verify that HomeMAC can provide low delay and low jitter to the real-time traffic by reservation of the bandwidth.

With the advances in services using the present Internet, an increasing demand for video-on-demand (VoD) on the Internet may be expected in the near future. However, because of the best-effort characteristics of the Internet, it is important to suppress the degradation of communication quality caused by packet dropping when Internet traffic is congested. This paper focuses on MPEG transmission over the Internet, and suitable control mechanisms are established for an acceptable quality of service (QoS) improvement through detailed evaluation. Packets are classified using a frame-based scheme. The server applies the proposed end-to-end control scheme and shuffles the order of packets to avoid burst dropping, and may omit selected packets belonging to certain frames prior to forwarding. At the intermediate routers, transferred packets are transmitted according to round robin (RR) or weighted round robin (WRR) scheduling, and are dropped statistically using selective random early detection (RED) corresponding to frame attributes when there is congestion. We evaluate the proposed transmission method using both simulation of transmission characteristics and empirical measurements of picture quality.

In 1999, HomePNA2.0 using phone lines was proposed, and we believe it is one of the most promising solutions, because of its cost-effectiveness. However, owing to the adaptation of the mature IEEE802.3 CSMA/CD technology used for Ethernet, the QoS cannot be guaranteed. In light of this, we present the design, simulation evaluation, software implementation and empirical evaluation of a new MAC protocol for the home network called the HomeMAC. In this paper, the software-based HomeMAC is implemented by programming the kernel space of FreeBSD. The HomeMAC features a hybrid CSMA/CD-timed token protocol, which combines the CSMA/CD for non-real-time traffic with a timed token protocol for real-time traffic. In addition, by providing flexible bandwidth allocation based on a QoS level table (QLT), the HomeMAC can serve high QoS covering the whole offered load. From the results of the evaluation of the software implementation, we verified that HomeMAC can provide low delay, low loss, and low jitter to real-time traffic by reserving bandwidth.

The home network is one of the most important information infrastructures of the future. Among the home network solutions available, one promising technology is a HomePNA utilizing the phone line. However, it is not able to guarantee the QoS due to the adaptation of the mature IEEE802.3 CSMA/CD technology which is used for Ethernet. In light of this, we propose and evaluate a new MAC protocol for the home network called the HomeMAC that provides guaranteed QoS for appliances and PCs. HomeMAC features a hybrid CSMA/CD-timed token protocol which combines the CSMA/CD with timed token protocol and transmits real-tune traffic based on the QoS level table (QLT) for guaranteeing QoS. In the HomeMAC, there are two different transmission modes, namely, the CSMA/CD mode when there is no real-time traffic, and the timed token mode when there is real-time traffic taking place. By dynamically switching the transmission mode between CSMA/CD mode and timed token mode in accordance with the different kinds of traffic, the hybrid protocol provides low delay, low jitter, and low loss rate to multimedia appliances such as TVs, DVDs, and PCs. Moreover, by providing flexible bandwidth allocation based on QLT, the HomeMAC can serve high QoS whole covering entire offered load.

The requirement to realize large-capacity, high-speed and guaranteed quality of service (QoS) communications in IP networks is a new development. A technique to satisfy these requirements, multiprotocol label switching (MPLS) is the focus of this paper. In the future, it is expected that congestion and faults on a label switched path (LSP) will seriously affect service contents because various applications are densely served in a large area. In MPLS, however, methods to solve these problems are not clear. Therefore, this study proposes a concrete traffic engineering method to avoid heavy congestion, and at the same time, endeavors to realize a fault-tolerant network by autonomous restoration, or self-healing.

Previous studies on traffic measurement analysis in the various networks have shown that packet traffic exhibits long range dependent properties called self-similarity. Some papers reported that self-similarity degrades the network performance, such as buffer overflow. Thus, we need new network control considering self-similar properties. Network control considering the self-similarity requires high speed calculation method of the Hurst parameter. However, such a method has not been proposed yet. In this paper, we propose high-speed calculation method of the Hurst parameter based on the variance-time plot method, and show its performance. Furthermore, we try to show the effectiveness of the network control with self-similarity.

With the Internet transforming into multimedia and high-speed, the various traffic requiring different qualities of service (QoS) will co-exist in the network. Also, classified service based on Diff-Serv (differentiated service), MPLS (Multi Protocol Label Switching) and so on, have come into wide use. In today's environment of the Internet, it is necessary for the routers to perform the control mechanism to guarantee various QoS. We propose the buffer management scheme of the Internet router with class-based priority control, focusing on per-flow queueing, and evaluate the performance. We aim at the realization of differentiated services and the dissolution of the buffer occupation by specific flow.

We have already proposed a photonic IP switching system for the next generation Internet, in which a set of multiple encoding CDM signals are utilized as routing information. However, in the method, interference noise between channels may restrict routing operations. Thus a new restrict-complied routing method based on the Bellman-Ford algorithm is proposed. Also, we show that the proposed algorithm is more suitable for the multiple-encoding scheme in the optical CDM network than the original Bellman-Ford algorithm. Also, we study some topologies suitable for the multiple encoding CDM switching scheme.

Optical switching networks to transport vast amounts of information are important for B-ISDN services. Wavelength division multiplexing (WDM) is emerging as the dominant technology for future optical networks. A large capacity photonic ATM switch architecture using WDM technology is proposed in this paper. The proposed switch is suitable for WDM optical ATM networks, that is, the input and an output of the switch are wavelength division multiplexed. The switch can provide very large ATM cell switching capacity, for instance, 10 Tbit/s, with reasonable complexity. The basic switch module of the proposed switch has a simple architecture, and reduces the amount of a buffer hardware by introducing the WDM concept.

Using the rate control to restrict the peak cell rate of traffic is addressed. The asynchronous transfer mode (ATM) environment is considered as a discrete-time environment in which one slot is equal to one cell transmission time. A rate control that can restrict the number of cells during any fixed-length time interval in a discrete-time environment is studied. The scheme is realized by means of simple counter accumulation per slot, and it is easy to synchronize with slots. The simplicity of this scheme may make possible the flexible restriction of the peak cell rate according to network conditions. The scheme is analyzed, using matrix analytic techniques, in order to investigate the smoothing effect of bursty traffic and the queue length required to satisfy a certain quality of cell loss probability. Numerical results show the effect of restricting the peak cell rate of bursty traffic and the tradeoff between the smoothing effect and the required queue length.>

A traffic control scheme for FDDI (fiber distributed data interface) interconnection through an ATM (asynchronous transfer mode) network is presented. The scheme consists of two parts, a VP (virtual path) bandwidth allocation algorithm and a feedback type buffer protection flow control. Both use the acknowledge for the frame of FDDI as a control unit. Simulation results show that VP bandwidth allocated by the scheme increases according to arrival traffic when the destination FDDI has enough capacity and allocated bandwidth is not increased when its capacity is less than arrival traffic. Thus, the scheme has desirable VP bandwidth allocation characteristics. Simulation results also show that the scheme prevents cell loss at a destination gateway completely.>

The authors propose two multicast routing algorithms which distribute copy operation of packets over all nodes along the multicast path: a link-added type algorithm and a loop-constructed type algorithm. Both algorithms, at first, derive an approximate solution for minimum cast path, and then improve the solution to prevent concentration of packet copy operation at one switching node at a little sacrifice of total cost along the path. Computer simulation results show that too much copy operation per node can be avoided by these algorithms. Compared to the minimum cost solution of the tree-shaped multicast path, the solution of the proposed algorithms makes the average distance connecting a source-destination pair longer, but the sacrifice of total cost is verb small. These algorithms can be applied not only to packet networks but also asynchronous transfer mode (ATM) networks.>

An asynchronous transfer mode (ATM) switch architecture that uses the broadcasting transmission medium for transmission of cells from input ports to output ports is introduced. Cell transmission and its control are separated completely, and cell transmission control, i.e. header operation, is executed before cell transmission (control ahead). With this operation, cell transmission and its control can be executed in a pipeline style, allowing high-speed cell exchange and making 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. The switch fabric performance is analyzed according to its switching speed. Numerical results show that the ATM switch proposed shows good cell loss performance even when its switching speed is restricted, provided that switch utilization is below 1. Extensions to the switch that lead to robustness against bursty traffic are shown.>