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.

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.

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.

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.

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.

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.

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.

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.

With advances in communication technologies, network services provided via the Internet have been widely diversified, and people can use these services not only via wired networks but also via wireless networks. There are several wireless systems in practical use such as cellular, WiMAX and WiFi. Although these wireless network systems have developed independently of each other, they should be integrated for seamless access by users. On the other hand, each system uses an individual spectrum prescribed by law to avoid radio interference. To overcome this problem, dynamic spectrum access technology receives much attention. We propose a dynamic spectrum assignment method in which a WiFi system temporarily uses a spectrum band of the WiMAX system in WiFi/WiMAX integrated networks to reduce call blocking probability of multimedia communication services. We confirm the effectiveness of the proposed method by simulation experiments.

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.

When multiple overlay networks are constructed over the same underlying IP network, overlay paths on different overlay networks may share some links on the IP network. In general, however, because overlay networks do not recognize the existence of each other, they have no consideration on such shared physical links. In this paper, a cooperative routing method for multiple overlay networks is proposed to improve the throughput of all 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.

Multiagent systems for information retrieval over large-scale networks are considered an effective technology and have received much attention. In a typical case, such information retrieval has time constraints, since the network contains too much information. In this paper, we assume that each retrieval result can be scored and propose a method to control agent execution to improve the score of the result that each agent has acquired. This method terminates the agents that are less likely to be able to complete their tasks by a given deadline and/or to acquire the result of a higher score than that they have already obtained. Finally, simulation experiments show the excellent performance of the proposed method.

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.

The development and enhancement of wireless communication systems such as cellular network and wireless LAN initiate the interests of heterogeneous networks where mobile users can make use of desirable wireless communication system. Furthermore, streaming services in a mobile environment have much attention with the wider bandwidth of wireless networks. Especially, TCP-based streaming service have more attention rather than UDP-based ones, since then can be used via some wide-accepted TCP-based protocols such as HTTP. However, in heterogeneous networks, the throughput of TCP-based communication is drastically degraded by decreasing of TCP congestion window size because of communication interrupt at vertical handoff. Therefore, QoS requirement of applications is not kept until the congestion window size increases enough again. In this paper, we proposed the method to immediately recover the TCP congestion window size by applying IEEE 802.11e EDCA. It can be implemented without extension of the existing TCP. Finally, we show the effectiveness of the proposed method by simulation experiments.

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.

Recently, as the performance of mobile hosts rapidly improves and the bandwidth of wireless access networks grows up, the expectation for mobile multimedia communication services including many-to-many communications such as videoconferencing begins building. Multicasting is one of the key technologies to provide such services efficiently. The existing multicast protocol called core based trees (CBT) is suitable for many-to-many communications in wired networks. In mobile networks, however, it has problems that the number of links of the multicast tree may increase and the end-to-end delay may not satisfy the QoS requirement from users due to tree reconstruction after handovers. Thus, we propose to change the core in consideration of the delay from the core to member nodes when a handover occurs and a new path is constructed. As a result, the proposed method decreases both the number of members who do not satisfy the delay constraint and the average number of total links of the multicast tree

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.

Mobile multicasting is a key technology to provide broadband services for mobile hosts. This paper also describes existing mobile multicast protocol and the proposed MFA selection method. Thus the proposed method provides relatively stable broadband services continuously.

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.

In the near future, it will be essential for multimedia networks to satisfy multiple quality of service (QoS) requirements, such as bandwidth, transmission reliability, end-to-end delay etc. On the other hand, multicast technology is being increasingly used in various continuous media applications that involve video and audio. In particular, multipoint-to-multipoint communication services, including multimedia collaborations such as video conferencing and shared workspaces, will be widespread. However, when a conventional multicast routing algorithm is applied to such services it may result in traffic concentration on some links. Therefore, we propose a new multipoint-to-multipoint routing method based on the Fallback+ algorithm, which can satisfy multiple QoS requirements and avoid traffic concentrations. Finally, simulation experiments show that this method results in improved traffic load balance and efficient use of network resources.

Hierarchical Mobile IPv6 (HMIPv6) has been proposed as one solution to accommodate the frequent movements of mobile terminals (MTs) in IP-based mobile communication networks. HMIPv6 utilizes a mobility anchor point (MAP) that allows the network to manage the mobility of MTs without causing any overflow of outgoing and incoming messages beyond the network perimeter. However, there is a problem in applying HMIPv6 to a wide area network having a multilevel hierarchy. The distribution of several MAPs within such a network can be used as a means of optimizing the different mobility scenarios experienced by MTs. However, HMIPv6 allows MTs to select their own MAP, so with HMIPv6, the MTs may not select the MAP that would be chosen in an optimized scheme. This may cause load concentration at a particular MAP and/or overflow of signaling messages to and from outside networks. An effective IP mobility management method is proposed that focuses on the speed of movement of the MTs. It reduces the number of signaling messages to and from outside networks without suffering from load concentration at particular MAPs, and we show, by simulation experiments, that the proposed method works effectively.

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.

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.

This paper presents an extended SIMD form data operation for multi-media signal processing and a performance evaluation of data-driven processor Qv-K1. By appending proposed data-parallel operation mechanism, the number of executed instructions is reduced compared to SIMD. The processing ability of this processor could be increased.

Intelligent routing control is defined as the process in which the network interrogates the databases containing the relationships between logical numbers, such as personal or information identifiers, and physical addresses in the transport network to find the terminal having the information required to process a user request. The routing control system presented uses distributed databases, each of which manages a switching system and all of which are connected through high-speed signalling networks separate from the transport network. If the requested physical address cannot be found in one database, search requests are distributed at the same time to all other databases. For up to 100 million subscribers, the routing control system can find a physical address within 1 s when each database uses ten memories accessed at 200 ns with an interdatabase linkage speed of 14 Mb/s.>

A key point in realizing broadband switching networks is architectual efficiency for switching various types of traffic with speeds ranging from several kbits/s to several Mbits/s. We propose a new switching technique, that is, a high-speed time division switching based on the single slot switching technique. This technique has several switching modules corresponding to standardized bit rates of traffic, and each cell is switched in the switching module corresponding to its speed. With this technique, all types of traffic will be equivalent with respect to the grade of service. This paper outlines its principle, clarifies the performance of traffic capacity,and finally proves its feasibility.

This paper outlines the features of a new integrated switching system for voice and data. It is composed of integrated subscriber terminals, integrated subscriber lines, an integrated digital switching system, and integrated storage facilities. This paper shows that the introduction of the integrated storage facility can create various new service utilities and improve the traffic capacity of the system.