Hierarchical Mobile IPv6 (HMIPv6) has been proposed to accommodate frequent mobility of terminals within the Internet. It utilizes a router, named Mobility Anchor Point (MAP), so that networks can manage mobile terminals without floods of signaling message. Note here that, particularly in a wide area network, such as a mobile communication network, it is efficient to distribute several MAPs within the same network and make the MAP domains cover overlapped areas. This enables the network to manage the terminals in a flexible manner according to their different mobility scenarios. The method described in the Internet-Draft at the IETF, however, lets mobile terminals select its MAP. This may cause load concentration at some particular MAPs and/or floods of signaling messages, because the terminals may not select a feasible MAP in a desirable manner. In this paper, a MAP selection method in distributed-MAPs environment is proposed. It reduces signaling messages to/from outside networks without load concentration at any particular MAPs. Finally, we show that the proposed method works effectively by simulation experiments.

Recently, wireless LAN is achieving remarkable growth and maturity. On the other hand, by the advance of the Internet, the demand for 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.

The recent explosive growth in information networks has driven a huge increase in content. For efficient and flexible information retrieval over such large networks, agent technology has received much attention. We previously proposed an agent execution control method for time-constrained information retrieval that finds better results by terminating an agent that has already acquired results of high-enough quality or one that is unlikely to improve the quality of results with continued retrieval. However, this method assumed that all agents have identical time constraints. This leads to a disparity in the obtained score between users who give individual time constraints. In this paper, we propose a fair and efficient scheduling method based on the expected improvement of the highest score (EIS). The proposed method allocates all CPU resources to the agent that has the highest EIS to decrease the difference between users' scores and to increase the mean highest score of requested results.

The authors have established new switching architecture of all optical WDM network suitable for the application of video transmission. In this paper, emphasis is put on a setup of a wavelength connection based on contiguous wave-band pool and we have proposed new Wave-Band Routing and Assignment (WBRA) method which provides simple switching and high speed wavelength assignment. Assuming the environment without wavelength convertor, our wave-band switching scheme is applied to several network topologies for performance evaluation. Then effectiveness and feasibility of this scheme are confirmed from a viewpoint of the number of required wavelengths. Simulation results indicate that our proposal scheme attains lower number of required wavelengths as compared to the fixed wave-band scheme. Assuming to use wavelength convertors, we have also evaluated the situation that the number of hops is restricted.

Multimedia traffic on the Internet is rapidly increasing with the advent of broadband networks. However, the Best-Effort (BE) service used with Internet Protocol (IP) networking was never intended to guarantee Quality of Service (QoS) for each user. Therefore, the realization of QoS guarantees has become a very important issue. Previously, we have proposed a queue management scheme, called Dual Metrics Fair Queuing (DMFQ), to improve fairness and to guarantee QoS. DMFQ improves fairness and throughput by considering the amount of instantaneous and historical network resources consumed per flow. In addition, DMFQ has characteristics of high speed and high scalability because it is hardware oriented. However, DMFQ may be unable to adapt to network fluctuations, given that it has static setup parameters. Moreover, DMFQ is unable to support a multiclass environment. In this paper, we propose a new buffer management scheme based on DMFQ that can adapt flexibly to network conditions and can provide classified services. The proposed scheme stabilizes buffer utilization within a fixed range by controlling the buffer threshold, which affects the calculated packet discard probability. Moreover, by applying the proposed scheme to Differentiated Services (DiffServ), we achieve prioritized buffer management.

Progress in the field of broadband access network and information appliances has led to the advent of a new network field called Home Network. In 1999, HomePNA2.0 using phone line was proposed, and we believe that it is one of the most promising solutions because of its cost-effectiveness. However, due to adaptation of the mature IEEE802.3 CSMA/CD technology used for Ethernet, it is not able to guarantee the QoS. We present the design, implementation and empirical evaluation of a new MAC protocol for the Home Network called HomeMAC. In this paper, the software based HomeMAC is implemented by programming the kernel space of FreeBSD. HomeMAC features a hybrid CSMA/CD-Timed Token protocol which combines the CSMA/CD for non-real-time traffic with timed token protocol for real-time traffic. In addition, by providing flexible bandwidth allocation based on QoS Level Table (QLT), HomeMAC can serve high QoS covering the whole offered load. From the results of evaluation of software implementation, we verify that HomeMAC can provide low delay, low loss, and low jitter to the real-time traffic by reservation of the bandwidth.

IEEE 802.11 DCF is a contention-based channel access protocol, and medium access delay greatly increases as the number of contending stations in a service area increases. This severely affects on delay-sensitive video applications. In this paper, we focus on MAC-layer solutions for realizing high quality video transmission in 802.11 DCF networks. A new channel access protocol based on the timestamp (TS) of video packets is proposed for real-time video. The TS information is carried by RTP header from the video application to 802.11 MAC. For video packets with the same RTP TS, they are simultaneously transmitted in a single channel access. Additional contention and back-off processes can be avoided because the whole packets of a video frame are completely delivered. The proposed TS-based access protocol can be easily implemented by the DCF with packet bursting. In addition, it is backward compatible to the legacy DCF. Extensive simulations show that the TS-based channel access achieves lower cumulative distributions of application-level video frame delay when compared to the DCF protocol.

To establish seamless and highly robust content distribution, we proposed the new concept of Inter-Stream Forward Error Correction (FEC), an efficient data recovery method leveraging several video streams. Our previous research showed that Inter-Stream FEC had significant recovery capability compared with the conventional FEC method under ideal modeling conditions and assumptions. In this paper, we design the Inter-Stream FEC architecture in detail with a view to practical application. The functional requirements for practical feasibility are investigated, such as simplicity and flexibility. Further, the investigation clarifies a challenging problem: the increase in processing delay created by the asynchronous arrival of packets. To solve this problem, we propose a pragmatic parity stream construction method. We implement and evaluate experimentally a prototype system with Inter-Stream FEC. The results demonstrate that the proposed system could achieve high recovery performance in our experimental environment.

Recently, various types of traffic have increased on the Internet with the development of broadband networks. However, it is difficult to guarantee QoS for each traffic type in current network environments. Moreover, it has been reported that bandwidth can be allocated to flows unfairly, and this can be an important issue for QoS guarantees. Therefore, we have proposed a flow-based queue management scheme, called Dual Metrics Fair Queueing (DMFQ), to improve the fairness and QoS per flow. DMFQ discards arrival packets by considering not only the arrival rate per flow but also the flow succession time. In addition, we have confirmed the effectiveness of DMFQ through several computer simulations. In this paper, we implement DMFQ with hardware for high-speed operation. Concretely, we propose the design policies and show the hardware design results.

The block-based fast transmission scheme, which is one of typical stored video delivery schemes, is reasonable in terms of its bandwidth efficiency and tolerance to the delay jitter, etc. However, it causes packet loss because of its burst data transmission method. Thus, we propose 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. Our 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 evaluate our proposal from some viewpoints with a computer simulation. We also measure the quality of the video reflected the result of a computer simulation. From these results, we confirm performance effectiveness of our proposal.

To guarantee strict Quality of Service (QoS) for real-time applications, we have previously proposed an output buffer control mechanism in IP routers, confirmed its effectiveness through simulations, and implemented a prototype. This mechanism can guarantee strict QoS within a single router. In this paper, we propose a control scheme of cooperation between IP routers equipped with this mechanism by using one of the signaling protocols. Our proposed scheme aims to stabilize End-to-End (E2E) flow delay within the target delay. In addition, our mechanism dynamically updates reserved resources between IP routers to improve E2E packet loss rate. We present an implemented design of our scheme and an empirical evaluation of the implementation. These results show quantitatively how our scheme improves the quality of video pictures.

B-ISDN is expected to be applied in the near future to video delivery systems for the broadcast of news and television programs. The demand for such services is increasing, and in particular, on-demand services are becoming more desirable. On-demand services allow viewers to request their favorite programs at the time that is convenient, hence catering for the wide range of modern lifestyles. As for on-demand services, there already exist Video on Demand (VoD) systems such as the original VoD or Near VoD. However, such systems have not yet been widely implemented because of the inefficient cost of communication resources, and storage. The authors' research is aimed at producing an efficient VoD system based on a high speed network. We are focused in particular on the forms of data transmission, and in this paper, we propose a new VoD system called Burst VoD. Burst VoD aggressively utilizes the multicasting technique, and involves dividing the program resource data into block files and transmitting them to viewer terminals as burst traffic over a high speed network. Simulation results comparing Burst VoD with conventional VoD show that Burst VoD achieves lower request blocking rates, efficient utilization of networks with multicasting, and almost on-demand response time to requests.

It is known that TCP data transfer in a wireless multihop network experiences a degradation in inter-connection fairness and throughput. This is because TCP is designed for use in wired networks, and the wireless multihop network has characteristics of sharing of the medium resources among nodes, which wired networks do not have. In particular, in wireless multihop networks where wireless nodes widely exist, hidden/exposed terminal problems are caused even if an RTS/CTS handshake is used. In this paper, two methods are proposed to improve fairness and throughput, without any feedback information from the intermediate nodes or cross-layer information. One method restricts the transfer period, while the other restrains the TCP congestion window. We evaluated these methods using computer simulations.

This paper presents novel multimedia ATM networks which are capable of transmitting voice data efficiently and unify the switching methods among heterogeneous traffic. Fully ATMized multimedia networks are using fellow cell switches. The proposed assembly method can pack plural calls which have different virtual channel connection (VCC) into one cell. Every call in cells is able to be dynamically rearranged by the fellow cell switch to achieve an efficient use of network resources. The switching functions are supported by shared virtual channel identifier (VCI) cells and fellow cells in it. The fellow cell switch for 622 Mbps links is integrated into a single chip. The multimedia ATM networks including voice transmission can be constructed by the fellow cell switches being attached to the standard ATM switches.

The requirement to realize large-capacity, high-speed and guaranteed Quality of Service (QoS) communications in IP networks is a recent development. A technique to satisfy these requirements, Multi-Protocol 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.

In Optical Burst Switching (OBS) networks, one of the main problems is collision between bursts. Most of the previous collision avoidance algorithms divide the Routing and Wavelength Assignment (RWA) problem into two partial problems and treat them separately. This paper focuses on the collision avoidance problem in distributed OBS networks. Our proposal involves cooperation between the routing and the wavelength assignment tasks. The main idea is to classify each wavelength at an output link of a node as suited either to sending or to relaying data bursts. The wavelength most suitable for transmitting bursts changes along the transmission route. Thus, we introduced a novel index called the "Suitability Index" (SI). The SI is a priority index assigned to each pair of output link and wavelength, and its value represents the suitability of that pair for sending or relaying data bursts. The proposed method uses the SI for both routing selection and wavelength assignment. Simulation results show that the proposed method can reduce the burst loss probability, particularly for long distance transmissions. As a result, unfairness in the treatment of short hop and long hop bursts can be reduced.

In future high-speed networks, provision of diverse multimedia services with strict quality-of-service (QoS) requirements, such as bandwidth, delay and so on, is desired. QoS routing is a possible solution to handle these services. Generally, a path selection for QoS routing is formulated as a shortest path problem subject to multiple constraints. However, it is known to be NP-complete when more than one QoS constraint is imposed. As a result, many heuristic algorithms have been proposed so far. The authors proposed a path selection algorithm Fallback+ for QoS routing, which focuses not only on the path selection with multiple constraints but also on the efficient use of network resources. This paper proposes an enhanced version of Fallback+, named Enhanced Fallback+, where in a shrewd way, it keeps tentative paths produced in the conventional Fallback algorithm with Dijkstra's algorithm. Simulation experiments prove the excellent performance of Enhanced Fallback+, compared with the original Fallback+ and other existing path selection algorithms.

The demand for large-capacity photonic switching systems will increase as regular broadband ISDN (B-ISDN) spreads and full-motion video terminals replace telephones. Large-scale and economical optical fiber transmission lines have been built based on time-division (TD) multiplexing. To reduce costs, it is important to increase the channel multiplexity of both transmission and switching systems by using TD and wavelength-division (WD) or frequency-division (FD) technologies. We surveyed photonic switching systems' architecture and switching network structures. Switching can be divided into circuit or synchronous transfer mode (STM) switching, and asynchronous transfer mode (ATM) switching. A variety of photonic STM and ATM switching systems based on the two switching technologies have recently been proposed and demonstrated.

This paper presents the switching control and VLSI architecture for the AAL2 switch. The ATM network with the AAL2 switch can efficiently transmit low-bit-rate data, even if the network has many endpoints. The switch is capable of not only switching AAL2 cells but also converting the header of other types of ATMs. The AAL2 switch is integrated into a single chip. The proposed ATM network is constructed by AAL2 switches attached to the ATM switches.

Recently, voice over WLAN has become an attractive service, and it is expected to be the most popular application in the near future due to its low cost and easy deployment. It has been reported that there occurs unfairness between downlink and uplink in the 802.11 WLAN. This is mainly caused by CSMA/CA employed in DCF. All stations including an AP fairly compete for shared wireless medium. Thus, in particular, the unfairness 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 with the legacy DCF. In addition, it requires only protocol modifications of an 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 can improve fairness between downlink and uplink, and consequently, support larger number of voice calls than DCF.