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 code. Implementation results show that the proposed scheme can scale with high-speed link, achieving the maximum rate of 4.0 Gbps by using the 3.5 µm CMOS technology.

Existing research related to RSVP with mobility support has mainly focused on maintaining reservation state along the routing path, which changes continuously with the movements of mobile host (MH), without much overhead and delay. However, problems such as deepening RSVP's inherent scalability problem and requiring significant changes in the existing network infrastructure have not been adequately addressed. In this paper, we propose a new approach, known as Concatenation and Optimization for Reservation Path (CORP), which addresses these issues. In CORP, each BS pre-establishes pseudo reservations to its neighboring BSs in anticipation of the MH's movement. When the MH moves into another wireless cell, the associated pseudo reservation is activated and concatenated to the existing RSVP session to guarantee continuous QoS support. Because a pseudo reservation is recognized as a normal RSVP session by intermediate routers, little change is required in the current Internet environment to support both movements within a single routing domain and between two different routing domains. CORP also dynamically optimizes the extended reservation path to avoid the infinite path extension problem. Multicast addressing is used to further reduce resource consumption in the optimization process. The experimental results of the CORP implementation demonstrate that it significantly reduces the delay and overhead caused by handoffs compared to the case of establishing a new RSVP session. The improvement increases as the distance between the MH and its correspondent host (CH) grows.

In the past, enhancement techniques for the end-to-end quality of a networked application were studied by looking at each individual layer. Examples of such techniques include the error resilience/concealment methods in the application layer, the FEC/ARQ in the transport layer, and the Quality of Service (QoS) techniques in the network layers. However, an integrated approach that would look across all related layers had yet to be investigated. This paper proposes an approach that combines priority-aware video packetization, adaptively used layered FEC, and QoS controlled networks such as IntServ and DiffServ in order to provide an efficient end-to-end quality in layered streaming video. The combination is more efficient in terms of a simple network price mechanism, that is, in getting the best end-to-end quality under a given total cost constraint. Our proposed approach in DiffServ with video packet (VP) data-splitting and layered FEC guarantees minimal service quality in a scalable and cost effective manner without introducing resource reservation. For video, we also propose performance metrics such as corrupted/frozen frame ratio (CFR, FFR). These provide objective metrics like PSNR as well as a measurement for subjective perceptions. Our approach, which combines related layers such as video coding, transport, and network, has yielded results that have proven to be more cost-effective and practical than the supporting network QoS.

In recent years, requests for new advanced features, decentralization, and improvements in the forwarding speed, have been made for IX. A new IX architecture using MPLS (Multiple Label Switching) technology (MPLS-IX) is proposed to solve these requirements and rectify the problems. In this paper, a comparison between current IX and MPLS-IX based on router's packet forwarding performance is examined, and the feasibility of MPLS-IX is discussed. Moreover, the QoS requirements to provide IX service are discussed.

At present, in the Intelligent Transport Systems (ITS) field, research continues In-The-Vehicle Communication (ITVC), Inter-Vehicle Communication (IVC), Road-to-Vehicle Communication (RVC), etc. All information communications technology, especially radio-communications technology, was applied. For example, wireless 1394 is used in ITVC, millimeter-wave communication is used in IVC, and Radio on Fiber (ROF) communication technology is used in RVC. However, it is actually very difficult to design for and accommodate all of these systems. This research, to simplify a design, equipment, structure, etc. of ITCV, IVC, and RVC, utilizes a wireless Bluetooth technology system which is the global radio-communications standard which is capturing the spotlight from various fields. In recent years, demand for a radio-communications service which can be used anywhere is increasing. Furthermore, demand for information service by the Internet is also increasing. This paper proposes simulation of Bluetooth wireless communication for ITS field such as ITVC, IVC and RVC system combined to the global Internet connection. Of course, there are many aspects should be thought carefully when the simulation system will be applied to the ITS infrastructures later. On this paper we have been designing the whole system and the evaluation of the simulation have been testing carefully.

This paper investigates the stochastic property of packet destinations in order to describe Internet access patterns. If we assume a sort of stationary condition for the address generation process, the process is an LRU stack model. Although the LRU stack model gives appropriate descriptions of address generation on a medium/long time-scale, address sequences generated from the LRU stack model do not reproduce Zipf-type distributions, which appear frequently in Internet access patterns. This implies that the address generation behavior on a short time-scale has a strong influence on the shape of the distributions that describe frequency of address appearances. This paper proposes an address generation algorithm that does not meet the stationary condition on the short time-scale, but restores it on the medium/long time-scale, and shows that the proposed algorithm reproduces Zipf-type distributions.

In this paper, we propose and analytically evaluate the use of punctured convolutional codes for recovering packets lost in multicast transmission. An independent erasure channel is assumed for packets transmission over a star topology. The analysis provides a method for determining the recoverability and the post-reconstruction receiving rate for a given convolutional code. We theoretically evaluate the effectiveness of the proposed approach taking into account two different parameters: the number of transmissions per packet and the number of packets needed to be sent to guarantee the reception of data. Finally, we compare the proposed approach with the scheme when parity packets are generated based on Reed-Solomon codes.

With increasing Internet traffic congestion, the provision of reliable transmission and packet loss recovery continues to be of substantial importance. In this paper, we analyze a new recovery method using punctured convolutional codes, demonstrating the simplicity and efficiency of the proposed method for the recovery of lost packets. The analysis provides a method for determining the recoverability and the post-reconstruction receiving rate for a given convolutional code. The exact expressions for calculating the recovery rate are derived for a number of convolutional codes and the (2, 1, m) punctured convolutional code. Where packet loss probabilities are in the range typically found in Internet transmissions, the convolutional code-based method delivers superior performance over the traditional parity method with the same redundancy.

Agent-based computing has been advocated by many researchers and software developers as a promising and innovative way to model, design and implement complex Web-related applications. KODAMA (Kyushu university Open & Distributed Autonomous MultiAgent) project, which is described in this paper, is our endeavour to advance both the technology and engineering of well-known multiagent systems. In particular, we have noted that software agents might be the potential solution to many problems faced by today's Web. However, building a high-quality, large-scale multiagent system, which can operate in open environments, is a great challenge. So far, we have devoted a lot of effort to design and implement a generic agent architecture, a hierarchical agent community structure, and an independent network communication middleware. To ensure that KODAMA can be used to create Web-agent applications, its network communication performance and a prototype distributed database retrieval system have been tested. The result shows that KODAMA is suitable for developing network-aware applications.

In this paper, we consider optimal mirror allocation problems for the purpose of load balancing in network servers. We focus on constructing high-reliability networks and propose the optimal mirror allocation model such that the system reliability is maximized subject to costs and delays, in view of the trade-off between the reliability and cost. This optimization model is capable of dealing with various kinds of network topologies, although for simplicity, we assume the read-only situation. We formulate this optimization problem into a 0-1 integer programming model, and we use an approximate method for numerical analysis in order to analyze more large-scale systems. Our objective is to find the optimal mirror allocation by solving this model, and to show quantitatively the general characteristics of the load balancing and the improvement of the system reliability by the distributed mirror allocation.

Guarantees of quality-of-service (QoS) in the real-time transmission of video on the Internet is a challenging task for the success of many video applications. The Internet Engineering Task Force (IETF) has proposed the Guaranteed Service (GS) in the Integrated Service model with firm delay and bandwidth guarantees. For the GS, it is necessary to provide traffic sources with the capability of calculating the traffic characteristics to be declared to the network on the basis of a limited set of parameters statistically characterizing the traffic and the required level of QoS. In this paper, we develop an algorithm for the evaluation of the traffic parameters which characterize the video stream when a QoS requirement is given. To this end an analytical traffic model for the VBR MPEG video is introduced. Simulation results show that the proposed method can evaluate the traffic parameters precisely and efficiently.

We proposed an improved Hierarchical Packet Fair Queueing (H-PFQ) mechanism, using ACK Spacing, for efficient bandwidth management of TCP traffic over Internet. According to the pre-determined bandwidth sharing and the class hierarchy of all TCP sessions, we design an algorithm to calculate the required time intervals between consecutive ACK packets of each TCP session to avoid packet drops due to buffer overflow. We demonstrated via computer simulations that the proposed improvement techniques may result in much better performance than merely original H-PFQ mechanism used in the forward direction in the sense that not only effective throughput of the bottleneck link is improved but also the fairness among TCP sessions can be maintained.

Recently, as multimedia and high-speed traffic become more popular on the Internet, the various traffic requiring different qualities of service (QoS) will co-exist. In addition, classified service 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 buffer management scheme for the Internet router that uses class-based priority control. This paper focuses on per-flow queueing, and evaluates the performance of the proposed buffer management scheme. Realization of differentiated services and dissolution of buffer occupation by specific flow is expected by the proposed control.

The provisioning of QoS in the Internet is gaining an increasing attention, thus the importance of methods capable of estimating the bandwidth requirement of traffic flows is constantly growing. This information can be used for a wide range of purposes. Admission control, QoS routing and load sharing all need the same basic information in order to be able to make decisions. This paper describes a number of methods that can be used to arrive at precise estimates of the bandwidth requirement focusing on those that are based on the theory of large deviations. A methodology is presented that allows the reformulation of earlier solutions based on the estimation of some form of an overflow probability so that their output becomes a bandwidth-type quantity, the format preferred by Internet control applications. The methodology provides two tracks for the conversion: an indirect method that encapsulates the overflow probability-type approach as an embedded calculation and a direct method that immediately results in the estimate of the bandwidth requirement. The paper introduces a novel method for the direct computation of the bandwidth requirement of Internet traffic flows using the many sources asymptotic regime of the large deviation theory. The direct bandwidth estimator method reduces the computational complexity of the calculations, since it results directly in the bandwidth requirement, allowing the omission of the frequent and costly computation of the buffer overflow probability. The savings arising from the reduction in computational complexity are demonstrated in a numerical example.

We describe a scheme of secret communication over the Internet utilizing the potentiality of the TCP/IP protocol suite in a non-standard way. Except for the sender and the receiver of the secret communication it does not need any entities installed with special software. Moreover it does not require them to share any key beforehand. Such features of the scheme stem from the use of IP datagrams with spoofed source addresses and their related error messages for the Internet Control Message Protocol (ICMP) induced by artificial faults. Countermeasures against IP spoofing are deployed in various places since it is often used together with attacks such as distributed denial of service (DDoS) and SPAM mailing. Thus we examine the environment where the scheme works as an intention and also clarify the conditions to obsolete the scheme. Furthermore we estimate the amount of secretly communicated data by the scheme and storage requirements for the receivers and those for the observers who monitor the traffic to detect the very existence of such a secret communication. We also discuss various issues including the sender anonymity achieved by the scheme.

For several years, more and more people are joining the Internet and various kind of packets (so called transaction-, block-, and stream-types) have been transmitted in the same network, so that poor network conditions cause loss of the stream-type data packets, such as voices, which request smaller transmission delay time than others. We consider a switching node (router) in a network as an N-series M/G/1-type queueing model and have mainly evaluated the fluctuation of packet delay time and end-to-end delay time, using the two moments matching method with initial value, then define the delay jitter D of a network which consists of jointed N switching nodes. It is clarified that this network is not suitable for voice packets transmission media without measures.

In this paper, a network dimensioning approach suitable to the Internet is discussed. Differently from the traditional telephone networks, it is difficult to guarantee QoS for end-users even in a statistically sense due to an essential nature of an end-to-end communication architecture in the Internet. We should therefore adopt another approach, based on the traffic measurement. In the approach, the traffic measurement is performed for monitoring the end-to-end QoS. Then, the network adaptively controls the link capacities to meet the user's QoS demands. For this purpose, the underlying network should support such a capability that the link capacities can be flexibly reused. With the WDM network as an underlying network, an example scenario for network provisioning is finally illustrated.

Fast packet switches for variable-size packets have become an everyday necessity with the rapid growth in the volume of Internet traffic. Such switches can be designed in two different ways, either by segmenting packets into smaller fixed-size cells and designing packet switches for such cells, or by designing generic packet switches for variable-size packets, where packet segmentation and reassembly can be omitted. The second option is investigated in this paper. The synchronous operation mode with time-limited bulk service is selected. The switching fabric is assumed to be internally non-blocking and provided with input queues. A previous maximum switch throughput analysis has been done under the assumption that the length of the time slot is fixed set to its minimum allowed value (Tmin). In this work, a so-called time-slot stretch factor (SF) is introduced. The actual time-slot length is determined by multiplying Tmin with the SF, where SF. Next, a so-called Internet traffic-source model is proposed based on findings from real IP traffic measurements. The performance implications of the proposed time-slot length modification are analyzed by discrete-event computer simulation. The maximum switch throughput is increased by increasing the SF value, e.g. for uniform packet size distribution and SF=10, the maximum switch throughput is increased from 75% to 97%. The influence of the traffic-source characteristics on the maximum switch throughput is decreased when SF value is increased. In order to prevent any possible throughput degradations, it is advisable to use integer SF values. Packet delay analysis has revealed that by increasing the SF value, the mean packet delay is also increased. Nevertheless, it is shown that the number of switch input and output ports is the most important factor to be considered when packet delay is at stake. Service class differentiation inside investigated packet switch is possible and is not affected by the increasing SF value. Such a packet switch is suitable for implementation in wide area networks, due to high transmission speeds and the small number of switch ports.

Teletraffic characteristics of a mobile packet communication network, which supports mobile Internet, were quantitatively evaluated by using a terminal migration model in which the cell dwell time possesses self-similarity. I used a migration model in which the migration speed of the terminal is determined by the density of the dwell terminals in a cell (determined from measured vehicular mobility characteristics). The transmission rates per terminal in a cell were estimated as teletraffic on the mobile packet communication networks using this migration model. I found that when there is self-similarity in the terminal cell dwell time, communicating terminals may be concentrated in the cell and restricted for an indefinite period of time to using only a narrow bandwidth.

In this paper, we propose a novel electronic bearer check system (EBC). This system allows the consumer to pay any amount of money below an upper-boundary on the Internet within an expiration period. During each transaction, the consumer does not need to contact the bank's server. Furthermore, this electronic bearer check can be transferred to any third party. The off-line characteristic of our system is very convenient for the consumer. Moreover, the double spending and double depositing problem will not occur in this system. More importantly, the framework of this system provides anonymity to protect customer privacy.