In the road-to-vehicle communication (RVC) on intelligent transport systems (ITS), the frequent occurrence of shadowing caused by other vehicles deteriorates wireless transmission quality because of a small sized zone. However, a diffraction wave generated at the edge of vehicle can be utilized in applying adaptive modulation method with decreased modulation level. Therefore, it can be expected to keep communication only with a diffraction wave under shadowing. Hence this paper proposes an application of adaptive modulation for RVC system. This paper first reveals its improved effect in shadowing duration by computer simulation considering practical traffic flow, radio reflection and diffraction, and then shows that applying adaptive modulation can increase throughput performance largely.

In this paper, we propose a novel traffic engineering architecture for IP networks with MPLS backbones. In this architecture, two link-disjoint label switched paths, namely the primary and secondary paths, are established among every pair of IP routers located at the edges of an MPLS backbone network. As the main building block of this architecture, we propose that primary paths are given higher priority against the secondary paths in the MPLS data plane to cope with the so-called knock-on effect. Inspired by the ABR flow control mechanism in ATM networks, we propose to split traffic between a source-destination pair between the primary and secondary paths using explicit rate feedback from the network. Taking into consideration the performance deteriorating impact of packet reordering in packet-based load balancing schemes, we propose a traffic splitting mechanism that operates on a per-flow basis (i.e., flow-based multipath routing). We show via an extensive simulation study that using flow-based multipath traffic engineering with explicit rate feedback not only provides consistently better throughput than that of a single path but is also void of out-of-order packet delivery.

In this paper, we consider a direct sequence spread spectrum (DS-SS) microphone system in 2.4 GHz ISM band under indoor environments. It can be modeled as the asynchronous DS-CDMA system in which multiple transmitters share a common channel to transmit information to a single receiver. However, this system has the two critical defects, one is that many burst errors are occurred because of the long-term deep fade in an indoor channel, and the other is that power control is impossible due to the unidirectional structure. Since the reliable performance of DS-CDMA systems can be guaranteed under a reliable bit error rate performance and perfect power control, the solution to these problems is imperative. Hence, in order to overcome these problems, we propose a zero forcing equalizer (ZFE) based multi-user detection (MUD) receiver for this system, since an MUD receiver doesn't require rigorous power control and is able to remove multiple access interference using linear equalization. We derive the modeling and the tap coefficients of the proposed ZFE, and also analyze the performance of the proposed receiver. Based on the performance comparison with the conventional receivers, such as single user detection and post-detection combining diversity receivers, it is shown that the proposed receiver can achieve a lower bit error rate performance.

In mobile cellular networks it is crucial to be able to use the available radio spectrum as efficiently as possible while providing a certain level of Quality of Service (QoS) for the users. Emergence of miscellaneous services has dramatically increased the complexity of this problem by creating a heterogeneous traffic environment. In this paper an efficient resource allocation scheme between two real-time services with different bandwidth requirements has been proposed for cellular networks with multimedia offered traffic and highly mobile users, which combines classical intera-cell resource borrowing concept with a novel inter-cell resource-sharing scheme between diverse classes of traffic. By assuming the heterogeneous offered traffic to be a combination of audio and video traffic types, through extensive simulations it will be shown that HCBA-UCB is capable of significantly improving audio teletraffic performance of the system while preserving fairness in service provision, i.e. without imposing additional expenses upon video QoS performance.

This paper outlines and analyses major P2P technologies currently being actively studied at various research organisations or already in use in both business and personal environments. Various technical features of P2P as well as its history and market trends are shown. P2P principles focussing on discovery and network-organisational mechanisms of major P2P systems such as JXTA, SIONet, Freenet, and Chord are first summarised. Also, other technical issues related to delivery, network control, security, digital rights management, and distributed computing are described. Then, P2P technologies to promote next-generation community networks with autonomous and intelligent characteristics are addressed. Based on these, P2P applications are analysed and some examples are shown. Studies of P2P traffic measurement, which is difficult by using traditional evaluation methods, are taken up to consider total network design in the P2P era.

Both the Hopfield network and the genetic algorithm are powerful tools for object recognition tasks, e.g., subgraph matching problems. Unfortunately, they both have serious drawbacks. The Hopfield network is very sensitive to its initial state, and it stops at a local minimum if the initial state is not properly given. The genetic algorithm, on the other hand, usually only finds a near-global solution, and it is time-consuming for large-scale problems. In this paper, we propose an integrated scheme of these two methods, while eliminating their drawbacks and keeping their advantages, to solve object recognition problems under affine transformations. Some arrangements and programming strategies are required. First, we use some specialized 2-D genetic algorithm operators to accelerate the convergence. Second, we extract the "seeds" of the solution of the genetic algorithm to serve as the initial state of the Hopfield network. This procedure further improves the efficiency of the system. In addition, we also include several pertinent post matching algorithms for refining the accuracy and robustness. In the examples, the proposed scheme is used to solve some subgraph matching problems with occlusions under affine transformations. As shown by the results, this integrated scheme does outperform many counterpart algorithms in accuracy, efficiency, and stability.

In this letter, a resource allocation scheme is proposed to efficiently utilize the resource of CDMA systems with respect to the throughput. When we let the throughput be a function of various data rates for multimedia traffics, the scheme is to find the optimum set of data rates for concurrent user groups with which we can achieve the maximum throughput. It is also shown that the optimum data rate set always includes one non-trivial rate while all the others keep the minimum required data rate.

We study the statistical multiplexing performance of self-similar traffic. We consider that input streams have different QoS (Quality of Service) requirements such as loss and delay jitter. By applying the FBM (fractal Brownian motion) model, we present methods of estimating the effective bandwidth of aggregated traffic. We performed simulations to evaluate the QoS performances and the bandwidths required to satisfy them. The comparison between the estimation and the simulation confirms that the estimation could give rough data of the effective bandwidth. Finally, we analyze the bandwidth gain with priority multiplexing against non-prioritized multiplexing and suggest how to get better performance with the right configuration of QoS parameters.

Based on Radon transform, a novel method for registering a periodic (self-referencing) watermark is presented. Although the periodic watermark is widely used as a countermeasure for affine transformation, there is no known efficient method to register it. Experimental results show that the proposed method is effective for registering the watermark from an image that had undergone both affine transformations and severe lossy compression.

We propose a new dynamic routing algorithm that uses traffic and network state information to minimize the blocking rate and link congestion level. Our scheme uses the currently available link capacity in calculating the link weight by modifying Wang's approach, and computes the shortest path when a new call comes into the network. We consider the blocking count based update mechanism and the timer based mechanism, and conclude that the former is better than the latter in terms of efficiency and complexity.

This paper presents a new automatic-frequency control (AFC) configuration capable of removing wide range frequency offsets (up to about 0.625 fs, where fs is signal symbol rate). The new configuration consists of an AFC that removes frequency offsets between 0.125 fs and another AFC that detects the frequency offset range coarsely between 0.625 fs. This paper describes the principle of the new AFC configuration. The proposed AFC configuration employs four correlators to enhance the acquisition range. It also adopts the reverse modulation scheme to decrease the acquisition time. The performance of the new AFC configuration is confirmed via computer simulations. It is shown that the proposed configuration can accommodate wide range frequency offsets as well as reduce the acquisition time.

A principal attraction of ATM networks, in both wired and wireless realizations, is that the key quality of service (QoS) parameters of every call, including end-to-end delay, jitter, and loss are guaranteed by the network when appropriate cell-level traffic controls are imposed at the user network interface (UNI) on a per call basis, utilizing the peak cell rate (PCR) and the sustainable cell rate (SCR) values for the multimedia--voice, video, and data, traffic sources. There are three practical difficulties with these guarantees. First, while PCR and SCR values are, in general, difficult to obtain for traffic sources, the typical user-provided parameter is a combination of the PCR, SCR, and the maximum burstiness over the entire duration of the traffic. Second, the difficulty in accurately defining PCR arises from the requirement that the smallest time interval must be specified over which the PCR is computed which, in the limit, will approach zero or the network's resolution of time. Third, the literature does not contain any reference to a scientific principle underlying these guarantees. Under these circumstances, the issue of providing QoS guarantees in the real world, through traffic controls applied on a per call basis, is rendered uncertain. This paper adopts a radically different, high level approach to the issue of QoS guarantees. It aims at uncovering through systematic experimentation a relationship, if any exists, between the key high level user traffic characteristics and the resulting QoS measures in a realistic operational environment. It may be observed that while each user is solely interested in the QoS of his/her own traffic, the network provider cares for two factors: (1) Maximize the link utilization in the network since links constitute a significant investment, and (2) ensure the QoS guarantees for every user traffic, thereby maintaining customer satisfaction. Based on the observations, this paper proposes a two-phase strategy. Under the first phase, the average "link utilization" computed over all the links in a network is maintained within a range, specified by the underlying network provider, through high level call admission control, i.e. by limiting the volume of the incident traffic on the network, at any time. The second phase is based on the hypothesis that the number of traffic sources, their nature--audio, video, or data, and the bandwidth distribution of the source traffic, admitted subject to a specific chosen value of "link utilization" in the network, will exert a unique influence on the cumulative delay distribution at the buffers of the representative nodes and, hence, on the QoS guarantees of each call. The underlying thinking is as follows. The cumulative buffer delay distribution, at any given node and at any time instant, will clearly reflect the cumulative effect of the traffic distributions of the multiple connections that are currently active on the input links. Any bounds imposed on the cumulative buffer delay distribution at the nodes of the network will also dominate the QoS bounds of each of the constituent user traffic. Thus, for each individual traffic source, the buffer delay distributions at the nodes of the network, obtained for different traffic distributions, may serve as its QoS measure. If the hypothesis is proven true, in essence, the number of traffic sources and their bandwidth distribution will serve asa practically realizable high level traffic control in providing realistic QoS guarantees for every call. To verify the correctness of the hypothesis, an experiment is designed that consists of a representative ATM network, traffic sources that are characterized through representative and realistic user-provided parameters, and a given set of input traffic volumes appropriate for a network provider approved link utilization measure. The key source traffic parameters include the number of sources that are incident on the network and the constituent links at any given time, the bandwidth requirement of the sources, and their nature. For each call, the constituent cells are generated stochastically, utilizing the typical user-provided parameter as an estimate of the bandwidth requirement. Extensive simulations reveal that, for a given link utilization level held uniform throughout the network, while the QoS metrics--end-to-end cell delay, jitter, and loss, are superior in the presence of many calls each with low bandwidth requirement, they are significantly worse when the network carries fewer calls of very high bandwidths. The findings demonstrate the feasibility of guaranteeing QoS for each and every call through high level traffic controls. As for practicality, call durations are relatively long, ranging from ms to even minutes, thereby enabling network management to exercise realistic controls over them, even in a geographically widely dispersed ATM network. In contrast, current traffic controls that act on ATM cells at the UNI face formidable challenge from high bandwidth traffic where cell lifetimes may be extremely short, in the range of µs. The findings also underscore two additional important contributions of this paper. First, the network provider may collect data on the high level user traffic characteristics, compute the corresponding average link utilization in the network, and measure the cumulative buffer delay distributions at the nodes, in an operational network. The provider may then determine, based on all relevant criteria, a range of input and system parameters over which the network may be permitted to operate, the intersection of all of which may yield a realistic network operating point (NOP). During subsequent operation of the network, the network provider may guide and maintain the network at a desired NOP by exercising control over the input and system parameters including link utilization, call admittance based on the requested bandwidth, etc. Second, the finding constitutes a vulnerability of ATM networks which a perpetrator may exploit to launch a performance attack.

A scheme to evaluate the number of users and cell coverage of a WCDMA supporting multi-rate traffic is newly presented through calculation of the realizable Erlang capacity from a derived blocking probability and the path loss from the COST231 Walfisch-Ikegami (WI) model. Based on this analytical scheme, we evaluate the voice-data Erlang capacities at various data rates of 15 kbps to 480 kbps and the relationship between the cell coverage and the number of active users from them. When the value of Eb/Io is low from 4 dB to 3 dB under voice user capacity of 50 Erlang at 8 kbps, the result shows that the data user capacity is increased to 10 Erlang at low rate of 15 kbps and the cell coverage is enlarged to 100 m, and it is also shown that its capacity is increased to 0.2 Erlang at high rate of 480 kbps and its coverage to 50 m.

As the Internet has become the infrastructure for the global communication, the quality degradation due to network failures and illegal traffic such as DDoS (Distributed Denial of Service) have become a serious problem. In order to solve the problem, a network monitoring system that monitors the traffic of Internet in real time is strongly desired. Traffic monitors that collect the statistics from captured packets play a key roll in the system; however, they are not flexible enough for being used in the rapidly changing Internet. The traditional approach such that a new traffic monitor is developed for a new requirement results in a long turn around time of the development. Therefore, we have proposed a flexible network monitoring system that consists of programmable traffic monitors. Traffic monitors are made programmable by introducing active network techniques; therefore, we call the network monitoring system as the programmable monitor network. This paper describes the implementation of the programmable monitor network and its application to DDoS (Distributed Denial of Service) attack detection.

Many techniques have been proposed for segmenting feature point trajectories tracked through a video sequence into independent motions, but objects in the scene are usually assumed to undergo general 3-D motions. As a result, the segmentation accuracy considerably deteriorates in realistic video sequences in which object motions are nearly degenerate. In this paper, we propose a multi-stage unsupervised learning scheme first assuming degenerate motions and then assuming general 3-D motions and show by simulated and real video experiments that the segmentation accuracy significantly improves without compromising the accuracy for general 3-D motions.

Ultrafast all optical switching using pulse trapping by 100 fs ultrashort soliton pulse across zero dispersion wavelength is investigated. The characteristics of pulse trapping are analyzed both experimentally and numerically. Using the pulse trapping, 1 THz ultrafast all optical switching is demonstrated experimentally. Arbitral one pulse is picked off from pulse train. Pulse trapping for CW signal is also demonstrated and ultrashort pulse is generated by pulse trapping. From these investigation, it is shown that ultrafast all optical switching up to 2 THz can be demonstrated using pulse trapping.

We have proposed a speculative selection function for adaptive routing, which uses idle cycles of the network physical links to exchange network information between nodes, thus helps to decide the best selection. Previous study on the mesh network showed that SSR gives message selection flexibility that improves network performance by balancing the network traffic in both global and local scopes. This paper evaluates the speculative selection function on 2D torus network with simulation. The simulation compares the network throughput and latency with various traffic patterns. The visualization graphs show how the speculative selection eliminates hotspots and disperses traffic in the global scope. The simulation results demonstrate that by using speculative selection, the network performance is increased by around 7%. Compared to the mesh network, the torus's version has smaller gain due to the high performance nature of the torus network.

Using ultrafast nonlinear-optical response of organic dye films, a train of picosecond optical pulses can be converted into a space pattern of a mm scale. As applications of this technique we demonstrate a single-shot multichannel optical switching for 1 Tbit/s pulse trains, and a timing jitter suppression of pulse trains using a control system with femtoseconds time resolution.

Interconnection networks play a crucial role in the performance of massively parallel computers. Hierarchical interconnection networks provide high performance at low cost by exploring the locality that exists in the communication patterns of massively parallel computers. A Hierarchical Torus Network (HTN) is a 2D-torus network of multiple basic modules, in which the basic modules are 3D-torus networks that are hierarchically interconnected for higher level networks. The static network performance of the HTN has already been studied and has been shown to be good. Dynamic communication performance has been evaluated under uniform traffic pattern but not under non-uniform traffic patterns. In this paper, we present a deadlock-free routing algorithm for the HTN using 3 virtual channels and evaluate the network's dynamic communication performance under three non-uniform traffic patterns, using the proposed routing algorithm. We evaluate the dynamic communication performance of HTN, H3D-mesh, H3D-torus, TESH, and mesh networks by computer simulation. We find that the dynamic communication performance of HTN is better than that of the H3D-mesh, H3D-torus, TESH, and mesh networks.

Differentiated Services architecture provides a framework that enables relative differentiation of Assured Forwarding (AF) service. The differentiation is quantified by QoS parameters in terms of loss probability and maximum delay. We develop herein an efficient model to compute resource allocation in terms of buffer and service rate that satisfies the QoS differentiation between classes of service. To evaluate the performance of the proposed model, we conducted extensive simulation on both single-node and multi-node cases. The simulation studies show that the model can provide an efficient method to allocate network resources for aggregated traffic.