ZCZ sets are families of sequences, whose periodic auto/cross-correlation functions have zero correlation zone at the both side of the zero-shift. They can provide approximately synchronized CDMA systems without intra-cell interference for cellular mobile communications. This paper presents ternary ZCZ sets achieving a mathematical bound, and investigates the average interference parameters for the sets in order to evaluate inter-cell interference. It is shown that they can provide AS-CDMA systems with efficiency frequency usage.

Differentiated Service (DiffServ) is a technology designed to provide Quality of Service (QoS) in the Internet, and is superior to Integrated Service (IntServ) technology with respect to the simplicity of its architecture and the scalability of networks. Although various simulation studies and estimations over testbeds have investigated the QoS that is offered via the DiffServ framework, almost all of them focused on the characteristics in a single DiffServ domain. However, the Internet is actually composed of a large number of AS domains, and thus packets are very likely to arrive at their destinations after going through many different domains. Therefore, we have analyzed the QoS performance in a model consisting of multiple DiffServ domains, and focused especially on the quality provided by Assured Forwarding Service (AF) to achieve statistical bandwidth allocation with AF-PHB. Our simulation results show some throughput characteristics of flows over multiple Diffserv domains, which clarify how network configurations impact the QoS over multiple DiffServ domains.

This paper proposes a new method to reduce sinusoidal noise components whose frequencies are known. The new method is based on the simultaneous equations technique. The technique does not require the secondary path filter: thereby the automatic recovering of the noise reduction effect deteriorated by secondary path changes becomes possible. This paper also presents computer simulation results to examine the performance of the new method.

A voltage-mode universal biquadratic filter using single current-feedback amplifier (CFA), two capacitors and three resistors is presented. The new circuit has four inputs and one output and can realize all the standard filter functions, that is, lowpass, bandpass, highpass, notch and allpass filters, without changing the circuit topology. The use of only one current-feedback amplifier simplifiers the configuration.

An image halftoning technique that uses a simple GA has proven to be effective generating bi-level halftone images with quality higher than conventional techniques. Many devices are designed to handle more than two halftone levels and a GA based multi-level halftoning technique is desirable. In this paper we extend the bi-level halftoning technique to generate multi-level halftone images. Also we introduce an improved GA (GA-SRM) into the proposed multi-level halftoning technique. Experimental results show that the proposed technique can effectively generate high quality multi-level halftone images and that the inclusion of GA-SRM substantially contributes reducing memory usage and accelerating image generation.

This paper presents the performance modeling, analysis, and simulation of SIP-T (Session Initiation Protocol for Telephones) signaling system in carrier class packet telephony network for NGN (Next Generation Networks). Until recently, fone of the greatest challenges in the migration from existing PSTN (Public Switched Telephone Network) toward NGN is to build a carrier class packet telephony network that preserves the ubiquity, quality, and reliability of PSTN services while allowing the greatest flexibility for use of new packet telephony technology. The SIP-T signaling system defined in IETF (Internet Engineering Task Force) draft is a mechanism that uses SIP (Session Initiation Protocol) to facilitate the interconnection of PSTN with carrier class packet telephony network. Based on IETF, the SIP-T signaling system not only promises scalability, flexibility, and interoperability with PSTN but also provides call control function of MGC (Media Gateway Controller) to set up, tear down, and manage VoIP (Voice over IP) calls in carrier class packet telephony network. In this paper, we derive the buffer size, the mean of queueing delay, and the variance of queueing delay of SIP-T signaling system that are the major performance evaluation parameters for improving QoS (Quality of Service) and system performance of MGC in carrier class packet telephony network focused on toll by-pass or tandem by-pass of PSTN. First, we assume a mathematical model of the M/G/1 queue with non-preemptive priority assignment to represent SIP-T signaling system. Second, we derive the formulas of buffer size, queueing delay, and delay variation for the non-preemptive priority queue by queueing theory respectively. Besides, some numerical examples of buffer size, queueing delay, and delay variation are presented as well. Finally, the theoretical estimates are shown to be in excellent consistence with simulation results.

This paper presents data coding techniques for a stable single-bit noise-shaping quantizer, which has a cascade structure of a multi-bit ΣΔ modulator and a binary interpolator. The binary interpolator chooses a pre-optimized binary vector for each input sample and successively generates the chosen binary vector as an output bit stream. The binary vectors can have different lengths. The paper also proposes two methods to evaluate and bound output errors of a binary interpolator. A multi-bit ΣΔ modulator is designed to cause no overload for all possible input signals whose amplitudes are bounded to a specified level, and thus the ΣΔ modulator rigorously guarantees the stability condition. In design examples, we have evaluated Signal-to-Noise and Distortion Ratios (SNDRs) and noise spectra and then confirmed that our stable quantizers can sharply shape output noise spectra.

Quality of service (QoS) technology has been implemented and started to be applied to new applications on the next-generation Internet. However, as new applications have many kinds of features and requirements, some additional features should be added to current QoS control technology. For example, they require a definition of a single policy to operate QoS control in the management domain consistently and efficiently. Policy definition for transport layer in a domain and among domains is being discussed at IETF to set a standard process, however detailed policy corresponding to the application or contents information according to the application semantics has not been discussed. Therefore we developed QoS policy control mechanism using metadata which is defined as a structured data according to the application semantics. Though metadata and transport mechanism can be located into quite different positions in the concept of network layers, we made them successfully collaborated by defining meta policy. In this paper, we describe our approach to define a meta policy based on the requirements and information contents from the application as a high level layer concept to be able to classify the network behavior. Our approach enables to multiple QoS control and collaboration among domains. We also report of the activities in IETF and ITU-T.

This paper investigates multiple-valued code-division multiple access (MV-CDMA) techniques and circuits for intra/inter-chip communication to achieve efficient data transmission in VLSI systems. To address the problems caused by interconnection complexity, we transmit multiplexed signals inside LSI systems employing pseudo-random orthogonal m-sequences as information carriers. A new class of multiple-valued CDMA techniques for intra-chip communication is discussed to demonstrate the feasibility of eliminating co-channel interference caused by a propagation delay of signals, e.g., clock skew. This paper describes the circuit configuration and performance evaluation of MV-CDMA systems for intra-chip communication. We first explain the principle of MV-CDMA technique, and then propose a bidirectional current-mode CMOS technique to realize compact correlation circuits for CDMA. Finally, we show the Spice and MATLAB simulation results of MV-CDMA systems, which indicate the excellent capabilities of eliminating co-channel interference.

Server performance is a major issue in improving the overall performance of the World Wide Web (WWW). This article introduces a dynamic mirroring-based approach to improve WWW servers' performance. In contrast to static mirroring, where mirror servers are allocated statically, our mirror servers' setup is driven by network traffic measurement. Performance in terms of latency is inferred from a queuing model. According to this model we show that latency of an overloaded server can be tuned by delegating a portion of the load to a cooperative mirror server. Cost is evaluated by the amount of load hosted by the mirror servers. The goal is then to keep the latency within a tolerable threshold, while minimizing the delegated load. This problem is formulated as a constrained optimization problem where the task is to assign a portion of load corresponding to each document to each mirror server. As the result of this work, we will have a balanced load among the servers, and a smoother traffic along the Internet, as well. Empirical results show that this approach can guarantee to maintain the performance while showing a significant decrease in the amount of load transferred to the mirror servers.

It is widely known that the family of projection filters includes the generalized inverse filter, and that the family of parametric projection filters includes parametric generalized projection filters. However, relations between the family of parametric projection filters and constrained least squares filters are not sufficiently clarified. In this paper, we consider relations between parameter estimation and image restoration by these families. As a result, we show that the restored image by the family of parametric projection filters is a maximum penalized likelihood estimator, and that it agrees with the restored image by constrained least squares filter under some suitable conditions.

A quasi-periodic signal is a periodic signal with period and amplitude variations. Several physiological signals, including the electrocardiogram (ECG), can be treated as quasi-periodic. Vector quantization (VQ) is a valuable and universal tool for signal compression. However, compressing quasi-periodic signals using VQ presents several problems. First, a pre-trained codebook has little adaptation to signal variations, resulting in no quality control of reconstructed signals. Secondly, the periodicity of the signal causes data redundancy in the codebook, where many codevectors are highly correlated. These two problems are solved by the proposed codebook replenishment VQ (CRVQ) scheme based on a bar-shaped (BS) codebook structure. In the CRVQ, codevectors can be updated online according to signal variations, and the quality of reconstructed signals can be specified. With the BS codebook structure, the codebook redundancy is reduced significantly and great codebook storage space is saved; moreover variable-dimension (VD) codevectors can be used to minimize the coding bit rate subject to a distortion constraint. The theoretic rationale and implementation scheme of the VD-CRVQ is given. The ECG data from the MIT/BIH arrhythmic database are tested, and the result is substantially better than that of using other VQ compression methods.

Intensive accesses to a web server causes its response delay and/or service suspension. We propose, to solve the problem in this paper, a service model for web servers grounded upon admission control and call waiting service, which are common approaches to designing telephone systems. Software architecture which consists of a front-end server and child servers is also designed based on the service model. Through performance evaluation for an implementation of the architecture, it is shown that the overall throughput becomes much better by keeping child web servers from overload. We drew a conclusion that the architecture enhances flexibility of a resulting software with discussions from the extensibility view.

As the number of Internet applications in critical situations increases, the quality assurance of the network infrastructure becomes more and more important. The operators of such networks have to maintain the network as reliable and available with sufficient performance. Measurement technology is the key to maintaining the network condition and provides the network as being the important infrastructure of the network society. To meet these requirements, a high-speed and accurate IP meter, HIM, has been developed. Specific hardware has been made to realize the necessary specifications that enable data capturing with 20-µsecond order timing information from a gigabit class network. The experimental results with HIM show that: 1) software-based equipment cannot handle 100-µsecond order timing information, 2) the current computer system can transmit a high-speed (30 Mpbs) multimedia stream in accurate timing (jitter less than 20 µseconds) with UDP, 3) but TCP's retransmission mechanism makes the jitter worse (up to 264 mil seconds) and, 4) WAN's one-way traffic exhibits some strange behavior, which requires further investigation.

In this paper, we propose a novel higher order time-frequency distribution (GDH) for a discrete time signal. This distribution is defined over the original discrete time-frequency grids through a delicate discretization of an equivalent expression of a higher order distribution, for a continuous time signal, in [4]. We also present a constructive design method, for the kernel of the GDH, by which the distribution satisfies (i) the alias free condition as well as (ii) the marginal conditions. Numerical examples show that the proposed distributions reasonably suppress the artifacts which are observed severely in the Wigner distribution and its simple higher order generalization.

Most of the QoS-based routing schemes proposed so far focus on improving the performance of individual service classes. In a multi-class network where high priority QoS traffic coexists with best-effort traffic, routing decision for QoS sessions will have an effect on lower ones. A mechanism that allows dynamic sharing of link resources among multiple classes of traffic is needed. In this paper we propose a multi-class routing algorithm based on inter-class sharing resources among multiple class of traffic. Our algorithm is based on the concept of "the virtual residual bandwidth," which is derived from the real residual bandwidth. The virtual residual bandwidth is greater than the residual bandwidth when the load of lower priority traffic is light, and smaller when the load of lower priority traffic is heavy. The idea of our approach is simple since the routing algorithm for individual traffic doesn't change, but the only change is the definition of the link cost. We demonstrate through some extensive simulations the effectiveness of our approach when the best effort distribution is uneven and when its load is heavy. Also better performance is noticed when using topology with large number of alternative paths.

This paper describes an effective technique for coding QCIF video sequences based on a JPEG2000 codec. In the proposed method, multiple frames are combined into one large picture. The larger picture enables images to be coded more efficiently. Image quality is further improved by combining the frames symmetrically. The video sequence is efficiently coded by adapting the time correlation of the video sequences to spatial correlation. We demonstrated the effectiveness of this method by encoding QCIF video sequences using JPEG2000.

The handoff of in Mobile IP causes packet sequence disruption during packet forwarding procedure, which may result in degradation of network performance in higher layer protocol. In this paper, we investigate the impact of handoff on out-of-sequence packets in Mobile IP with Route-Optimization extension. To evaluate the handoff procedure, we use the unstable time period (UTP) when the packet sequence could be mis-ordered. The results show that the UTP mainly depends on queuing delay of old Foreign Agent (FA) and link capacity between old FA and new FA. Further, we propose that the priority queuing for the astray packets in the old FA can increase Mobile IP performance during handoff procedure by reducing the UTP.

Future wireless communication systems will provide mobile terminals with high bit-rate transmissions for accessing broadband wired networks. In this paper, we envisage a Time Division Multiple Access - Time Division Duplexing (TDMA-TDD) air interface and we propose a Medium Access Control (MAC) protocol, named Dynamic Scheduling - TDD (DS-TDD), that allows guaranteeing the QoS of different traffic classes and efficiently supports uplink/downlink traffic asymmetries. The DS-TDD performance is theoretically analyzed. Moreover, the DS-TDD protocol is compared with another scheme proposed in the literature. Finally, the impact of packet errors on the DS-TDD performance is evaluated.

We propose a buffer management mechanism, called V-WFQ (Virtual Weighted Fair Queueing), for achieving an approximately fair allocation of bandwidth with a small amount of hardware in a high-speed network. The basic process for the allocation of bandwidth uses selective packet dropping that compares the measured input rate of the flow with an estimated fair share of bandwidth. Although V-WFQ is a hardware-efficient FIFO-based algorithm, it achieves almost ideal fairness in bandwidth allocation. V-WFQ can be implemented in the high-speed core routers of today's IP backbone networks to provide various high-quality services. We have investigated V-WFQ's performance in terms of fairness and link utilization through extensive simulation. The results of simulation show that V-WFQ achieves a good balance between fairness and link utilization under various simulation conditions.