Multiple access interference (MAI) due to many simultaneous users is the main factor that limits the performance of DS-CDMA system. Multiuser detection is a method to avoid performance degradation due to MAI. We propose a blind multiuser detection method based on the algorithm consisting of two-stage decoding process, i.e., linearly constrained constant modulus (LCCM) and iterative least squares (ILS). The computer simulations confirmed that the algorithm is near-far resistant and that the proposed method is effective in the application to the slow fading channels.

We investigated the suppression of multiuser interference in uplink multicarrier CDMA systems using the minimum mean squared error combining (MMSEC) method. In MMSEC, many pilot symbols are required to converge the weight vectors, and if we use just a few pilot symbols, the performance cannot be improved very much. We therefore developed a method for calculating weight vectors for MMSEC that uses just a few pilot symbols. The impulse responses of all users are first estimated using the pilot symbols in the time domain and modulated by a discrete Fourier transform. Next, the correlation matrices and correlation vectors are estimated from the impulse responses and the spreading codes of all users. Finally, the weight vectors that are obtained from the correlation matrices and correlation vectors are multiplied by the received signal to suppress the multiuser interference. The results of computer simulations indicated that the bit-error-ratio performance obtained using this method was better than that obtained when using the conventional fading compensation scheme or when using conventional MMSEC with the recursive least squares algorithm.

This paper proposes a stable rate allocation algorithm for ABR service in ATM networks. The main goals in designing this algorithm are to speed up the convergence according to the max-min fairness criterion and to maximize the network utilization while the switch queue length can be properly controlled. Importantly, the set goals should be achieved in a wide range of network conditions without the need for adjusting the algorithm parameters. The algorithm is targeted to work in various networking environments with additional criteria as extended from the work of E-FMMRA (Enhanced Fast Max-Min Rate Allocation) and ERICA+ (Explicit Rate Indication for Congestion Avoidance) . The additional design criteria include the ability to enhance a large number of ABR connections and staggered TCP connections as well as to perform an accurate traffic averaging. The algorithm is analytically proved to be convergent. Simulation results indicate that the proposed algorithm achieves the goals in all evaluated configurations. However, it has some limitations when working in the large-scale network due to its per-connection accounting. It is not recommended to implement the algorithm with a switch that has a small buffer size due to its relatively long averaging interval.

We have experimentally observed the dynamics of laser outputs in a microchip laser with frequency-modulated feedback light. We have observed rich dynamics that can be interpreted as three types of dynamics at different frequencies: (1) periodic bursts at twice of the frequency of the modulation of the external mirror, (2) chaotic pulsations at the relaxation oscillation frequency, and (3) clusters at the instantaneous Doppler-shifted frequency. We have confirmed these rich dynamics by using numerical simulations.

We propose and analyze a new efficient handoff scheme called Splitted-Rating Channel Scheme in UMTS networks, and we analyze the call level performance of splitted-rating channel scheme and then packet level performance of downlink traffic at UMTS circuit-switched networks. In order to reduce the blocking probability of originating calls and the forced termination probability of handoff calls, a splitted-rating channel scheme is applied to the multimedia UMTS networks. This multimedia network supports two classes of calls; narrowband call requiring one channel and wideband call requiring multiple channels. The channels in service for wideband call are splitted its channels for lending to originating call and handoff call according to threshold control policy. By assuming that arrivals of narrowband calls and arrivals of wideband calls are Poisson, we model the number of narrowband calls and the number of wideband calls in the one cell by Level Dependent Quasi-Birth-Death (QBD) process and obtain their joint stationary distribution. For packet level analysis, we first describe the downlink traffic from the base station to a mobile terminal in UMTS networks, and calculate the mean packet delay of a connected wideband call by using QBD analysis. Numerical examples show that our splitted-rating channel scheme reduces the blocking probability of originating call and the forced termination probability of handoff call with a little degradation of packet delay.

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 letter proposes a packet length-based group-wise transmission (LGT) rate scheduling scheme for non-real time data service for the uplink of direct sequence code division multiple access (DS/CDMA) system using the burst switching scheme to support the integrated voice/data service. The LGT scheme optimally determines two different rate groups and their optimal data rates so as to minimize the average packet transmission delay. It has shown that the packet transmission delay performance can be significantly improved over the conventional single-rate packet transmission scheme for integrated voice/data service. Furthermore, a main feature of the proposed scheme is simplicity in its implementation.

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.

The effect of subarray size (equal to the order of the prediction model plus one) on the estimation performance of a previously proposed forward-backward linear prediction (FBLP) based cyclic method is investigated. This method incorporates an overdetermined FBLP model with a subarray scheme and is used to estimate the directions-of-arrival (DOAs) of coherent cyclostationary signals impinging on a uniform linear array (ULA) from the corresponding polynomial or spectrum formed by the prediction coefficients. However, the decorrelation is obtained at the expense of a reduced working array aperture, as it is with the spatial smoothing (SS) technique. In this paper, an analytical expression of the mean-squared-error (MSE) of the spectral peak position is derived using the linear approximation for higher signal-to-noise ratio (SNR). Then the subarray size that minimizes this approximate MSE is identified. The effect of subarray size on the DOA estimation is demonstrated and the theoretical analysis is substantiated through numerical examples.

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.

In this paper, a new adaptive method is suggested using the complementally transformed minimum variance technique for the purpose of suppressing interference in additive white and colored Gaussian noise channels. The method is based on interference suppression by way of the resulting projection weight. The multiple access causes an interference problem in the code-division multiple access systems. An efficient adaptive algorithm should be used to suppress this interference for the improvement of system performance. Analytical and simulation results show that the new adaptive method has fast convergence rate and offers significant performance gain over the conventional detector, the MMSE detector, and the linear decorrelator. Finally, multipath fading induced performance loss, which leads to error probability floor, is established for the proposed method with combining schemes and shown by computer simulation.

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.

In this paper, a novel continuous complementary sliding control was proposed to improve the tracking performance given the available control bandwidth and the extend of parameter uncertainty. With this control law, the ultimate bound of tracking error was shown to be reduced at least by half, as compared with the conventional continuous sliding control. More strikingly, the proposed control can effectively improve the error transient response during the reaching phase. We presented a composite complementary sliding control scheme for a class of uncertain nonlinear systems including the nonlinear electrohydraulic position servo control system, which will be used as an illustrated example. Simulation results indicated exceptional good tracking performance to step and sine wave reference inputs can be obtained. In addition, the disturbance rejection property of the controller to single-frequency sinusoidal disturbances is also outstanding.

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.

For future adaptive networking services, it will be effective if autonomous agents utilize widely spread information that continuously changes. A communication network should provide a framework for agents to utilize information in a large-scale network and should positively support communications among agents. This paper proposes EcAgent (Environment for Communities of Agents). EcAgent provides agents with access to communities that are loose collections of agents. It gathers global information about communications among all agents and makes the agents share it in communities. It makes possible adaptive networking services that use various types of information. We implemented the EcAgent execution environment and some experimental applications to make an adaptive networking service. We also implemented a simulation system to evaluate how accurately the library recognized the embedded relationships as communities. The results of experiments show that the proposed methods for community recognition work efficiently.

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.

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.

In this paper, we propose a new fair queueing algorithm to improve cell delay variation (CDV) for real-time service categories and to make efficient use of system resources for multimedia traffic in high speed ATM networks. The proposed algorithm is called the delay variation-based fair queueing (DVFQ) algorithm, which is based on per-VC queueing to improve CDV and fairness for each VC of real-time services such as CBR and rt-VBR. In DVFQ algorithm, we define two fairness indexes, which indicate the degree of the fairness of CDV at the rate of each VC, and the degree of impartially sharing the bandwidth between the scheduled cells for each VC. The simulation results for both heavily and lightly loaded conditions show that DVFQ algorithm provides better performances in terms of the CDV, the CDV fairness, and the service fairness than those of FCFS for real-time service.

This paper presents an efficient scheme for access bandwidth control for VBR (Variable Bit Rate) traffic between radio mobile terminals and their base stations in a WATM (Wireless ATM) network. After introducing the wireless ATM system model, we describe a new algorithm that enables dynamic slot allocation under TDMA/TDD (Time Division Multiple Access/Time Division Duplex) Media Access Control, making use of UPC (Usage Parameter Control) parameters and traffic characteristics. We show more efficient bandwidth utilization with our proposed algorithm, compared to other conventional algorithms. Moreover, we reveal that our algorithm improves cell transmission delays.

A Multiwavelength Brillouin/Erbium fibre laser which operates in long-wavelength band (L-band) is demonstrated with an approximate comb spacing of 10 GHz. The laser configuration consists of a long EDF to enable the L-band amplification and two 3-dB couplers to take a portion of the generated BEFL signal and re-inject it into the SMF to seed a cascaded BEFL line in the same direction as the first BEFL line. 20 lines including the anti-Stokes are obtained with the maximum Brillouin pump and 980 nm-pump power of 8.9 mW and 92 mW, respectively.