This paper considers the reliable multicast transport protocols used in hybrid networks that include wired and wireless networks and transparent proxy servers. We present four analytic performance models of two extreme reliable multicast transport protocols, sender-initiated and receiver-initiated, and supported and unsupported by transparent proxy servers are considered in each reliable multicast protocol. We analyze the throughputs of these four different models mathematically. Numerical results show that transparent proxy servers give good effects to overall performance. Furthermore, the receiver-initiated reliable multicast supported by transparent proxy servers gives better performances of total throughput than sender-initiated reliable multicast supported by transparent proxy servers. We provide efficiency criterion of the optimal number of transparent proxy servers for each protocol under varying wireless loss probabilities. Numerical results are verified by simulations.

Since their inception almost fifty years ago, hidden Markov models (HMMs) have have become the predominant methodology for automatic speech recognition (ASR) systems--today, most state-of-the-art speech systems are HMM-based. There have been a number of ways to explain HMMs and to list their capabilities, each of these ways having both advantages and disadvantages. In an effort to better understand what HMMs can do, this tutorial article analyzes HMMs by exploring a definition of HMMs in terms of random variables and conditional independence assumptions. We prefer this definition as it allows us to reason more throughly about the capabilities of HMMs. In particular, it is possible to deduce that there are, in theory at least, no limitations to the class of probability distributions representable by HMMs. This paper concludes that, in search of a model to supersede the HMM (say for ASR), rather than trying to correct for HMM limitations in the general case, new models should be found based on their potential for better parsimony, computational requirements, and noise insensitivity.

Properties of the quantum noise and the optical feedback noise in blue-violet InGaN semiconductor lasers were measured in detail. We confirmed that the quantum noise in the blue-violet laser becomes higher than that in the near-infrared laser. This property is an intrinsic property basing on principle of the quantum mechanics, and is severe subject to apply the laser for optical disk with the small consuming power. The feedback noise was classified into two types of "low frequency type" and "flat type" basing on frequency spectrum of the noise. This classification was the same as that in the near infra-red lasers.

This paper describes a BIST circuit for testing SoC integrated multi-channel serializer/deserializer (SerDes) macros. A newly developed packet-based PRBS generator enables the BIST to perform at-speed testing of asynchronous data transfers. In addition, a new technique for chained alignment checks between adjacent channels helps achieve a channel-count-independent architecture for verification of multi-channel alignment between SerDes macros. Fabricated in a 0.13-µm CMOS process and operating at > 500 MHz, the BIST has successfully verified all SerDes functions in at-speed testing of 5-Gbps20-ch SerDes macros.

This letter analyzes the performance of statistical cooperative diversity based on space-time block codes (STBC) (Statistical STBC cooperative diversity) considering the effects of quantization and observation noise. Binary quantization is used. The bit error rate (BER) and average mutual information of the statistical STBC cooperative diversity with Alamouti's STBC and two active nodes are derived in the presence of general observation noise. It is shown that the performance of the statistical STBC cooperative diversity depends on the effects of observation noise and the number of cooperating nodes largely. It is also shown how much the communication between sensor nodes or feedback from the fusion center improves the performance of STBC cooperative diversity.

This paper describes a novel pilot symbol aided up-link channel estimation scheme for a multi-user MIMO-OFDM system. A novel pilot-symbol pattern is proposed in order to overcome the interference from the multiple antennas of each user. Based on these periodically inserted pilot symbols, the channel state information (CSI) for each entire OFDM data sequence is reconstructed by using the maximum a-posteriori probability (MAP) estimation algorithm. The MAP estimation algorithm exploits channel correlations in time, frequency and space domains, which are obtained from a frequency-selective and time-variant Rayleigh fading channel model with multiple clusters and a defined complex direction of arrival (DOA). Simulation results demonstrate that it achieves almost the same performance as the ideal case by using the MAP-based estimation scheme with the well designed pilot-symbol pattern. Moreover, this model-based estimation scheme is also robust to errors in the estimation of its parameters. It will become one of the strong candidates for use in next generation mobile communication systems.

In this letter, an illustrative example is given, which shows that the number of integrators needed for the dynamic observer error linearization using integrators can not be bounded by a function of the dimension of the system and the number of outputs in contrast to dynamic feedback linearization results.

In this letter, we propose adaptive linear detectors in space-time block coded multiuser systems, by exploiting a particular property of the minimum mean square error multiuser detector. The proposed scheme can provide much faster convergence than the existing adaptive scheme [5] and so lower the system overhead requirements.

A statistic based algorithm for watermarking 2D vector maps is proposed. Instead of 2D coordinates, a one-dimensional distance sequence extracted from the original map is used as the cover data to achieve the shape-preserving ability. The statistical feature of the cover data is utilized for data embedding. Experiment results indicate the scheme's better performance in invisibility, as well as its robustness to certain attacks.

Sequences with ear zero correlation zones (EZCZs) are employed to suppress inter-symbol interference (ISI) and inter-user interference (IUI) in wireless communications. Theoretical limits on correlation functions of such sequences are investigated, lower bounds on the relations among length of sequence, width of EZCZs/ELCZs and family size are derived and presented, which play an important role in assessing performance of such sequences.

Admission control is becoming an essential technique for IP networks to provide full-fledged multimedia streaming services. Although signaling-based schemes are utilized to achieve this, these are difficult to deploy and can hardly achieve strict admission control taking the properties of packet arrival into consideration. In this paper, we propose a novel admission control strategy called the Tentative Accommodating and Congestion Confirming Strategy (TACCS). The main idea is to accommodate incoming flows tentatively and confirm congestion after a certain period. Since tentative accommodating enables us to generate the same situation as where incoming flows have been accommodated, TACCS makes it possible to control admission considering the properties of packet arrival after they have been accommodated, without collecting resource information in advance. From the results of mathematical analysis, we confirmed that TACCS enabled a domain to control admission without a centralized management agent and we provided guidelines for configuring parameters of TACCS.

In 2000, Sandirigama, Shimizu, and Noda proposed a simple password authentication scheme, SAS. However, SAS was later found to be flawed. Recently, Chen, Lee, Horng proposed two SAS-like schemes, which were claimed to be more secure than similar schemes. Herein, we show that both their schemes are still vulnerable to denial-of-service attacks. Additionally, Chen-Lee-Horng's second scheme is not easily reparable.

This paper analyzes the user-perceived QoS (Quality of Service) of interactive data application, the most notable being web browsing, in a shared wireless packet access network. We present elaborate analytical results regarding the mean page delay, the Equivalent Circuit Rate (ECR), and the system saturation point for interactive data users with different channel conditions. The ECR for a user in a shared access network is the channel rate that a dedicated connection would need in order to offer an equivalent user experience. The system saturation point is the maximum number of users that can be served with a given tolerable service delay in a shared packet access system. These analytical results are useful in the performance prediction of a real wireless environment with multiple user groups that are served with different service rates due to multiple channel conditions. Therefore, our analytical results can be used in the network planning, admission control, and service differentiation for different user groups. Our analysis is demonstrated to be in good accord with computer simulations performed on model systems based on EDGE TDMA and IEEE 802.16 OFDMA.

Popular Web sites form their Web servers into Web server clusters. The Web server cluster operates with a load-balancing algorithm to distribute Web requests evenly among Web servers. The load-balancing algorithms founded on conventional periodic load-information update mechanism are not scalable due to the synchronized update of load-information. We propose a load-balancing algorithm that the load-information update is not synchronized by exploiting variant execution times of executing scripts in dynamic Web pages. The load-information of each server is updated 'individually' by a new load-information update mechanism, and the proposed algorithm supports high scalability based on this individual update. Simulation results have proven the improvement in system performance through another aspect of high scalability. Furthermore, the proposed algorithm guarantees some level of QoS for Web clients by fairly distributing requests. A fundamental merit of the proposed algorithm is its simplicity, which supports higher throughput of the Web switch.

We propose the concept of an opinion model for interactive multimodal services and apply it to an audiovisual communication service. First, psychological factors of an audiovisual communication service were extracted by using the semantic differential (SD) technique and factor analysis. Forty subjects participated in subjective tests and performed point-to-point conversational tasks on a PC-based video phone that exhibited various network qualities. The subjects assessed those qualities on the basis of 25 pairs of adjectives. Two psychological factors, i.e., an aesthetic feeling and a feeling of activity, were extracted from the results. Then, quality impairment factors affecting these two psychological factors were analyzed. We found that the aesthetic feeling was affected by IP packet loss and video coding bit rate, and the feeling of activity depended on delay time, video packet loss, video coding bit rate, and video frame rate. Using this result, we formulated an opinion model derived from the relationships among quality impairment factors, psychological factors, and overall quality. The validation test results indicated that the estimation error of our model was almost equivalent to the statistical reliability of the subjective score.

This paper proposes a framework for session-level SLA (Service Level Agreement) and network-level SLA management to provide QoS-oriented application services over DiffServ/MPLS networks. DiffServ and MPLS technologies enhance the capability of QoS guarantee on IP network, and application service provider can provide QoS-oriented application services to their customers based on the transport networks. The example of using our approaches in the VoIP service involving the network service provider, the VoIP service provider, and the VoIP customers are examined. The session-level SLA between VoIP service provider and VoIP customer and the network-level SLA (N-SLA) between network provider and VoIP service provider are constructed in this paper. Besides, the VoIP service provider faces the QoS-mapping issue and the balance between revenue and cost, i.e., how to contract the N-SLA. Therefore, we introduce an off-line SLA evaluation scheme, a heuristic optimization algorithm and an on-line SLA process method to provide VoIP service policy, and then the optimal QoS-mapping can be resolved. The concept of this framework of session-level SLA and network-level SLA management can be extended easily into other real-time multimedia and non-real time data services.

Changes in recent business and scientific environment have created a necessity for more efficient and effective workflow infrastructure. With increasing emphasis on Service-oriented architecture, service composition becomes a hot topic in workflow research. This paper proposes a novel approach of using ECA rules to realize the workflow modeling and implementation for service composition. First of all, the concept and formalization of ECA rule-based workflow is presented. Special activities and data structures are customized for the purpose of service composition. Second, an automatic event composition and decomposition algorithm is developed to ensure the correctness and validness of service composition at design time. Finally, the proposed ECA rule-based approach for service composition is illustrated through the implementation of a workflow prototype system.

In the floorplan design of System-on-Chip (SOC), Buffer Site Approach (BSA) has been used to relax the buffer congestion problem. However, for a floorplan with dominant wide bus, BSA may instead worsen the congestion. Our proposed Enhanced Buffer Site Approach (EBSA) extends existing BSA in a way that buffers of dominant wide bus can be distributed more evenly while reserving the same fast operation speed as BSA does. Experiments have been performed to integrate our model into an iterative floorplanning algorithm, and the results reveal that buffer congestion in a floorplan with dominant wide bus can be much abated.

New service concepts involving mobile devices with a diverse range of embedded sensors are emerging that share contexts supporting communication on a wireless network infrastructure. To promote these services in mobile devices, we propose a method that can efficiently detect a context provider by partitioning the location, time, speed, and discovery sensitivities.

This paper discusses the least-squares linear filtering and smoothing (fixed-point and fixed-interval) problems of discrete-time signals from observations, perturbed by additive white noise, which can be randomly delayed by one sampling time. It is assumed that the Bernoulli random variables characterizing delay measurements are correlated in consecutive time instants. The marginal distribution of each of these variables, specified by the probability of a delay in the measurement, as well as their correlation function, are known. Using an innovation approach, the filtering, fixed-point and fixed-interval smoothing recursive algorithms are obtained without requiring the state-space model generating the signal; they use only the covariance functions of the signal and the noise, the delay probabilities and the correlation function of the Bernoulli variables. The algorithms are applied to a particular transmission model with stand-by sensors for the immediate replacement of a failed unit.