Obtaining current traffic matrices is essential to traffic engineering (TE) methods. Because it is difficult to monitor traffic matrices, several methods for estimating them from link loads have been proposed. The models used in these methods, however, are incorrect for some real networks. Thus, methods improving the accuracy of estimation by changing routes also have been proposed. However, existing methods for estimating the traffic matrix by changing routes can only capture long-term variations and cannot obtain current traffic matrices accurately. In this paper, we propose a method for estimating current traffic matrices that uses route changes introduced by a TE method. In this method, we first estimate the long-term variations of traffic by using the link loads monitored at previous times. Then, we adjust the estimated long-term variations so as to fit the current link loads. In addition, when the traffic variation trends change and the estimated long-term variations fail to match the current traffic, our method detects mismatch. Then, so as to capture the current traffic variations, the method re-estimates the long-term variations after removing monitored data corresponding to the end-to-end traffic causing the mismatches. We evaluate our method through simulation. The results show that our method can estimate current traffic matrices accurately even when some end-to-end traffic changes suddenly.

In this letter we provide a steering law for redundant single-gimbal control moment gyros. The proposed steering law is an extended version of the singular direction avoidance (SDA) steering law based on the singular value decomposition (SVD). All internal singularities are escapable for any non-zero constant torque command using the proposed steering law.

The present paper describes quality enhancement of speech corrupted by an additive background noise in a single-channel system. The proposed approach is based on the introduction of a perceptual criterion using a frequency-weighting filter in a subtractive-type enhancement process. Although this subtractive-type method is very attractive because of its simplicity, it produces an unnatural and unpleasant residual noise. Thus, it is difficult to select fixed optimized parameters for all speech and noise conditions. A new and effective algorithm is thus developed based on the masking properties of the human ear. This newly developed algorithm allows for an automatic adaptation in the time and frequency of the enhancement system and determines a suitable noise estimate according to the frequency of the noisy input speech. Experimental results demonstrate that the proposed approach can efficiently remove additive noise related to various kinds of noise corruption.

Orthogonal Frequency Division Multiple Access (OFDMA) is the technique for the next generation wireless networks, whose enhanced capacity is to serve a combination of traffic with diverse QoS requirements. To realize this, the resource allocation scheme has to be carefully designed so that the instantaneous channel condition, QoS provision, and the network utilization are integrated. In this paper, we propose the resource allocation scheme for downlink traffic of 2 classes; guaranteed and non-guaranteed, having different traffic contracts. We provide guaranteed throughput for the guaranteed class by considering the cost incurred from serving this class. Then, we formulate the assignment problem with the objective of minimizing this cost. For the non-guaranteed class, we aim to maximize network utilization and to maintain throughput fairness, by employing Proportional Fairness (PF) utility function and emphasizing on the portion of network resource that the user received and the individual user's queue length. We use a heuristic approach to schedule users' data into the downlink subframe by exploiting multi-user multi-channel diversity to utilize system's bandwidth efficiently. Intensive simulation shows that our scheme differentiates classes of traffic and provides satisfied throughput, lower packet drop rate, and lower queuing delay to the guaranteed class, comparing with those of the non-guaranteed class. Furthermore, the results also show that the scheme is fair to users in the same class in both throughput and service time.

Traffic Engineering (TE) is important for improving QoS in forwarding paths by efficient use of network resources. In fact, MPLS allows several detour paths to be (pre-)established for some source-destination pair as well as its primary path of minimum hops. Thus, we focus on a two-phase path management scheme using these two kinds of paths. In the first phase, each primary path is allocated to a flow on a specific source-destination pair if the path is not congested, i.e., if its utilization is less than some predetermined threshold; otherwise, as the second phase, one of the detour paths is allocated randomly if the path is available. Therefore, in this paper, we analytically evaluate this path management scheme by extending the M/M/c/c queueing system, and through some numerical results we investigate the impact of a threshold on the flow-blocking probability. Through some numerical results, we discuss the adequacy of the path management scheme for MPLS-TE.

As the IEEE 802.16e Wireless Metropolitan Access Network (WMAN) supports the mobility of a mobile station (MS), increasing MS power efficiency has become an important issue. In this paper, we analyze the sleep-mode operation for an efficient power saving mechanism for delay-guaranteed services in the IEEE 802.16e WMAN and observe the effects of the operating parameters related to this operation. For the analysis we use the M/GI/1/K queueing system with multiple vacations, exhaustive services and setup times. In the analysis, we consider the power consumption during the wake-mode period as well as the sleep-mode period. As a performance measure for the power consumption, we propose the power consumption per unit time per effective arrival which considers the power consumption and the packet blocking probability simultaneously. In addition, since we consider delay-guaranteed services, the average packet response delay is also considered as a performance measure. Based on the performance measures, we obtain the optimal sleep-mode operation which minimizes the power consumption per unit time per effective arrival with a given delay requirement. Numerical studies are also provided to investigate the system performance and to show how to achieve our objective.

As a variety of digital services are provided through networks, more and more efforts are made to ensure dependability of software behavior implementing services. Formal methods and tools have been considered as promising means to support dependability in complex software systems during the development. On the other hand, there have been serious doubts on practical applicability of formal methods. This paper overviews the present state of formal methods and discusses their applicability, especially focusing on two representative methods (SPIN and B Method) and their recent industrial applications. This paper also discusses applications of formal methods to dependable networked software.

The expansion and diversification of wireless systems are proceeding rapidly, and the wireless communication environment will become a hybrid environment composed of multiple wireless systems. On the other hand, the resources of wireless systems are finite. Therefore, there is increasing concern that the growing use ofwireless systems will exhaust the finite wireless resources. Cognitive radio has been proposed as a solution to this problem. Cognitive radio aims to optimize the utilization of wireless resources by combining multiple wireless interfaces. Therefore, the future wireless access networks will be composed of multiple wireless systems. In this paper, we propose a packet distribution for communications using multiple IEEE802.11 wireless interfaces. The proposed method optimizes the utilization efficiency of multiple IEEE802.11 wireless interfaces based on a link cost which shows the load of the wireless link, and it improves delay and throughput in the access network. Furthermore, we show its impact on TCP.

This letter proposes a novel method of large-scale IP traffic matrix (TM) estimation, called algebraic reconstruction technique inference (ARTI), which is based on the partial flow measurement and Fratar model. In contrast to previous methods, ARTI can accurately capture the spatio-temporal correlations of TM. Moreover, ARTI is computationally simple since it uses the algebraic reconstruction technique. We use the real data from the Abilene network to validate ARTI. Simulation results show that ARTI can accurately estimate large-scale IP TM and track its dynamics.

IEEE 802.11 Media Access Control (MAC) is the most widely used protocol in Wireless Ad-hoc Networks (WANETs). However, it cannot guarantee fair channel allocation among the flows contending for the channel. In this paper, we propose a new media access control algorithm to achieve per-flow fairness while maximizing WANETs throughput.

Improving the overall performance of reliable speech communication in ultrashort wave radios over very noisy channels is of great importance and practical use. An iterative joint source-channel (de-)coding and (de-)modulation (JSCCM) algorithm is proposed for ITU-T Rec.G.729EV by both exploiting the residual redundancy and passing soft information throughout the receiver while introducing a systematic global iteration process. Being fully compatible with existing transmitter structure, the proposed algorithm does not introduce additional bandwidth expansion and transmission delay. Simulations show substantial error correcting performance and synthesized speech quality improvement over conventional separate designed systems in delay and bandwidth constraint channels by using the JSCCM algorithm.

In this letter, a two channel frequency domain speech enhancement algorithm is proposed. The algorithm is designed to achieve better overall performance with relatively small array size. An improved version of adaptive null-forming is used, in which noise cancelation is implemented in auditory subbands. And an OM-LSA based postfiltering stage further purifies the output. The algorithm also features interaction between the array processing and the postfilter to make the filter adaptation more robust. This approach achieves considerable improvement on signal-to-noise ratio (SNR) and subjective quality of the desired speech. Experiments confirm the effectiveness of the proposed system.

This letter proposes an identity-based authenticated key agreement protocol. Different from available comparable ones, the new protocol realizes implicit authentication without bilinear pairings which makes it more efficient. The security of proposed protocol can be reduced to the standard Computational Diffie-Hellman problem. Two variants of the protocol are also given, with one achieving the security-efficiency trade-off and the other providing authenticated key agreement between users of different domains.

This letter analyzes the outage probability of limited feedback beamforming systems with receive antenna selection. Tight analytical closed-form expressions of outage performance are derived for both cases, with and without spatial fading correlation, which allow for evaluation of the performance as a function of the codebook size, the level of fading correlation, and the number of transmit and receive antennas. Simulation results are also provided to verify the analysis.

In this study we propose a manufacturing control framework based on radio-frequency identification (RFID) technology and a distributed information system to construct a mass-customization production process in a loosely coupled shop-floor control environment. On the basis of this framework, we developed RFID middleware and an integrated information system for tracking and controlling the manufacturing process flow. A bicycle manufacturer was used to demonstrate the prototype system. The findings of this study were that the proposed framework can improve the visibility and traceability of the manufacturing process as well as enhance process quality control and real-time production pedigree access. Using this framework, an enterprise can easily integrate an RFID-based system into its manufacturing environment to facilitate mass customization and a just-in-time production model.

Many p2p based wide-area storage networks have been proposed to provide scalable storage services by combining the idle resources of many unreliable nodes. These storage networks can also provide highly available and reliable storage services, by replicating each data on several nodes. The popular approach is availability based replication which uses individual node availability. However, some replicas leave within a short time under high churn in p2p networks. This results in heavy and bursty data traffic, and sometimes some data are lost. This paper presents the lifetime-aware replication which uses the lifetime of each node to prevent the bursty failures and the data loss. It keeps a primary replica which has enough time to replace a lost redundancy. It also spreads replicas on the timeline to reduce the overlapped replicas as best as it can. Results from event-driven simulations show that the lifetime-aware replication keeps high data durability with less data traffic.

A new online, unsupervised voice activity detection (VAD) method is proposed. The method is based on a feature derived from high-order statistics (HOS), enhanced by a second metric based on normalized autocorrelation peaks to improve its robustness to non-Gaussian noises. This feature is also oriented for discriminating between close-talk and far-field speech, thus providing a VAD method in the context of human-to-human interaction independent of the energy level. The classification is done by an online variation of the Expectation-Maximization (EM) algorithm, to track and adapt to noise variations in the speech signal. Performance of the proposed method is evaluated on an in-house data and on CENSREC-1-C, a publicly available database used for VAD in the context of automatic speech recognition (ASR). On both test sets, the proposed method outperforms a simple energy-based algorithm and is shown to be more robust against the change in speech sparsity, SNR variability and the noise type.

To be adaptive to the bursty traffic of the wireless personal area network (WPAN), a superframe structure adjustment algorithm of IEEE 802.15.4 is proposed. According to the channel utilization ratio which is an index of the traffic load, the algorithm adjusts the duty cycle of the superframe automatically. The simulation results show that the algorithm is adaptive to the traffic variations effectively and saves much more energy not only for the end devices but also for the PAN coordinator. Moreover, the algorithm results in better performance on the lower delay and lower packet dropping rate.

We propose multitone-hopping code-division multiple access (MH-CDMA) using a feedback-controlled hopping pattern (FCHP) (FCHP/MH-CDMA). In the FCHP/MH-CDMA, part of the filter coefficients of an adaptive finite-duration impulse response (FIR) filter receiver are fed back to a transmitter, in which they are used as an updated hopping pattern. Each chip of the updated hopping pattern consists of plural tones. As a result, it is shown that the FCHP/MH-CDMA provides us with an excellent asynchronous, decentralized multiple-access performance over time-invariant multipath channels.

Recently, real-time media delivery services such as video streaming and VoIP have rapidly become popular. For these applications requiring high-level QoS guarantee, our research group has proposed a transport-layer approach to provide predictable throughput for upper-layer applications. In the present paper, we propose a congestion control mechanism of TCP for achieving predictable throughput. It does not mean we can guarantee the throughput, while we can provide the throughput required by an upper-layer application at high probability when network congestion level is not so high by using the inline network measurement technique for available bandwidth of the network path. We present the evaluation results for the proposed mechanism obtained in simulation and implementation experiments, and confirm that the proposed mechanism can assure a TCP throughput if the required bandwidth is not so high compared to the physical bandwidth, even when other ordinary TCP (e.g., TCP Reno) connections occupy the link.