We propose a probing-based channel adaptive video streaming method to provide seamless video transmission over the wireless 3G network. In the proposed method, the available bandwidth (AB) of the end-to-end path is estimated by analyzing RTP packets at the streaming client and the network/client buffer status (NCBS) is predicted by examining the RTCP receiver report (RR) and the application defined packet (APP) including the AB information. Using the estimated AB, the NCBS, and the stored multirate bitstream information, the proposed streaming method determines the next transmission bitrate precisely. Experimental results indicate that the proposed method can be effectively used for video streaming.

We present a design procedure of a leaky-wave antenna with low sidelobes based on the stub-loaded ridge-rectangular waveguide. As a typical example, we desig the antenna with the Taylor distribution of -30 dB sidelobes and fabricated it. The agreement between the measured and the numerical results validate the proposed antenna.

A false lock free delay-locked loop(DLL) achieving a wide frequency operation and a fine timing resolution is presented. A novel false lock detection technique is proposed to solve the trade-off between a wide frequency range and false locks. This technique enables a fine timing resolution even at a high frequency. In addition, the duty cycle of the input clock is not required to be 50%. This technique is applied to the DLLs in analog front-end LSIs of digital camera systems, with a range of 465 MHz (16) and a timing resolution of 9(40 stages).

By applying the differential space-time block code (DSTBC) to wireless multicarrier transmission, Diggavi et al. were the first to propose the two-input-multiple-output (2IMO) differentially space-time-time block coded OFDM (TT-OFDM) system. In this paper, we propose three novel differentially transmit-diversity block coded OFDM (DTDBC-OFDM) systems, namely, the FT-, FF-, and TF-OFDM systems. For instance, the TF-OFDM stands for the differentially space-time-frequency block coded OFDM. Moreover, the noncoherent maximum-likelihood sequence detector (NSD), and its three special cases, namely, the noncoherent one-shot detector, the linearly predictive decision-feedback (DF) detector, and the linearly predictive Viterbi receiver are incorporated to the 2IMO DTDBC-OFDM systems. Furthermore, a simple closed-form BER expression for the systems utilizing the noncoherent one-shot detector in the time-varying multipath Rayleigh fading channels is given. Numerical results have revealed that 2IMO DTDBC-OFDM systems employing the noncoherent one-shot detector can obtain significant performance improvement. However, when few antennas are available, the implementation of the linearly predictive DF detector or the linearly predictive Viterbi receiver is necessary for achieving better performance.

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 this letter, we study the capacity of fading channels with perfect channel side information (CSI) at the receiver and quantized CSI at the transmitter. We present a general algorithm for the joint design of optimal quantization and power control for maximizing the forward link capacity over flat fading channels. Numerical results for Rayleigh fading are given.

In this letter, a novel channel estimation method is proposed for frequency-domain equalization of OFDM systems in fast fading multipath channels. It is shown by computer simulations that the proposed method can not only estimate the channel impulse response (CIR) accurately but also achieve lower BER than conventional method at low signal-to-noise ratio (SNR).

This paper proposes a subjective model for estimating the quality of video streaming services with dynamic bit-rate control. In a subjective quality assessment test, we clarify users' perceptions of distributed video signals whose quality is time-variant due to dynamic bit-rate control. Using this result, we constructed an estimation model considering the following three characteristics: 1) the influence of the video section where quality degradation is large will strongly affect the overall quality, 2) the impression of a past quality weakens with the passage of time, and 3) the range of evaluation scores becomes wider when the time duration of an evaluation is longer. We found that the proposed model enables the accuracy of estimating overall subjective quality to be dramatically improved compared with that of a model that averages segmental quality. The estimation error of the proposed model is less than the statistical reliability of the subjective score even for verification data. We also show that our findings are applicable to QoS design/management issues for video streaming services with dynamic bit-rate control.

With the increasing demand for IP telephony services using Voice over IP (VoIP) technology, techniques for monitoring speech quality in actual networks are required to manage the quality of VoIP services constantly. Since the speech quality of VoIP is affected by IP network performance factors, non-intrusive methods of monitoring the quality of service (QoS) by passively measuring network performance are being watched with keen interest. VQmon technology is one of the non-intrusive quality monitoring methods. Although the monitoring functions of the VQmon for post-arrived packet behavior events at VoIP-gateways are effective, the estimating algorithm does not take differences in the implementations of VoIP-gateway products into account. We therefore propose a non-intrusive method of monitoring QoS that works in conjunction with ITU-T Recommendation P.862 "PESQ" that takes the characteristics of VoIP-gateway products into consideration. We compared the performance of non-intrusive quality monitoring technology such as VQmon and the proposed method in terms of estimating the accuracy of speech quality and mouth-to-ear delay. The experimental results revealed that the proposed method outperforms the conventional one, achieving sufficient accuracy for quality monitoring of VoIP services.

Seamless interconnection with wireless LAN and 3G technologies is essential for the future wireless environment. The Wireless Local Area Network (WLAN) integrated mobile device is designed to extend the reach of enterprise applications and to create new collaboration environments. Also rapid new service development has started to change traffic mixes in the cellular networks towards IP dominating carriers like GPRS (General Packet Radio Service) and WLAN. These drastic changes require new research in the network as well as system inter-working areas for both the cellular and WLAN technology areas. This is the main challenge our research is trying to solve giving answers to rising inter-working and interoperability questions.

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.

For alleviating the high peak-to-average power ratio (PAPR) problem of orthogonal frequency division multiplexing (OFDM), the OFDM combined with time division multiplexing (TDM) using frequency-domain equalization (FDE) was proposed. In this paper, the theoretical bit error rate (BER) analysis of the OFDM/TDM in a frequency-selective fading channel is presented. The conditional BER expression is derived, based on a Gaussian approximation of the inter-symbol interference (ISI) arising from channel frequency-selectivity, for the given set of channel gains. Various FDE techniques as in multi-carrier code division multiple access (MC-CDMA), i.e., zero forcing (ZF), maximum ratio combining (MRC) and minimum mean square error (MMSE) criteria are considered. The average BER performance is evaluated by Monte-Carlo numerical computation method using the derived conditional BER expression.

This paper introduces a State Dependent Random Early Detection (SDRED) scheme that can improve delay and jitter performances by adjusting RED parameters such as maxth and wq according to queue status. The SDRED is designed to adapt to traffic conditions by adjusting the maximum threshold and queue weight to four different levels. From the simulation results, we show that the SDRED decreases queuing delay and has more stable jitter characteristics than the existing RED, BLUE, ARED and DSRED schemes.

Differentiated-services model has been prevailed as a scalable solution to provide quality of service over the Internet. Many researches have been focused on per hop behavior or a single domain behavior to enhance quality of service. Thus, there are still difficulties in providing the end-to-end guaranteed service when the path between sender and receiver includes multiple domains. Furthermore differentiated-services model mainly considers quality of service for traffic aggregates due to the scalability, and the quality of service state may be time varying according to the network conditions in the case of relative service model, which make the problem more challenging to guarantee the end-to-end quality-of-service. In this paper, we study class mapping mechanisms along the path to provide the end-to-end guaranteed quality of service with the minimum networking price over multiple differentiated-services domains. The proposed mechanism includes an effective implementation of relative differentiated-services model, quality of service advertising mechanism and class selecting mechanisms. Finally, the experimental results are provided to show the performance of the proposed algorithm.

This paper presents a way of using a linear regression model to produce a single-valued criterion that indicates the perceived importance of each block in a stream of speech blocks. This method is superior to the conventional approach, voice activity detection (VAD), in that it provides a dynamically changing priority value for speech segments with finer granularity. The approach can be used in conjunction with scalable speech coding techniques in the context of IP QoS services to achieve a flexible form of quality control for speech transmission. A simple linear regression model is used to estimate a mean opinion score (MOS) of the various cases of missing speech segments. The estimated MOS is a continuous value that can be mapped to priority levels with arbitrary granularity. Through subjective evaluation, we show the validity of the calculated priority values.

This paper introduces a network-aware video delivery framework where the quality-of-service (QoS) interaction between prioritized packet video and relative differentiated service (DiffServ) network is taken into account. With this framework, we propose a dynamic class mapping (DCM) scheme to allow video applications to cope with service degradation and class-based resource constraint in a time-varying network environment. In the proposed scheme, an explicit congestion notification (ECN)-based feedback mechanism is utilized to notify the status of network classes and the received service quality assessment to the end-host applications urgently. Based on the feedback information, DCM agent at ingress point can dynamically re-map each packet onto a network class in order to satisfy the desired QoS requirement. Simulation results verify the enhanced QoS performance of the streaming video application by comparing the static class-mapping and the class re-mapping based on loss-driven feedback.

In everyday life, a situation often occurs wherein two or more persons with different personal schedules must determine a single job schedule. The authors focus on the practical concept of rent and loan and propose a scheduling system. This system generates a schedule that automatically coordinates with a state involving minimum rent and loan. They also propose a method that employs the analytic network process (ANP) for setting individual priorities based on the rent and loan information. Furthermore, the authors implement the proposed system as a simulation system and verify whether it generates a fair schedule by computing the sum of the rent and loan of different individuals. The result shows that in comparison with human scheduling, the proposed method generates a fairer schedule by computing the rent and loan of each individual.

XML query processing is one of the most active areas of database research. Although the main focus of past research has been the processing of structural XML queries, there are growing demands for a full-text search for XML documents. In this paper, we propose XICS (XML Indices for Content and Structural search), which aims at high-speed processing of both full-text and structural queries in XML documents. An important design principle of our indices is the use of a B+-tree. To represent the structural information of XML trees, each node in the XML tree is labeled with an identifier. The identifier contains an integer number representing the path information from the root node. XICS consist of two types of indices, the COB-tree (COntent B+-tree) and the STB-tree (STructure B+-tree). The search keys of the COB-tree are a pair of text fragments in the XML document and the identifiers of the leaf nodes that contain the text, whereas the search keys of the STB-tree are the node identifiers. By using a node identifier in the search keys, we can retrieve only the entries that match the path information in the query. The STB-tree can filter nodes using structural conditions in queries, while the COB-tree can filter nodes using text conditions. We have implemented a COB-tree and an STB-tree using GiST and examined index size and query processing time. Our experimental results show the efficiency of XICS in query processing.

Digital watermarks on pictures are more useful when they are better able to survive image processing operations and when they cause less degradation of picture quality. Random geometric distortion is one of the most difficult kinds of image processing for watermarks to survive because of the difficulty of synchronizing the expected watermark patterns to the watermarks embedded in pictures. This paper proposes three methods to improve a previous method that is not affected by this difficulty but that is insufficient in maintaining picture quality and treating other problems in surviving image processing. The first method determines the watermark strength in L*u*v* space, where human-perceived degradation of picture quality can be measured in terms of Euclidian distance, but embeds and detects watermarks in YUV space, where the detection is more reliable. The second method, based on the knowledge of image quantization, uses the messiness of color planes to hide watermarks. The third method reduces detection noises by preprocessing the watermarked image with orientation-sensitive image filtering, which is especially effective in picture portions where pixel values change drastically. Subjective evaluations have shown that these methods improved the picture quality of the previous method by 0.5 point of the mean evaluation score at the representative example case. On the other hand, the watermark strength of the previous method could be increased by 30% through 60% while keeping the same picture quality. Robustness to image processing has been evaluated for random geometric distortion, JPEG compression, Gaussian noise addition, and median filtering and it was clarified that these methods reduced the detection error ratio to 1/10 through 1/4. These methods can be applied not only to the previous method but also to other types of pixel-domain watermarking such as the Patchwork watermarking method and, with modification, to frequency-domain watermarking.

In ubiquitous environment that users access resource anytime and anywhere, access control model should consider user's location information. The proposed uT-RBAC includes the location information for user's least privilege. It also supports time related information, which enables the access control model to accommodate various ubiquitous environments. The proposed uT-RBAC can be dynamically applied to various ubiquitous computing envrionment.