This paper describes the author's perspective on multimedia quality prediction methodologies for multimedia communications in advanced mobile and internet protocol (IP)-based telephony, and reports related experiments and trials. First, the paper describes the need for perceptual QoS (Quality of Service) assessment in which various quality factors in multimedia communications for advanced mobile and IP-based telephony are analyzed. Then an objective quality prediction scheme is proposed from the viewpoints of quality measurement tools for each quality factor and an opinion model for compound quality factors in mobile and IP-based communications networks. Finally, the author's current trials of measurement tools and opinion models are described.

Next generation codecs, benchmarked by the H.264/AVC standard, are providing substantial compression efficiency for the coding and transmission of video. Coupled with technologies offering larger transmission bandwidths over DSL, wireless and satellite networks, the capability of delivering high quality video services to the home is now a reality. The perceptual quality of the content delivered over communications networks will be crucial in ensuring a first-class customer experience. It is therefore important to assess the advantages and disadvantages of the optional features offered by next generation codecs. This paper describes a subjective assessment that was carried out to investigate the perceptual effects of switching the in loop de-blocking filter within the H.264/ AVC CODEC on or off. Although the filter is believed to substantially improve the perceptual quality of video, it has been suggested that in some cases negative perceptual effects can be produced. The H.264/AVC architecture allows de-blocking to be switched off in cases where there are limited processing resources or it is considered a negative perceptual effect may be introduced. This paper describes a study that examined the perceptual effects of de-blocking by employing a standardised subjective assessment methodology. The Absolute Category Rating (ACR) method was used to capture Difference Mean Opinion Scores (DMOS) for a range of video. Content was selected to span a wide and representative range of coding complexity. This content was then encoded at a variety of bit-rates to represent high, medium and low qualities. Results were used to examine the end-user perception of video quality when the de-blocking filter is switched on or off. The experimental design allowed the overall effects of the de-blocking filter to be examined and additionally the relationship between content and quality on the filter performance. The experiment found that the performance of the de-blocking filter was content-dependent. Results were used to discuss the advantages and disadvantages of in-loop de-blocking and there is an examination of content properties (e.g. spatial and temporal complexity) that influence the performance of de-blocking.

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 describes an objective measurement method designed to assess the perceived quality for digital videos. The proposed approach can be used either in the context of a reduced reference quality assessment or in the more challenging situation where no reference is available. In that way, it can be deployed in a QoS monitoring strategy in order to control the end-user perceived quality. The originality of the approach relies on the very limited computation resources which are involved, such a system could be integrated quite easily in a real time application. It uses a convolutional neural network (CNN) that allows a continuous time scoring of the video. Experiments conducted on different MPEG-2 videos, with bit rates ranging from 2 to 6 Mbits/s, show the effectiveness of the proposed approach. More specifically, a linear correlation criterion, between objective and subjective scoring, ranging from 0.90 up to 0.95 has been obtained on a set of typical TV videos in the case of a reduced reference assessment. Without any reference to the original video, the correlation criteria remains quite satisfying since it still lies between 0.85 and 0.90, which is quite high with respect to the difficulty of the task, and equivalent and more in some cases than the traditional PSNR, which is a full reference measurement.

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.

This paper investigates the effects of group (or inter-destination) synchronization control over haptic media in a networked game where two players move objects competitively by manipulating haptic interface devices. The group synchronization control adjusts the output timing of haptic media among multiple players. By experiment, we demonstrate the effectiveness of the control. We also discuss the fairness between the two players quantitatively.

The "Proactive Desk" is a new human-machine interface for the desktop operations of computers. It provides users with tactile sensation in addition to visual sensation. Two linear induction motors underneath the desk generate a two-dimensional force to move objects and control their positions on the desktop using feedback control, and users feel tactile sensation while handling those objects. In this paper, we examined the effects of adding haptic information to simple mouse operation using the Proactive Desk. In our experiment, we used a button-type visual stimulus with and without haptic information. When using haptic conditions, three types of force feedback pattern were displayed: "Edge," "Resistance to motion" and "Attractive force," and each had three force strength conditions: no, half and full. The subject was asked to push buttons twenty times as the buttons were shown one after the other on the desk as quickly as possible. Consequently, the reaction times for pushing the button for all haptic conditions, except for the half-force condition of "Attractive force," were significantly faster than no-force (without haptic information) condition. This result shows that the haptic information was advantageous for easy operation.

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.

Packet loss and delay cause degradation in the quality of real-time, interactive applications such as video conferencing. Forward error correction (FEC) schemes have been proposed to make the applications more resilient to packet loss, because the time required to recover the lost packets is shorter than that required to retransmit the lost packets. On the other hand, the codec generally used in real-time applications like MPEG4 has the feature that the sending bit rate and the packet size of the traffic vary significantly according to the motion of an object in a video. If the traditional FEC coding, which is calculated on the basis of a fixed-size block, is applied to such applications, a waste of bandwidth and a delay variation are caused and the quality is degraded. In this paper, we propose suitable FEC schemes for visual communication systems using variable bit-rate (VBR) codec and evaluate the effectiveness of these schemes using our prototype implementation and experimental network.

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.

In this paper, we propose a novel dynamic bandwidth allocation scheme for the downlink real-time video streaming in the wireless cellular networks. Our scheme is able to maximize the bandwidth utilization, while satisfying the required packet loss probability, a QoS constraint, by dynamically determining the amount of bandwidth to be allocated at each unit time interval by measuring the queue length and the packet loss probability. The simulation results show that, without the need of a priori knowledge about the traffic traces, our scheme is able to achieve the same level of performance as what can be accomplished with the pre-calculated effective bandwidth in terms of the bandwidth utilization and the packet loss rate.

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.

This paper proposes ABdis, a new active measurement method for estimating the available bandwidth on a communication network path. Due to the recent explosion in multimedia applications, it is becoming increasingly important to manage network QoS, and tools that can measure network quality precisely are necessary to do this. Many conventional active measurement methods/tools, however, can measure/estimate only the average of the available bandwidth in a given period but cannot measure its distribution. If the distribution of the bandwidth over short intervals can be measured, the information would be useful for network management, proxy selection, and end-to-end admission control. We propose an end-to-end active measurement method called ABdis, which can estimate the distribution of the available bandwidth in a network path. ABdis employs multiple probes having different rates and a parameter-matching technique for estimating distribution. Furthermore, we present the results of simulations and verify ABdis's performance under various conditions by changing probe parameters, amount of cross traffic, and network models.

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.

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.

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.

We propose a real-time measurement method, DPDC (Detection of Packet-Delay Correlation), which models both available bandwidth and the averaging time scale. In this method, measurement periods are short and constant, while the theoretical measurement error is reduced. DPDC is established based on the discussion of the systematic error of the packet pair/train measurement. We evaluate through simulations its accuracy and robustness against the multihop effect. We also verify the feasibility of real-time measurements through testbed experiments using a tool called Linear that implements DPDC. Efficiency is demonstrated through simulations and testbed experiments by analyzing accidental and systematic errors. Finally, we discuss the available bandwidth variation in an Internet path using real-time data produced by Linear measurements and passive monitoring.

This paper describes a high-speed MT connector assembly method. This technique uses adhesive with a short hardening time, is highly reliable and does not require a polishing process, thus reducing the connector assembly time. First, we investigated an alpha-cyanoacrylate adhesive that hardens quickly and whose adhesive strength does not decrease under high humidity and high temperature conditions, thus ensuring its excellent reliability for outside use. In addition, we investigated variations in the position of the fiber endface on the ferrule endface with a view to obtaining a low insertion loss. Based on the results, we assembled an MT connector using our proposed high-speed assembly method. We confirmed that the assembly time could be reduced to less than 70% of the time required with the conventional method. MT connectors assembled using this technique have a low insertion loss and stable environmental characteristics.

As an alternative solution to provide the quality of services (QoS) for broadband access over Ethernet Passive Optical Network (EPON), we present the usage of MAC control message for plural class queues and a traffic-class burst-polling based delta dynamic bandwidth allocation (DBA), referred to as TCBP-DDBA, scheme. For better QoS support, the TCBP-DDBA minimizes packet delays and delay variations for expedited forwarding packet and maximizes throughput for assured forwarding and best effort packets. The network resources are efficiently utilized and adaptively allocated to the three traffic classes for the given unbalanced traffic conditions by guaranteeing the requested QoS. Simulation results using OPNET show that the TCBP-DDBA scheme performs well in comparison to the conventional unit-based allocation scheme over the measurement parameters such as: packet delay, packet delay variation, and channel utilization.

Because of the development of recent broadband access technologies, fair service among users is becoming more important goal. The most promising router mechanisms for providing fair service is per-flow traffic management. However, it is difficult to implement in high-speed core routers because per-flow state management is prohibitively expensive; thus, a large number of flows are aggregated into a small number of queues. This is not an acceptable situation because fairness degrades as the number of flows so aggregated increases. In this paper, we propose a new traffic management scheme called Hierarchically Aggregated Fair Queueing (HAFQ) to provide per-flow fair service. Our scheme can adjust flow aggregation levels according to the queue handling capability of various routers. This means the proposed scheme scales well in high-speed networks. HAFQ improves the fairness among aggregated flows by estimating the number of flows aggregated in a queue and allocating bandwidth to the queue proportionally. In addition, since HAFQ can identify flows having higher arrival rates simultaneously while estimating the number of flows, it enhances the fairness by preferentially dropping their packets. We show that our scheme can provide per-flow fair service through extensive simulation and experiments using a network processor. Since the currently available network processors (Intel IXP1200 in our case) are not high capacity, we also give extensive discussions on the applicability of our scheme to the high-speed core routers.

In wavelength-routed optical networks, wavelength converters are considered as one of the most critical network resources because they can significantly reduce the blocking probability, but still remain quite expensive. Unfortunately, previous wavelength assignment algorithms have seldom considered their presence. Therefore, in this paper, we propose a novel dynamic algorithm that can minimize the number of wavelength translations. Our algorithm establishes lightpaths by connecting a minimum number of wavelength-continuous segments. We mathematically prove the correctness of our algorithm. Then, we carry out extensive performance evaluations over three typical topologies with full-range or limited-range converters to compare our proposed algorithm with first-fit and most-used algorithms. The simulations show that, to obtain similar blocking performance, our algorithm requires much fewer converters, or the same number of converters but with smaller conversion ranges. From another perspective, with the same conversion capacity, our algorithm can significantly improve the blocking performance. Our algorithm is also scalable due to its polynomial time complexity and insignificant local signaling overhead.

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.

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.

The capability of Hierarchical Mobile IP (HMIP) for intra-domain route optimization is impaired when it is combined with Network Mobility (NEMO) technology. Deviations from the optimum path, caused by traffic aggregation in the Mobility Anchor Point (MAP), can be observed within a hierarchical domain. The problem is particularly noticeable in domains that span the mesh network topology. The lack of intra-domain path optimization in multi-level Mobile IP (MIP) leads to inefficient use of network resources. A Proxy Mobility Anchor Point (PMAP) functionality is proposed in domain nodes to enable intra-domain path optimization in multi-level MIP. Numerical evaluation and simulations indicate that this proposal can improve routing efficiency and throughput. The solution can be especially rewarding in network architectures where access network is separated from global network by bottleneck links and where the majority of users accessing the network are mobile routers.

In this paper, we present Blueweb, a new Bluetooth-based multihop network with an efficient scatternet formation algorithm and a hybrid routing protocol. The Blueweb is designed from the original idea of Bluetree. Blueweb's scatternet formation uses two mechanisms. One is the role exchange mechanism in which only slave nodes serve as the role of relay through the whole scatternet. The other one is the return connection mechanism in which we convert the scatternet from a tree-shaped to a web-shaped topology. Meanwhile, a modified source routing protocol is designed for Blueweb in which we combine the proactive method with the reactive method to discover the optimal path for packet transmission. Furthermore, using computer simulations we compared the system performance of Blueweb and Bluetree with both a static model and a uniform traffic model. With the static model we evaluate the scatternet performance and with the uniform traffic model we evaluate the transmission performance. Our simulation results show that Blueweb achieves superior system performance than Bluetree on both scatternet performance and transmission performance.

We present a quantitative evaluation of speech quality using the multiplexing scheme for the efficient transmission of voice signals in order to reduce the number of the IP packets carrying voice signals (called VoIP packets) transferred. The multiplexing scheme is applicable to a variety of media gateways controlling the bulk of voice streams over IP-based networks, based on VoIP technology. We speculated that the multiplexing scheme would reduce the degradation of speech quality due to packet loss since it also has a similar effect to interleaving the voice signal streams. However, the interleaving effect for maintaining speech quality in the scheme characterized by the feature of IP-based multiplication is not quantitatively clear. Through our end-to-end quality evaluation results of speech, as transmitted via the multiplexing scheme using dedicated hardware, we confirm the advantages of the multiplexing scheme from the perspective of achieving improved speech quality without increasing the processing delay when considering practical packet loss conditions within an IP-based network.

In AMS/OFDM systems, a base station controls the modulation level of each subcarrier with feedback information (FBI), and then, adaptive modulated packets are transmitted from the base station to the mobile station. In this case, the mobile station requires modulation level information (MLI) to demodulate the received packet. The MLI is generally transmitted as a data symbol, so the throughput is degraded. To overcome this problem and increase the total throughput, in this paper, we propose superimposed frequency symbol based adaptive OFDM with frequency spreading and equalization. In the proposed system, each S/P transformed signal is spread by orthogonal spreading codes and combined. This means that each subcarrier holds several superimposed S/P transformed signals with the same power rate. In this case, the frequency-selective faded subcarriers obtain the same power rate for each S/P transformed signal. Therefore, the detected signals also obtain the same SINR, and as a result, we can assign the same modulation level for each frequency symbol spreading block. Hence, the proposed system requires only one piece of FBI and MLI for each frequency symbol spreading block, as compared with conventional adaptive OFDM.

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.

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.

An FRPA (frequency reuse power allocation) technique by employing the frequency reuse notion as a strategy for overcoming the ICI (intercell interference) and maintaining the QoS (quality of service) at the cell boundary is described for broadband cellular networks. In the scheme, the total bandwidth is divided into sub-bands and two different power levels are then allocated to sub-bands based on the frequency reuse for forward-link cell planning. In order to prove the effectiveness of the proposed algorithm, a Monte Carlo simulation was performed based on the Chernoff upper bound. The simulation shows that this technique can achieve a high channel throughput while maintaining the required QoS at the cell boundary.

The performance of a feedback-type adaptive array antenna (AAA) system placed only at a base station (BS) in an FDD/DS-CDMA system remains insufficiently clear. We evaluate the performance of this system by considering the effect of a rake receiver, spacing distance between antennas, the maximum Doppler frequency (f_d), and control delay time (T_d) on BER performance. In this system, the mobile station (MS) determines optimum weights of antenna elements and sends them back to BS as feedback information. We assume that the optimum weights are not quantized. Thereby, we estimate the performance degradation of 3GPP transmit diversity system, where the feedback information is quantized using a few bits. Computer simulation results show that the rake receiver achieves better BER performance because of the time diversity effect with rake receiver. The AAA with a wide antenna spacing gives high diversity gain for the received signals. For a high value of f_d T_d, BER performance becomes worse because weighting factors cannot follow the changing speed of channel characteristics. The degradation in performance of a 3GPP system is clarified.

A new seamless symbol rate switchable modem for multi-rate FDMA systems is proposed in this paper. In the new modem, a novel clock phase compensation algorithm makes it possible to switch the symbol rate synchronously between the transmitter and the receiver, with no degradation in BER when the symbol rate is changed. In addition, by matching the interpolation filter to the symbol rate, this modem is capable of operating at lower clock speeds, which greatly reduces the consumption power. Computer simulations confirm its fundamental performance. Simulation results show that the proposed power-efficient symbol rate switchable modem can change the symbol rate without degrading BER performance.

Recently, superimposed pilot channel estimation has been proposed for wireless communications, where the pilot symbol sequence is superimposed on a data symbol sequence and transmitted together, and thus there is no drop in information rate. In this scheme, the receiver correlates the received signal sequence with the pilot symbol sequence, and obtains the channel estimate. However, the correlation between the pilot symbol sequence and the data symbol sequence deteriorates the channel estimation accuracy. The use of the longer frame leads to the lower correlation, but also to the lower channel tracking capability. In this paper, we propose a selective superimposed pilot channel estimation scheme with selecting a pilot sequence that has a low correlation with a data symbol sequence from the set of the pilot sequences assigned to the transmitter. Note that the superimposed channel estimation scheme with one pilot sequence assigned to the transmitter is the conventional superimposed channel estimation scheme. We show that the proposed channel estimation scheme is superior to the conventional superimposed channel estimation scheme (pilot sequence = 1). We also show that the proposed channel estimation scheme can achieve the good channel estimate even under fast fading environments. Moreover, we show that the proposed channel estimation scheme is superior to the pilot assisted channel estimation scheme, although pilot symbol power is a deterioration factor in the proposed channel estimation scheme.

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 simple self-biased receiver system with a dual branch architecture consisting of a low-power consumption receiver and a rectenna is introduced. The system is efficiently integrated with a dual-fed circular sector antenna with harmonic rejection characteristics without a BPF. The receiver portion is designed by utilizing a low-noise amplifier (LNA) with low power consumption and a self-heterodyne mixer, while the rectenna achieves high conversion efficiency up to 80%, thanks to the harmonic rejection of the circular sector antenna. The rectified DC power from the rectenna is applied for a bias of the receiver without any external bias. Simultaneously, an ASK digital signal demodulation without an extra power supply are implemented successfully.

This study is devoted to a half-wave dipole with a conductor plane at a distance much smaller than a quarter wavelength which we designate as an ultra low profile dipole (ULPD) antenna in this paper. The concerns of ULPD antenna are the feeding method and the impedance matching, because the input impedance usually tends to be lowered by the existence of a metallic structure in its proximity. In this paper, we propose a ULPD antenna with an excellent impedance matching and a coaxial feed built within the antenna structure so that the external matching and a balun are not required. A coaxial cable is used as a feed line and extended to be a half of a half wavelength dipole. The other half is made up of a parasitic element, which is connected to the outer conductor of the coaxial radiator. To make a matching, the outer conductor of the coaxial radiator is stripped off at a suitable length, and the total length of a dipole is considered for its resonance at a desired frequency of 2 GHz. The experiment has been conducted. The results show the return loss of -27 dB and the maximum gain of 9 dBi in the normal direction to the conductor plane. The computational results are also obtained, which agree well with the experimental results.

Providing efficient Video-On-Demand (VOD) service on Broadband Cable Networks (BCNs) is still a challenging issue, although BCNs have been the most important form of broadband services in the US. To address this issue, we have proposed the optimal full-sharing scheduling approach that minimizes the bandwidth consumption of video sessions [1]. We have shown that full-sharing scheduling has remarkable advantages in minimizing the bandwidth consumption of VOD service on BCNs. Furthermore, we design two adaptation algorithms in this paper, which not only keep bandwidth consumption minimal but also significantly reduce the mean service delay.

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 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.

We propose an ASK modulator and Antenna driver for multi-standard 13.56 MHz RFID readers and NFC devices. The proposed transmitters of RFID readers and NFC devices consist of ASK modulator, inverting output driving buffer and off-chip antenna with a matching circuit. The ASK modulation depth can easily be controlled by adjusting duty ratio by the delay line circuit. The test chip was fabricated by using a 0.35 µm double poly CMOS process. The measured results demonstrated that the proposed circuit has met the standard specifications.

We present a numerical analysis of the optimal number of groups for minimizing the average paging delay. In the analysis, we consider uniform distributions for location probability conditions and apply the M/D/1 queueing model to paging message queues of cells. We also compute the lower bounds of group numbers and investigate the minimum transmission capacity under average paging delay constraints.

In this letter we examine two transport protocols, HighSpeed TCP [1] and Scalable TCP [2] which are both sender-side varieties of TCP. Based on the fluid flow theory, we develop a general nonlinear model and use gain margin and phase margin to evaluate the stability of a closed-loop system which is composed of a transport protocol and an active queue management scheme. Our results indicate that HSTCP and STCP are stabler than standard TCP when link bandwidth, flow number and round-trip time vary.

Clustering is an effective self-organization approach in wireless sensor networks. LEACH--a representative distributed clustering scheme has been considered an effective model to offer energy-efficient communication for sensor networks. However, its randomness may result in faster death of some nodes i.e. shorten system life. In this letter, we first analyze the reasons why the uncertainty in LEACH degrades system life, and then present a distributed clustering algorithm based on an adaptive backoff strategy. Simulation experiments illustrate that our algorithm is able to significantly prolong system life compared with LEACH.

This paper introduces a State Dependent Random Early Detection (SDRED) scheme that can improve delay and jitter performances by adjusting RED parameters such as max_th and w_q 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.

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).

The guard channel scheme in wireless mobile networks has attracted and is still drawing research interest owing to easy implementation and flexible control. Dynamic guard channel schemes have already been proposed in the literature to adapt to varying traffic load. This paper presents a novel control-theoretic approach to dynamically reserve guard channels called PI-Guard Channel (PI-GC) controller that maintains the handoff blocking probability (HBP) to a predefined value; while it still improves the channel resource utilization.

Resource management for infrastructure-based two-hop fixed relay systems which are applicable to TDD-FH-OFDMA based cellular systems with sectorization is proposed in this paper. The severe interference problem caused by both inter-sector and inter-cell can be tackled by employing 6-sector directional antennas combined with the resource allocation. The simulation results demonstrate that at the outer-region of the cell, the high data rate service coverage can be extended.

A novel error detection method based on coded block pattern (CBP) information verification is proposed for error concealment of inter-coded video frames transmitted in wireless channel. This method firstly modifies the original video stream structure by the aggregation of certain important information, and then inserts some error verification bits into the video stream for each encoded macro block (MB), these bits can be used as reference information to determine whether each encoded MB is corrupted. Experimental results on additive Gauss white noise simulation wireless channel and H.263+ baseline codec show that the proposed method can outperform other reference approaches on error detection performance. In addition, it can preserve the original video quality with a small coding overhead increase.

Based on the observation that foveation analysis can be used to find most critical content in terms of human visual perception in video and image, one effective error resilience method is proposed for robust transmission of H.264 intra-coded frame in wireless channel. It firstly exploits the results of foveation analysis to find foveated area in picture, and then considers the results of pre-error concealment effect analysis to search for the center of foveation macro-blocks (MB) in foveated area, finally new error resilient alignment order of MB and new coding order of MB are proposed that are used in video encoder. Extensive experimental results on different portrait video sequences over random bit error wireless channel demonstrate that this proposed method can achieve better subjective and objective effect than original JM 8.2 H.264 video codec with little effect on coding rate and image quality.