High-speed streaming services are considered to be among the most promising services for the broadband Internet of the next generation. A contents delivery network (CDN) that allows effective and scalable content delivery will be required in this role. However, the actual requirements for the CDN, such as quality of service (QoS), the service deployment strategy and technical issues are not yet to be defined. In this paper, the bandwidth requirement of the CDN is derived through QoS evaluation. On the basis of this requirement, we propose two feasible CDN configurations: an IP technology based configuration and an optical video distribution technology based configuration. Finally, we make lists of technical issues for the configurations and discuss some of the possible solutions.

Gnutella's service announcement in March 2000 stirred worldwide interest by referring to P2P model. Basically, the P2P model needs not the broker "the centralized management server" that until now has figured so importantly in prevailing business models, and offers a new approach that enables peers such as end terminals to discover out and locate other suitable peers on their own without going through an intermediary server. It seems clear that the wealth of content made available by peer-to-peer systems like Gnutella and Freenet have spurred many authors into considering how meta-data might be used to support more effective search in a distributed environment. This paper has reviewed a number of these systems and attempted to identify some common themes. At this time the major division between the different approaches is the use of a hash-based routing scheme.

Multicast communications have been expected as an effective way to disseminate information to potentially large number of receivers. IP multicast is a well- known multicast protocol for the Internet. However, IP multicast has several technical problems to be resolved until it is widely deployed in the Internet. These includes service-model of multicast group, reliable transport protocol, congestion control and security. A lot of researches trying to resolve these technical problems make multicast communications a hot research area in these couple of decades. This paper overviews the present style of IP multicast and clearlify technical issues of the present multicast communications. The paper also surveys important approaches to these problems and discuss about future directions of multicast communications.

This paper proposes an IPv6-based network service control architecture for providing a variety of customized services to both stationary and mobile users in a unified manner. Recent trends in the Internet indicate its evolution into a combination of broadband and mobile-aware networks. One means of providing users with cost-efficient customized services in such large-scale IP networks is to introduce flexible network intelligence capabilities for managing network resources and services. The purpose of the proposed network architecture is to upgrade the Internet so that it functions more intelligently by using service profiles (data sets containing the service specifications of individual users) and mechanisms for their distribution. It is possible to make network services intelligent by using network application programming interfaces (APIs), which have been under study in international standardization groups. We apply the open API concept to our proposed architecture to produce a wide variety of services. We also propose a new open API to support Web content adaptation services, which add value to Web access.

In this paper, we present a CDN (Content Delivery Network) architecture for mobile streaming service in which content segmentation, request routing, pre-fetch scheduling, and session handoff are controlled by SMIL (Synchronized Multimedia Integration Language) modification. In this architecture, mobile clients simply follow modified SMIL files downloaded from a portal server; these modifications enable multimedia content to be delivered to the mobile clients from the best surrogates in the CDN. The key components of this architecture are 1) content segmentation with SMIL modification, 2) on-demand rewriting of URLs in SMIL, 3) pre-fetch scheduling based on timing information derived from SMIL, and 4) SMIL updates by SOAP (Simple Object Access Protocol) messaging for session handoffs due to client mobility. This architecture enhances streaming media quality for mobile clients while utilizing network resources efficiently and supporting client mobility in an integrated and practical way. The current status of our prototype on a mobile QoS testbed "MOBIQ" is also reported in this paper.

Multicast transmission, which can send the same information simultaneously to multiple users, is a key technology in content delivery networks. In this paper, we discuss a new multicast architecture with network coding proposed by Li et al. , which breaks limitation of existing IP multicast in terms of network resource utilization. Network coding based multicast can achieve the max-flow, which is the theoretical upper bound of network resource utilization. However, the max-flow transmission is not always effective and may not be robust against congestion because it maximally uses link capacity of multicast distribution tree. In this paper, we first introduce a load balancing method of network coding as an alternative use to the max-flow transmission. Next, we study the feasibility of network coding based multicast architecture from performance aspect and evaluate the network coding in terms of the max-flow and load balancing with a computer simulation. There has been no evaluation of network coding in practical network environment with packet losses and propagation delay. We also describe required key techniques and technical problems to implement network coding on the current IP networks. Our results will offer valuable insight for designing the future Internet with higher and more effective network utilization.

Load balancing among multiple mirror servers located at distributed positions in the network is a key technique for content delivery services. For bandwidth allocated services, we consider how to select a suitable server from several candidates containing the same content at the time of a request. We propose limiting the candidates in advance and selecting a server from the limited set of servers in a round-robin fashion. The server sets that minimize the variance of the link load are derived using a greedy method for a given network topology and service demand. Through numerical evaluation, we demonstrate that the proposed method is superior to previous methods.

In content distribution networks (CDNs), the content routing which directs user requests to an adequate server from the viewpoint of improvement of latency for obtaining contents is one of the most important technical issues. Several information, e.g. server load or network delay, can be used for content routing. Network support, e.g. active network, enables a router to select an adequate server by using these information. In the paper, we investigate a server selection policy of a network support approach from the viewpoint of which information to be used for effective server selection. We propose a server selection policy using RTT information measured at a router. Simulation results show that our proposed server selection policy in content routing selects a good server under both conditions where server latency and network delay is a dominant element of user response time. Furthermore, we also investigate about location of routers with network support bringing good performance for our proposed scheme.

A content-based switch makes forwarding decisions (server selections) based on an application layer information and forwards data in the application layer. After making forwarding decisions, existing content-based switches improve their forwarding performance by TCP splicing, which releases them from maintaining TCP endpoints and allows them to forward data by packet forwarding. However, once content-based switches invoke TCP splicing, they are unable to use the application layer information for forwarding decisions. Thus the existing content-based switches cannot perform a handoff of pipelined HTTP transactions, which can greatly reduce client perceived latencies. This paper proposes an asymmetric TCP splicing and a method to perform the handoff of HTTP transactions between servers. Asymmetric TCP splicing allows the content-based switches to use all the application layer information in the TCP data stream from clients to servers, although it allows the switches to forward the TCP data stream from servers to clients by packet forwarding. The proposed handoff method content-based switches support pipelined HTTP transactions in combination with asymmetric TCP splicing. In the proposed method, the content-based switch utilizes the common function of TCP (TCP half-close) to detect the end of the series of responses from the currently selected server, and it changes the forwarding destination after the client finishes receiving a series of responses from the server. Our evaluation validates that the content-based switch which supports pipelined HTTP transactions by our method can reduce client-perceived latencies when there is a large correlation between destinations of any two consecutive requests.

Reliable multicast is an interesting application of distributing data to lots of clients at the same time. In heterogeneous environment, it is necessary to adjust the transmitting rate corresponding to the bandwidth of receivers. Placed at a network bottleneck point, an active server can buffer the multicast packets and control the transmitting rate to the downstream multicast receivers independently so as to absorb bandwidth differences. If wireless and wired receivers coexist, the best position for the active server is at the edge of the wired and wireless links because the bandwidth of wireless receivers are lower than that of wired receivers. However, it is not enough that an active server only controls the transmitting rate in such environment because wireless receivers tend to lose packet by the wireless transmission error. This paper proposes a scheme in which the active server independently controls a reliable multicast scheme that is robust against packet loss due to wireless transmission error. Simulation results show that rate-based reliable multicast congestion control is more appropriate than window-based control for wireless links. We also show that FEC applied only to the wireless link improves the throughput of wireless multicast receivers. Finally, we show that combining rate-based reliable multicast congestion control scheme with FEC only for the wireless link makes reliable multicast more practical and friendly with TCP even if packets are lost due to transmission errors.

This paper proposes a new dynamic multicast routing algorithm for layered streams. Since a layered multicast technique accommodates different types of users in the same multicast group, it helps to provide multicast services in a heterogeneous environment. However, this makes it difficult to construct an efficient routing tree when receivers join or leave a multicast session dynamically. In the proposed algorithm, we adopt a pre-determined path approach to handle such dynamic membership of a layered multicast session without the burden of much additional traffic. Simulation results show that the proposed algorithm can minimize the average multicast tree cost, and that it works well on large-scale networks and those with traffic heterogeneity and a small number of routing control messages.

In this paper, we first investigate the characteristics of movie contents over the Internet. As in the previous studies, we found the lognormal-distribution well fits the distribution of file size for the whole set of general movie contents. When we specifically focus on the subset that consists of movie trailers, however, it shows different distribution from the lognormal-distribution. Our analysis shows it is similar to an exponential-distribution. We here assume that movie trailers are one of the relevant contents for Contents Delivery Networks (CDN) or Peer-to-Peer (P2P) file exchange communities. We further studies the relationship between playing duration and file size for the movie trailers and we did not find any linear correlation among them. We next consider bandwidth requirements to retrieve movie trailer contents. Our objective is to make it possible for user to view the contents in real-time. Many previous studies investigate bandwidth requirement based only on the file size distribution. In this paper, we analyze the traffic design criteria for CDN or P2P by taking into account both of the results for the file size distribution and the relationship between playing duration and file size for movie trailers. Simulation studies reveal the impact for the bandwidth requirement.

The proxy mechanism widely used in WWW systems offers low-delay data delivery by means of "proxy server." By applying proxy mechanisms to video streaming system, we expect that high-quality and low-delay video distribution can be accomplished without introducing extra load on the system. In addition, it is effective to adapt the quality of cached video data appropriately in the proxy if user requests are diverse due to heterogeneity in terms of the available bandwidth, end-system performance, and user's preferences on the perceived video quality. In this paper, we propose proxy caching mechanisms to accomplish high-quality and low-delay video streaming services. In our proposed system, a video stream is divided into blocks for efficient use of cache buffer. A proxy cache server is assumed to be able to adjust the quality of cached or retrieved video blocks to requests through video filters. We evaluate our proposed mechanisms in terms of the required buffer size, the play-out delay and the video quality through simulation experiments. Furthermore, to verify the practicality of our mechanisms, we implement our proposed mechanisms on a real system and conducted experiments. Through evaluations from several performance aspects, it is shown that our proposed mechanisms can provide users with a low-latency and high-quality video streaming service in a heterogeneous environment.

With the advance of high-speed network technologies, availability and popularity of streaming media contents over the Internet has grown rapidly in recent years. Because of their distinct statistical properties and user viewing patterns, traditional delivery and caching schemes for normal web objects such as HTML files or images can not be efficiently applied to streaming media such as audio and video. In this paper, we therefore propose an integrated caching scheme for streaming media with segment-based caching and hierarchically distributed proxies. Firstly, each stream is divided into segments and their caching algorithms are considered to determine how to distribute the segments into different level proxies efficiently. Secondly, by introducing two kinds of segment priorities, segment replacing algorithms are proposed to determine which stream and which segments should be replaced when the cache is full. Finally, a Web-friendly caching scheme is proposed to integrate the streaming caching with the conventional caching of normal web objects. Performance of the proposed algorithms is verified by carrying out simulations.

Takine investigated the continuous time queueing system with single server and Markovian arrival streams (MAS) where service times of customers depend on the states of the underlying Markov chain immediately before and after arrivals and may differ from one another among streams. In this paper, we consider the discrete version of Takine's model. Since a MAS can be used to model the correlated traffics of several classes, we consider the discrete time queueing system with MAS as an input traffic. Moreover packets from different classes of traffic may require different service times. So service times of customers which are different from one another among streams and may depend on the states of the underlying Markov chain immediately before and after arrivals, are also considered. We obtain the distribution of the workload in the system using Neuts' matrix analytic method. Based on this result, we obtain the distributions of the actual waiting time and the sojourn time of a customer of each class and the joint distribution of the numbers of customers of each class. We also provide a recursion formula for computing the joint queue length distribution. We give an application to multimedia conference system and numerical example.

We describe a low-cost and extremely stable millimeter-wave transmission system that uses a double-side-band (DSB) millimeter-wave self-heterodyne transmission technique. This technique allows us to use a comparatively low-cost and unstable millimeter-wave oscillator regardless of the modulation format. Furthermore, a transmission band-pass-filter (BPF) is not needed in the millimeter-wave band. The system cost can therefore be substantially reduced. We have theoretically and experimentally evaluated the carrier-to-noise power ratio (CNR) performance that can be obtained when using this technique relative to that attainable through a conventional millimeter-wave self-heterodyne technique where a single-side-band signal is transmitted. Our results show that the DSB self-heterodyne transmission technique can improve CNR by more than 3 dB.

Two implementation schemes for a two-step SOVA (Soft Output Viterbi Algorithm) decoder are proposed and verified in a chip. One uses the combination of trace back (TB) logic to find the survivor state and double trace back logic to find the weighting factor of a two-step SOVA. The other is that the reliability values are divided by a scaling factor in order to compensate for the distortion brought by overestimating those values in SOVA. We introduced a fixed scaling factor of 0.25 or 0.33 for a rate 1/3 and designed an 8-state Turbo decoder with a 256-bit frame size to lower the reliability values. The implemented architecture of the two-step SOVA decoder allows important savings in area and high-speed processing compared with the one-step SOVA decoder using register exchange (RE) or trace-back (TB) method. The chip is fabricated using 0.65 µm gate array at Samsung Electronics and it shows higher SNR performance by 2 dB at the BER 10^-4 than that of a two-step SOVA decoder without a scaling factor.

We propose an extended one-shot decorrelating detector (EOS-DD) which may be viewed as a generalized double window multiuser detector (DW-MD) for asynchronous direct-sequence code-division multiple-access (DS/CDMA) systems in frequency selective fading environments. The EOS-DD extends a processing window and the received signal over an extended window is utilized for decorrelating. The effects of the window size on BER performance are investigated by numerical analysis. Analysis and simulation show that the EOS-DD is superior to the one-shot decorrelating detector (OS-DD) and finite memory length truncated decorrelating detector (FIR-DD) in terms of noise enhancement and near-far resistance. It is also shown that the EOS-DD with window size 4 can provide significantly improved performance compared to the EOS-DD with window size 2.

This paper proposes a new signal peak power reduction technique, Peak Reduction based on Control Signal Insertion (PRCSI), for broadband mobile communications based on multi-channel signaling schemes. PRCSI reduces the peak power with a peak control signal that is generated symbol-by-symbol; no signal band expansion is incurred because the peak control signal is inserted into the transmission signal band. PRCSI can achieve 4 dB peak power reduction for 8-carrier signaling, while the E_b/N₀ value required to achieve 10^-3 average BER is 1 dB larger with PRCSI than without it. This BER performance degradation can effectively be compensated by the proper use of Trellis coding. The proposed technique is applied to OFDM transmission systems with large carrier number. The proposed technique can achieve 3-dB peak power ratio for 128-carrier OFDM signals with less than 1-dB performance degradation at the BER of 10^-3.

Adaptive Battery Conservation Management (ABCM), an effective form of power conservation for mobile terminals in an always-connected environment, was proposed and evaluated in a previous published work. The ABCM method employs three states: active, dormant, and the Battery Saving Mode (BSM). The BSM is defined as a battery-saving state; in the BSM, the mobile terminal saves power by intermittently receiving paging notifications via a paging channel between the packet bursts of a session. Two control parameters, the sleep-timer and paging interval, are set up according to packet class and are the keys to the performance of a system with this method. In real-time communications, a long sleep-timer and short paging interval are selected to minimize buffering delay. In non-real-time communications, on the other hand, a short sleep-timer and long paging interval are chosen to reduce power consumption by the mobile terminal. Our previous evaluation showed that the method is effective as a means for power conservation in non-real-time communications. In real-time communications, on the other hand, the ABCM method provides shorter buffering delays and the same battery-conservation performance as the conventional method. To further improve the ABCM method's performance, we now propose an enhanced ABCM method that employs multiple BSM sub-modes, each of which has a different paging interval. As dormant periods become longer, the mobile terminal makes transition to successive sub-modes, each of which has a longer interval than the previous one. In this paper, we evaluate the battery conservation effect of the ABCM method through theoretical analysis and computer simulation. Numerical evaluation indicates that the ABCM method will be suitable for the broadband multimedia packet-radio systems of the future.

The IMT-2000 service launched in 2001 in Japan is expected to popularize multimedia services such as videophone, visual mail, video, and music distribution. With the rapidly increasing demand for high-speed mobile multimedia and the need to support diversified requirements of users, wireless Quality of Service (QoS) resource management has become an important and challenging issue. In order to improve the system capacity and rate of satisfied users, in this paper, a novel wireless QoS resource management scheme is proposed to carry out radio resource cooperative control among base stations. Computer simulations indicate that the proposed QoS resource cooperative control exhibits superior performance over conventional ones, and that a higher rate of satisfied users is achieved.

It is well known that Orthogonal Frequency Division Multiplexing (OFDM) scheme is robust to frequency selective fading in wireless channels. However, once delayed signals beyond a guard interval of an OFDM symbol are introduced in a channel with large delay spread, inter-symbol interference causes a severe degradation in the transmission performance. In this paper, we propose a novel pre-Fast Fourier Transform (FFT) OFDM adaptive antenna array, which requires only one FFT processor at a receiver, for suppressing such delayed signals. We analytically derive the optimum weights for the beamformer based on the Maximum Signal-to-Noise-and-Interference power Ratio (SNIR) and the Minimum Mean Square Error (MMSE) criteria, respectively. Computer simulation results show its good performance even in a channel where Directions of Arrival (DoAs) of arriving waves are randomly determined.

In this paper, to enhance the power efficiency a new simple space-time coding scheme is devised with application to the OFDM based Wireless LAN system. The basic idea is from the receiver's point of view and is referred to as Virtual Constellation Mapping (VCM). We designed a new combination of the channel coding (Turbo Code) along with multiple transmit antennas (Two antennas) to achieve transmit diversity and space division multiplexing transmission. Computer simulation results showed that with the same transmission data rate, our proposed scheme can achieve better bit error rate (BER) compared with the conventional space-time trellis coded OFDM scheme in high Doppler fading channels.

A field trial, within a suburban macro-cell environment, of a space-time (ST) equalizer for TDMA mobile communication systems is described. The ST equalizer was a cascade connection of two array processors for a four-antenna array and a two-branch-metric-combining maximum-likelihood sequence estimation (MLSE) that was designed to obtain full space- and path-diversity gains from first-arrival and one-symbol-delayed signals while suppressing excessively long-delayed inter-symbol interference (ISI). The radio frequency was 3.35 GHz, the transmission rate was 4.096 Mb/s, and the modulation was QPSK. The long-delayed ISI reduction and the space-path diversity effect of the ST equalizer was validated by E_b/N₀ vs. bit-error-rate (BER) curves with respect to delay spread and antenna spacing as compared with the case of an array processor alone being used.

In this paper, we propose a new multistage (iterative) structure where Kalman channel estimation and parallel interference cancellation multiuser detection are conducted in every stage (iteration). The proposed scheme avoids the complexity of the decorrelator in front of Kalman channel estimator, and has better performance than the previous scheme.

The demand for wireless mobile communications has grown at a very high rate, recently. In order to solve the non-uniform traffic rates, the use of cell splits is unavoidable for balancing the traffic rate and maximizing total system capacity. For cell planning, a DS-CDMA cellular system can be comprise of different cell sizes because of different demands and population density of the service area. In this paper, we develop a general model to study the forward link capacity and outage probability of a DS-CDMA cellular system with mixed cell sizes. The analysis of outage probability is carried out using the log-normal approximation. When a macrocell is split into the three microcells, as an example, we calculate the multi-cross interferences between macrocells and microcells, and the forward link capacities for the microcells and the neighboring macrocells. The maximum allowable capacity plane for macrocell and microcell is also investigated. The numerical results and discussions with previous published results of reverse link are summarized.

A new fuzzy logic-based buffer management scheme (BMS) called fuzzy early detection (FED) is presented to improve transmission control protocol (TCP) performance over the unspecified bit rate (UBR) service of asynchronous transfer mode (ATM) networks. FED uses early congestion detection and selective packet discard (ECD/SPD) strategies to achieve its design goals, and adopts early packet discard (EPD) to prevent frequent buffer overflows. Two variants of FED are constructed for comparison purposes. FED performance is then evaluated against those of its variants, pure EPD and P-random early detection (P-RED) through various simulations. The effects of TCP implementation, TCP maximum segment size (MSS), network's propagation delay (PD) and switch buffer size (SBS) on their performances are studied. The overall result shows that FED is superior to the other BMSs when the criteria of efficiency, fairness, scalability and switch buffer requirement are considered collectively.

This paper describes an analysis of the electromagnetic interference (EMI) aspects of electrostatic discharge (ESD), which sometimes causes serious damage to electrical systems. To classify EMI-related properties resulting from ESD events, we used a self-organizing neural network, which can map high-dimensional data into simple geometric relationships on a low-dimensional display. Also, to clarify the effect of a high-speed moving discharge, we generated one-shot discharges repeatedly and measured the ESD current in the time domain to obtain its EMI-related characteristics of this phenomenon. Based on the measured data, we examined several differential properties of ESD waveforms including the maximum amplitude and energy level, and analyzed these multi-dimensional data using the self-organizing neural network scheme. The results showed that the high-speed moving discharges can increase the maximum amplitude, relative energy, and entropy of ESD events, and that the positioning of the EMI level of each ESD event can be effectively visualized in a two-dimensional space.

Resource-demanding services such as Multi-media-on-Demand (MOD) become possible as Internet and broadband connections are getting more popular. However, as the sizes of multimedia files grow rapidly, storage of such large files becomes a problem. Since multimedia contents will generally become less popular with time, it is desirable to design a prediction algorithm so that the multimedia content can be unloaded from the server if it is no longer popular. This can relieve the storage problem in an MOD system, and hence spare more space for new multimedia files. In this paper, we analyse the MOD viewing trend in order to understand the viewing behaviour of users and predict the viewing trend of a particular category of multimedia based on the knowledge obtained from its trend analysis. In trend analysis, two additive regression models, exponential-exponential-sum (EES) and exponential-power-sum (EPS), are proposed to improve the fitness of the trend. The most suitable model will then be used for trend prediction based on four proposed approaches, namely Fixed Regression Selection (FRS), Continuous Regression Updating (CRU), Historical Updating (HU) and Continuous Regression with Historical Updating (CRHU). From the numerical results, it is found that CRHU, which is constructed by considering historical trend and new incoming data of viewing requests, is in general the best method in forecasting the request trend of a particular category of multimedia clips.

The authors study the performance of a disk storing WWW data by using a broadcast polling model. The model has single message queues, an exponential server and asymmetric exponential arrivals. Numerical examples show that the analytical results are close to the simulation results.

The autocorrelation peak reduction due to wavelength mismatches between an encoder and a decoder in wavelength hopping-time spreading (WH-TS) two-dimensional optical CDMA systems is analyzed and verified by experiments. The tolerance of the wavelength mismatch is then obtained through the analysis of the system bit error rate (BER) performance. The results show that no significant BER performance is degraded only by the wavelength mismatches less than 0.1 nm which corresponds to 25 percent of the typical fiber Bragg gratings' spectral width.

We investigate the design of an interpolation filter of a multi-frequency time division multiple access (MF-TDMA) demodulator which is applied to digital video broadcasting-return channel system via satellite (DVB-RCS). We propose two interpolation filters for symbol timing recovery in a digital receiver where the input analog to digital conversion sampling clock is not synchronized to the transmitter symbol clock. The two proposed interpolation filters are designed by the least mean-square-error at the output of the receiver. Simulation results show that a performance improvement is achieved by employing the proposed interpolation filter.

A minor component analysis approach based on the generalized sidelobe canceler is presented to realize the blind suppression of multiple-access interference in multicarrier code division multiple access systems. With a rough user-code and timing estimations, this proposed method of less computation performs the same as minimum mean square error detectors and outperforms existing blind detectors. Simulation results illustrate the effectiveness of the blind multiuser detection.

A downlink transmit diversity scheme is proposed for [1] WCDMA low chip rate (1.28 Mcps) Time Division Duplex (LCR-TDD) systems. In the proposed Time Switched Transmit Diversity (TSTD) scheme, two spatially separated antennas are alternately used at a base station in order to transmit consecutive sub-frames on the downlink physical channels. The proposed TSTD scheme takes advantage of the frame structure of the WCDMA LCR-TDD, where TSTD transmission is possible with a single power amplifier. Simply adding switching components in front of the transmit antennas yields a considerable gain compared with a non-diversity mode system. A new power control algorithm is also proposed for generating power control command for efficient use of TSTD. A performance gain of up to 5 dB, compared with a non-diversity scheme, is achieved in terms of the frame error rate for varying Eb/No and mobile speeds. From the viewpoint of both complexity and performance, the TSTD scheme is a simple and effective transmit diversity method that can increase downlink capacity in the WCDMA LCR-TDD systems.

A new power and bandwidth efficient modulation technique - Superposed Quadrature Quadrature Amplitude Modulation (SQ²AM) - for use in nonlinear satellite channel is presented. SQ²AM technique expands 2-dimensional SQAM signals into 4-dimensional quadrature modulated signals by using orthogonal baseband waveforms and carriers. The power spectrum and BER performance of SQ²AM are analyzed and compared with those of QPSK, SQAM and Q²PSK in a nonlinearly amplified multi-channel environment.

An efficient performance enhancement scheme using the rotation of signal subspace (RSS) and Toeplitz matrix approximation (TMA) methods to enhance the performance of an adaptive antenna array in DS/CDMA systems is proposed. The basis of RSS is to find a transformation matrix in order to recover the desired complex array covariance matrix from a sampled complex array covariance matrix which is contaminated by an interference-plus-noise component. Also, the objective of TMA is to change the output matrix of RSS into a matrix having the theoretical properties such as Toeplitz structure matrix or a positive semidefinite matrix. Consequently, the proposed scheme using RSS and TMA methods can greatly improve the performance of an adaptive antenna array by reducing the interference-plus-noise effect from the sampled complex array covariance matrix of the pre-correlation received signal vector that is used to calculate a weight vector of an adaptive antenna array. Simulation results demonstrate the effectiveness of the proposed scheme.

Wireless LAN (WLAN) systems transmit and receive via a common frequency band. In this band, signals of other wireless applications operate on a WLAN beamformer as interferences, and so the problem in adaptive antenna is increasing the canceling performance in the presence of moving interference sources. The performance of conventional adaptive beamformer is severely degraded and the robust adaptive beamformer must be equipped with additional sensors to obtain desired performances. Therefore, in order to avoid having to install additional sensors, an efficient algorithm is necessary. In this paper, we introduce a fast adaptive algorithm with variable forgetting factor, which does not require any further additional modifications. Through computer simulations, we can obtain better performances than those of other techniques under a variety of operating conditions.

This Letter proposes a timer based ordered registration (TBOR) scheme for CDMA private user zone (PUZ) service. The proposed TBOR scheme solves the problem that both the PUZ system loads and the public system loads are increased due to excessive registration. Also the TBOR scheme is quite simple to implement and compatible with the existing public CDMA system.

We propose a new frequency and frame synchronization technique for orthogonal frequency division multiplexing (OFDM) systems in frequency domain by using the channel information. The proposed frequency synchronization scheme shows optimal performance. Also it is shown that the proposed frame synchronization scheme has significantly enhanced performance improvement and wide estimation range.

This letter addresses the QoS issue of a RSVP flow during handoff events. For the QoS guarantee of real-time applications with RSVP reservation in All-IP wireless networks, mechanisms are required to minimize the resource reservation path changes and the packet loss resulting from handoff events. If the new RSVP reservation is made along the path via a new base station (BS) in advance for soft handoff, on-going RSVP flow session can be kept with the guaranteed QoS. Therefore, we propose a seamless switching scheme of RSVP branch path for soft handoff. We also show that this scheme could provide the QoS guarantee by adaptively adjusting the pilot signal threshold values for soft handoff.

To recover the lost packets in VoIP systems, we propose a dynamic redundant audio transmission mechanism that responds to network status. A receiver computes the QoS information using timestamps and sequence numbers in RTP headers and sends the information to a sender using the RTCP protocol. The sender transmits redundant audio packets to the receiver to recover the lost packets. Experimental results show that the rate of recovering the lost packets is 63% in case of consecutive packet loss and is 42% in case of non-consecutive packet loss.

In this letter, we evaluate the performance of several adaptive and non-adaptive AQM schemes for congestion control in a dynamic network environment with variable bandwidth links. The AQM schemes examined are RED, BLUE, Adaptive RED, REM, AVQ and PI controller. We compare their queueing performance and show that none of them can derive stable queue length and low packet drop rate simultaneously in networks where both input traffic and available output bandwidth are time varying. Adaptive and efficient algorithms should be designed and applied in order to improve the adaptiveness and robustness of congestion control in dynamic networks such as Internet.

Though analysis of an indoor propagation channel has been conventionally used the ray-tracing method, in this paper using three dimension of finite difference time domain methods can easily and exactly be obtained three-dimensional complex structures. An excitation signal of FDTD made use of plane wave. An absorbing boundary condition used the most reflection less perfectly matched layer in the outset plane. An empty room surrounded a wall composed of brick, concrete and case that there are indoor furniture in the room were simulated. As simulation outcome, we could identify frequently rising reflection, refraction, scattering of objects and a fading effect of multipath at indoor propagation environment, calculate mean excess delay and rms delay spread for receiver design.

This paper describes the design, fabrication, and measurement of a sequential-rotation microstrip-array antenna for Electronic Toll Collection System (ETCS). The ETCS is made possible by using roadside equipment with a radiation pattern that can accurately pinpoint the designated communication area, without interference from other lanes. The sequential-rotation microstrip-array antenna is designed and an absorber attached to the antenna is considered to reduce the side lobe level (SLL) for the antenna of ETCS. Results show that the antenna yields a return loss at a center frequency of -20.675 dB, an axial ratio of 1.15 dB, and a gain of 20.26 dBi.

We have thoroughly investigated the effect of on-chip decoupling capacitors on the simultaneous switching noise (SSN) and the radiated emission. Furthermore, we have successfully demonstrated an efficient design method for on-chip decoupling capacitors on an 8-bit microcontroller without increasing the die size, which results in more than 10 dB of suppressed radiated emission.