Affine projection algorithms perform well for acoustic echo cancellation and adaptive equalization. Although these algorithms typically provide fast convergence, they are unduly complex when updating the weights of the associated adaptive filter. In this paper, we propose a new subband affine projection (SAP) algorithm and a facile method for its implementation. The SAP algorithm is derived by combining the affine projection algorithm and the subband adaptive structure with the maximal decimation. In the proposed SAP algorithm, the derived weight-updating formula for the subband adaptive filter has a simple form as compared with the normalized least mean square (NLMS) algorithm. The algorithm gives improved convergence and reduced computational complexity. The efficiency of the proposed algorithm for a colored input signal is evaluated experimentally.

The performance of the vehicular communication links used for Intelligent Transport Systems is investigated. Intervehicle communication (IVC) and combined vehicular communication implemented by IVC and additional communication media are studied and their performance is explicitly described. Through numerical studies, it is shown that performance varies according to parameter values such as the mean space headway, the speed of the vehicles, and the penetration ratio of the IVC device. To achieve a given level of performance, I propose (i) a design of the information delivery delay of additional communication media and (ii) a method determining the appropriate delay.

Combined input-crosspoint buffered (CICB) switches relax arbitration timing and provide high-performance switching for packet switches with high-speed ports. It has been shown that these switches, with one-cell crosspoint buffer and round-robin arbitration at input and output ports, provide 100% throughput under uniform traffic. However, under admissible traffic patterns with nonuniform distributions, only weight-based selection schemes are reported to provide high throughput. This paper proposes a round-robin based arbitration scheme for a CICB packet switch that provides 100% throughput for several admissible traffic patterns, including those with uniform and nonuniform distributions, using one-cell crosspoint buffers and no speedup. The presented scheme uses adaptable-size frames, where the frame size is determined by the traffic load.

This paper compares selected Optical Packet Switching architectures that use the wavelength conversion technique to solve the packet contention problem. The architectures in question share wavelength converters, which are needed to wavelength translate arriving packets. This paper focuses on two architectures: the Shared Per Output Line (SPOL) and the Shared Per Input Line (SPIL) architectures, in which the wavelength converters are shared per output and input fiber respectively. The performance of the proposed architectures is evaluated for all the balance/unbalance combinations of input/output traffic. Packet loss probability is expressed as a function of the number of wavelength converters used, by means of analytical models validated by simulations. The results obtained show that the SPIL architecture, when compared to the SPOL architecture, allows for greater economies in terms of number of wavelength converters needed. While the performance of the two architectures tends to have similar values in a scenario with unbalanced input traffic and balanced output traffic, in unbalanced output traffic scenarios the SPIL architecture requires about 50% less wavelength converters than the SPOL architecture does, for a given packet loss probability.

In this paper, we propose Split Multi-path Routing protocol with Load Balancing policy (SMR-LB) to improve TCP performance in mobile ad hoc networks. In SMR-LB, each intermediate node records how many primary paths are attempted to construct as well as which source nodes attempt to construct the primary path. Each intermediate node decides which primary path should be constructed by using the primary path and the source node ID information. As a result, SMR-LB can balance the loads and so reduce the probability of congestion and avoid the continuous link breakage time between the specific source and destination pair. Computer simulation results show that SMR-LB can improve TCP performance compared with the conventional protocols.

Metro networks are based on SONET/SDH which uses the circuit switching technology. Circuit switching is inappropriate for the Internet traffic which is very bursty nature. Therefore, metro networks can become a bottleneck. In order to resolve this problem, the IEEE 802.17 working group has proposed the Resilient Packet Ring (RPR) technology. In order to provide fairness among stations in the RPR network, two types of RPR fairness mechanisms have been defined. However, the IEEE 802.17 RPR fairness mechanisms have the problem of inefficient use of the available bandwidth after the congestion at a node has been resolved. In this paper, we propose an improved bandwidth allocation in which, after the congestion resolution at a node, the node estimates the number of effective nodes transmitting traffic, measures the remaining bandwidth and fairly allocates the available bandwidth to effective nodes. To show the performance of our proposed mechanism, we have performed simulation and show that the proposed mechanism gives higher bandwidth utilization than the existing RPR fairness mechanisms.

Live streaming is delay sensitive and can tolerate some amount of loss. The QoS Multicast for Live Streaming (QMLS) Protocol, focuses on the characteristics of live streaming. It has been shown to improve the performance of live streaming multicast by reducing the end-to-end packet loss probability. However, the placement of active routers performing the QMLS function has not been discussed. This paper proposes a dynamic method to activate and deactivate routers in order to minimize the number of active routers for each QMLS-packet flow and discusses its parameters. The results of an evaluation show that the proposed method can reduce the number of active routers for each flow and adjust the active routers according to changes in the multicast tree.

Recently, voice over WLAN has become an attractive service, and it is expected to be the most popular application in the near future due to its low cost and easy deployment. It has been reported that there occurs unfairness between downlink and uplink in the 802.11 WLAN. This is mainly caused by CSMA/CA employed in DCF. All stations including an AP fairly compete for shared wireless medium. Thus, in particular, the unfairness has an adverse impact on bi-directional voice calls. Downlink voice connections become a primary factor to limit voice capacity. In this paper, we propose a novel medium access protocol, so called DCFmm, in order to improve QoS of downlink voice traffic as well as fairness between bi-directional voice connections. DCFmm is designed to enhance 802.11 DCF, and is fully compatible with the legacy DCF. In addition, it requires only protocol modifications of an AP. Thus, it can be easily implemented into existing 802.11 WLANs. DCFmm is compared with two conventional techniques through computer simulations. Extensive simulation results show that the proposed DCFmm can improve fairness between downlink and uplink, and consequently, support larger number of voice calls than DCF.

Researchers have proposed numerous approaches to providing Quality-of-Service (QoS) across the Internet. The IETF has proposed two reservation approaches: hop-by-hop bandwidth reservation (IntServ); and per-hop behaviour bandwidth reservation (DiffServ). An edge router generates traffic, accepts per-flow reservation and classifies them into predetermined service class; while a core router ensures different QoS guarantees for each service class. We propose an Edge-to-Edge Quality-of-Service Domain in which packet trains with the same service requirements aggregated using packet deadline at edge router. The properties of a packet train like Inter-Packet Departure Time, Inter-flow Departure Time and accumulated packet delay are embedded and used by our quantum-based scheduler and QoS packet forwarding scheme in core routers. Thus, we are able to extract per-queue and per-flow information. Each queue is reconstructed at core router with packets having an expected departure time that is relative to the ingress router. Useful functions like instantaneous service rate and fine granular dropping scheme can be derived with a combination of embedded information and relative virtual clock technique. The encapsulation of our packet train information converges mathematically. Through simulations, we show that our architecture can provide delay and rate guarantees and minimise jitter for QoS-sensitive flows that requires LR-coupled or LR-decoupled reservations.

In this paper, we propose a protocol for multicast communication called Multicast Datagram Transfer Protocol (MDTP) to provide multicast for video broadcasting service on the Internet. MDTP is a one-to-many multicast communication protocol, which is constructed based on IPv4 unicast protocol by utilizing IP Router Alert Option, and it uses unicast addressing and unicast routing protocol. A mechanism is presented to allow a router to remove identical video stream, to duplicate a video stream, and to forward each copy of the duplicated video stream to its destinations. Ordinary IP routers that do not support MDTP will treat the MDTP packets as normal unicast packets. Hence, gradual deployment is possible without tunneling technique. With a delegation mechanism, MDTP router is also able to handle request from clients, and serve the requested video stream. The simulation results show that the average bandwidth usage of MDTP is close to the average bandwidth usage of IP multicast. MDTP also has greater efficiency than XCAST, and its efficiency becomes significant for a large number of clients.

Tunneling is commonly used in several transition mechanisms. Some of the mechanisms discover the tunnel endpoint automatically by their own means. This paper describes several tunnel discovery mechanisms for IPv6 traversal beyond IPv4 legacy networks and suggests a new mechanism by which the DHCPv4 server can automatically provide configuration information about IPv6-in-IPv4 tunnel. Dual stack nodes attached to IPv4 network can communicate with other IPv6 networks by this mechanism beyond IPv4 networks without user intervention. Implementations have shown that the proposed mechanism can be a convenient and valuable tunnel discovery mechanism in some cases.

Post-FFT type orthogonal frequency division multiplexing (OFDM) adaptive array antennas can reduce the co-channel interference with a few antenna elements under multi-path fading environments. However, the Post-FFT type OFDM adaptive array antennas require a lot of pilot symbols in order to determine the optimal weights in each subcarrier. In packet communication systems, since the data are transmitted burst by burst, the ratio of the effective data in a channel decreases when the long pilot symbols are used. Recursive least squares (RLS) algorithm is one of the weight optimization algorithm with fast convergence based on minimum mean square errors (MMSE). However, the optimal weight determination with a few pilot symbols is difficult. Therefore, in this paper, we propose a novel multi-stage RLS OFDM adaptive array antenna for realizing weight determination with a few pilot symbols. In the proposed method, the weights are optimized by using a multiple stage structure with the stored pilot symbols. Here, the initial weights and the initial inverse matrix of correlation matrix are decided by the results of the weight determination in the adjacent subcarriers of the previous stage. As a result, the weight determination with a few pilot symbols can be achieved.

Adaptive modulation and coding scheme (AMC) is an effective way to achieve high data rate communication. In AMC system, the key issue is to determine the rule for switching among different Modulation and Coding Schemes (MCSs). In this paper, adaptive threshold controlling scheme for AMC is proposed. The proposed scheme controls switching thresholds according to target block error rate. Simulation results have shown that the throughput performance of the proposed scheme is very close to the performance, which obtained by the optimum SIR thresholds. We also proposed downlink transmission power control (TPC) scheme suitable for AMC. The throughput of the lowest MCS is improved and the tranmission power of the highest MCS can be reduced with the proposed algorithm.

The medium access control (MAC) protocol is the main determiner of the system throughput in Wireless Local Area Networks (WLANs). The MAC technique of the IEEE 802.11 protocol is called Distributed Coordination Function (DCF). DCF is based on a carrier sense multiple access with collision avoidance (CSMA/CA) scheme with binary slotted exponential backoff. Each station generates a random backoff interval before transmitting a packet to minimize the probability of collision with packets being transmitted by other stations. However, when the number of stations increases, the system throughput decreases. This paper proposes a new backoff algorithm that uses finish tags. The proposed algorithm uses the finish tag of each station to control the backoff intervals so as to improve system throughput. The finish tag is updated when a packet reaches the front of its flow, and it is attached to the packet just prior to transmission. When a station receives packets with older finish tags, its backoff time interval is increased. For this reason, the more the stations there are, the larger the backoff time becomes. Simulations confirm that the proposal improves system throughput of a IEEE 802.11 network under saturation conditions.

Recently with the high expectation of voice over WLAN service, to support fast inter-AP security transition in WLAN Access Point (AP) is one of the most actively investigating issues. It is also very important problem to minimize inter-AP security transition (IAPST) latency, while maintaining constantly the secure association from old AP when a station transits to new AP. With this background, this paper presents a novel association control mechanism whose objective pursues to minimize IAPST latency time and to take countermeasures against attacks of rogue transition station. Experiment shows that the proposed scheme outperforms the legacy AP over 70% with regard to the transition latency.

Wireless multi-hop networks have drawn much attention for the future generation mobile communication systems. These networks can establish multiple routes from a source node to a destination node because of flexible construction of network topology. Transmissions by multiple routes have enough capability to achieve reliable communication because we can expect to obtain diversity gain by multiple routes. In this paper, we propose the multi-route coding scheme. At first, we discuss a channel model in multi-hop networks employing regenerative relay, which we named the virtual channel model. By using the virtual channel model, a packet is encoded on multiple routes as follows; a bit sequence of a packet is encoded and divided into subpackets, and each subpacket is transmitted on each route. We evaluate its packet error rate performance, and clarify effectiveness of the proposed scheme. In general, we should face degradation of a route condition such as the case when a subpacket does not reach a destination node. Hence, we have to consider the influence of subpacket loss. We also investigate it, and show tolerance of the proposed scheme over that.

The authors have proposed a 1 m² single-layer slotted waveguide array consisting of conducting baffles and quartz glass strips positioned in front of the slot aperture, which is referred to as a vacuum window, for microwave plasma excitation. The effect of the complicated outer vacuum window hinders the realization of uniform distribution. In this paper, a unit-cell of the alternating-phase fed single-layer slotted waveguide array with the vacuum window is analyzed by generalized scattering matrix method (GSM)-method of moments (MoM) hybridization analysis, and the array is designed to realize uniform aperture electromagnetic field distribution, where the plasma and the chamber is neglected. The GSM-MoM analysis gives reliable numerical results while the MoM has numerical errors due to singularities of Green's function for a long cavity. Uniform aperture EM field distribution outside of the vacuum window is observed in near field measurements using a 1/5 scale model antenna, and the validity of the analysis and design is verified.

A probe-fed U-shaped cross-sectional antenna with tuning stubs on a U-shaped ground plane is proposed for wideband applications. The bottom of the antenna is etched to form tuning stubs for impedance matching. The simulated results of return loss, co- and cross-polarized patterns are presented and compared with the measured ones. Characteristics of a constructed antenna prototype at the operating frequency show that the antenna has an impedance bandwidth (2:1 VSWR) of 37.44% and average gain level of 8.5 dBi. Good radiation characteristics of the proposed antenna have been obtained that is the cross-polarization level and front-to-back ratio in both E- and H-planes across the large bandwidth are better than 22 dB and 12 dB, respectively.

Interfaces between a coaxial structure and a post-wall waveguide are proposed as the essential components for cost-effective millimeter-wave modules. PTFE substrate is selected in terms of loss and manufacturability. The reflection and the transmission characteristics are investigated. The short-stepped and the short-taper-stepped feeding structures provide 14.7% and 13.2% bandwidths for the reflection smaller than -15 dB, respectively. The 4640 mm² size antenna fed by the short-stepped structure in PTFE substrate gives 27.3 dBi with 58.2% efficiency at 60.0 GHz. Feeding structures in PTFE substrate fulfill electrical and manufacturing demands in millimeter-wave bands.

This paper presents a modal-expansion analysis of the electromagnetically coupled coaxial dipole antenna. The analysis of the antenna problem is initially simplified using the even-odd mode excitation and then the resultant half structure is divided into two parts; one is the characterization of a coaxial feeding network and the other is the modeling of a sleeve monopole antenna driven by a coaxial line. The formally exact modal-expansion method is employed to analyze both parts. The analysis of the sleeve monopole antenna is facilitated by introducing a perfectly conducting boundary at a distance from the monopole's top end. The current distribution and input impedance of the electromagnetically coupled coaxial dipole antenna are obtained by finding expansion coefficients through enforcing the continuity of tangential field components across regional interfaces and cascading the two parts together. Numerical results for the coaxial dipole antenna's radiation characteristics are presented and discussed.

This paper describes a two-dimensional clustering scheme-based analysis of audible noises induced at telephone terminals. To analyze EMI sources that cause telephone-audible noise, we use a self-organizing map, which provides a way to map high-dimensional data onto a two-dimensional domain. Also, in order to discriminate EMI sources without using particular resonance frequencies that have peaks in the frequency domain, we use the energy spectra of telephone-audible noises as input for training the self-organizing map. In applying this method in actual environments, we measured ten kinds of telephone-audible noises (due to Radio waves and cross-talk noises, etc.) and then derived their energy spectra for eight frequency bands: 1-250 Hz, 250-500 Hz, 500-1 kHz, 1 k-1.5 kHz, 1.5 k-2 kHz, 2 k-3 kHz, 3 k-4 kHz, and over 4 kHz. We visually confirmed that the measured telephone-audible noise data could be projected onto the map in accordance with their properties, resulting in a combined depiction of the composition of derived energy spectra in the frequency bands. The proposed method can deal with multi-dimensional parameters, projecting its results onto a two-dimensional space in which the projected data positions give us an effective depiction of EMI sources that cause disturbances at telephone terminals.

Recent studies of high quality, high resolution traffic measurements have revealed that network traffic appears to be statistically self similar. Contrary to the common belief, aggregating self-similar traffic streams can actually intensify rather than diminish burstiness. Thus, traffic prediction plays an important role in network management. In this paper, Least Mean Kurtosis (LMK), which uses the negated kurtosis of the error signal as the cost function, is proposed to predict the self similar traffic. Simulation results show that the prediction performance is improved greatly over the Least Mean Square (LMS) algorithm.

The quadrature component unbalance generated by a non-ideal component such as an imperfect 90-degree phase shifter is an inevitable physical phenomenon and leads to performance degradation in a practical coherent M-ary phase shift keying (MPSK) transceiver. In this letter, we present an exact and general expression involving the one- and two-dimensional Gaussian Q-functions for the symbol error rate (SER) of MPSK with I/Q phase unbalance over an additive white Gaussian noise (AWGN) channel. The SER expression provided here offers a convenient way to evaluate the performance of MPSK systems for various cases of practical interest.

This paper describes a max-min fair rate allocation scheme in multicast networks based on the supervisory control framework for discrete event systems. Based on the discrete event model of multicast networks, we design a supervisor to guarantee the max-min fair allocation of bandwidth. Analysis and simulations are used to show that the controlled networks guarantee the max-min fair sharing with a low message exchange overhead and fast convergence time.

For backward compatibility, ECN-capable networks should be capable of handling both ECN-capable and ECN-incapable TCP flows. In this letter, we present a backward congestion notification (BCN) scheme that can provide fast congestion indication delivery, while improving fairness between ECN-capable and ECN-incapable flows. Simulation results reveal that the BCN scheme is more effective than the original ECN mechanism in terms of stability, throughput, and fairness.

We have developed an open-architecture router (OAR) prototype using industrial standard hardware, software components, and interfaces. The prototype is built with Advanced Telecom Computing Architecture (ATCA)-compliant hardware. Carrier-grade Linux (CGL) is used as the operating system. A new OAR configuration method is described where industrial standard hardware and software interfaces are used. Basic forwarding functions with routing protocol processing are demonstrated for the first time.

The complexity of the optimal decoding for vector quantization (VQ) in code-division multiple access (CDMA) communications prohibits implementation. It was recently shown in [1] that a suboptimal scheme that combines a soft-output multiuser detector and individual VQ decoders provides a flexible tradeoff between decoder's complexity and performance. The work in [1], however, only considers an AWGN channel model. This paper extends the technique in [1] to a frequency-selective Rayleigh fading channel. Simulation results indicate that such a suboptimal decoder also performs very well over this type of channel.

In this letter, multi user detection process in Code Division Multiple Access (CDMA) is performed by fuzzy inference system (FIS) and the bit error rate (BER) performance was compared with the single user bound, the matched filter receiver and neural network receiver. The BER performance of the matched filter receiver degrades as the number of the active users and the power level differences among the users increase. The neural network receiver needs the training process. Optimal receiver that has the best BER performance is too complex for practical application. The BER performance near the optimal case was obtained with the proposed receiver. The proposed receiver uses the FIS without training process and it has lower time complexity than the optimal receiver.

In this letter, we propose a combined scheme of minimum mean square error (MMSE) detection and successive interference cancellation (SIC) for multiuser space-time block coded orthogonal frequency division multiplexing (STBC-OFDM) systems. With the same complexity order, the proposed scheme provides significant bit error rate (BER) performance improvement over the linear MMSE multiuser detector.

This letter investigates the effects of the Space-Time Block Code (STBC) transmitter diversity and Time Switched Transmitter Diversity (TSTD) schemes while simultaneously considering the time diversity effect derived from Chase combining and the frequency diversity effect for downlink Spread OFDM packet wireless access in a broadband multipath fading channel. The simulation results show that when pilot channel based channel estimation is performed, TSTD improves the required average received signal energy per symbol-to-noise power spectrum density ratio by approximately 1.0 and 0.5 dB compared to one-branch transmission, without and with antenna diversity reception, respectively, while the benefit of STBC is small due to channel estimation error, when the maximum number of retransmissions in Chase combining is three in a 12-path Rayleigh fading channel with the root mean squared delay spread of 0.3 µsec and the maximum Doppler frequency of 20 Hz.

A novel blind channel estimation scheme is proposed for OFDM systems employing PSK modulation. This scheme minimizes the number of possible channels by exploiting the constant modulus property, chooses a best fit over the possible channels by exploiting the finite alphabet property of information signals, and achieves competitive performance with low computational complexity. Results comparing the new scheme with the finite-alphabet based channel estimation are presented.

In orthogonal frequency-division multiplexing (OFDM)-based wireless local area networks (WLANs), phase noise (PHN) and residual frequency offset (RFO) can cause the common phase error (CPE) and the inter-carrier interferences (ICI), which seriously degrade the performance of systems. In this letter, we propose a combined pilot symbol assisted and decision-directed channel estimation scheme based on the least-squares (LS) and the maximum-likelihood (ML) algorithms. Simulation results present that the proposed scheme significantly improves the performance of OFDM-based WLANs.

This paper proposes a low complexity composite CDMA system based on MIMO (Multiple-Input-Multiple-Output) processing and LDPC (low-density parity-check) codec based a CDMA system, which works well even if the complicated case of multipath, multiusers and short length LDPC codes. To explore the practical application, the mobile user in the composite CDMA systems is with only two antennas and adopts short length irregular LDPC codec, each user's data has been divided two LDPC encoded substreams being sent to two transmitter antennas at base station. Since the LDPC encoded substreams of reaching the mobile user are orthogonal to each other in space and time, the CDMA system performances (BER and SINR) can be improved much, but the multipath may ruin the orthogonalilty. To solve the problems, the paper provides the algorithms of main function modules of transmitter and receivers, gives a simple method to test the girth of LDPC codes, and analyzes the performance of MIMO-LDPC CDMA systems theoretically and experimentally. The simulation results show that the hybrid CDMA systems can have better performance than the conventional CDMA systems based on single transmitted antenna at a base station.

In this letter, we present timing synchronization method using two consecutive orthogonal frequency division multiplexing (OFDM) symbols which are designed to be cyclic-shifted against each other. Our approach can be viewed as an expansion of Minn's correlation methods. Using the proposed OFDM signal, however, we can estimate the timing offset without any training symbol.

A novel technique is proposed for measuring the distributed strain and temperature in a fiber with a very high resolution. This technique makes use of the jagged appearance of Rayleigh backscatter traces from a single-mode fiber measured by using a coherent OTDR with a precisely frequency-controlled light source. Our preliminary experiment indicated the possibility of measuring temperature with a resolution of better than 0.01 and a spatial resolution of one meter. This temperature resolution is two orders of magnitude better than that provided by Brillouin-based distributed sensors.

The release of image processing techniques make image modification and fakery easier. Image fakery, here, is defined as a process to copy a region of source image and paste it onto the destination image, with some post processing methods applied, such as boundary smoothing, blurring, etc. to make it natural. The most important characteristic of image fakery is object copy and paste. In order to detect fake images, this letter introduces a blind detection scheme based on singular value decomposition (SVD). Experimental results also show the effectiveness of the proposed scheme.