In this paper, we consider QoS-guaranteed wavelength allocation for WDM networks with limited-range wavelength conversion. In the wavelength allocation, the pre-determined number of wavelengths are allocated to each QoS class depending on the required loss probability. Moreover, we consider two wavelength selection rules and three combinations of the rules. We analyze the connection loss probability of each QoS class for a single link using continuous-time Markov chain. We also investigate the connection loss probability for a uni-directional ring network by simulation. In numerical examples, we compare connection loss probabilities for three combinations of selection rules and show how each combination of selection rules affects the connection loss probability of each QoS class. Furthermore, we show how wavelength conversion capability affects the connection loss probability. It is shown that the proposed allocation with appropriate wavelength selection rule is effective for QoS provisioning when the number of wavelengths is large. We also show the effective combination of wavelength selection rules for the case with small wavelength conversion capability.

Because of the polarization dependencies of optical fiber transmission equipment, the polarization state of lightwaves inherently varies at the optical receiver input. Therefore the receiver output, especially the beat component, is very dependent on the polarization state. We derive a simple relation showing that the beat power of the lightwaves is proportional to the inner product of their Stokes vectors. Using this relation, we can systematically calculate the beat power. This calculation method can be applied to lightwaves of arbitrary polarization, such as a polarization-scrambled signal or partially polarized amplified spontaneous emission (ASE) in long-haul IM/DD systems, as well as to the signal and local oscillator lightwaves of coherent systems.

We numerically and experimentally investigated the upgradability of the longest and the typical segments of the JIH system. Through these studies, we confirmed that a 100 GHz-spaced 25 42.7 Gbit/s transmission with the total capacity of 1 Tbit/s can be attainable even by using NRZ signal and standard FEC for the typical segments. We also found that RZ signal format was desirable for the longest segment and a further wide system margin could be expected by using adjacent channel polarization control and advanced FEC technologies.

The accuracy of channel estimation significantly affects the performance of coherent rake receiver in DS-CDMA systems. It is desirable for improved channel estimation to employ a channel estimation filter (CEF) whose bandwidth is adjustable to the channel condition. In this paper, we consider the use of moving average (MA) FIR filters as the CEF since it is simple to implement and can provide relatively good receiver performance. First, we optimize the tap size of the MA FIR CEF so as to minimize the mean squared error of the estimated channel impulse response. For practical applications, we propose a low-complexity adaptive channel estimator (ACE), where the tap size of the MA FIR CEF is adjusted based on the estimated channel condition by exploiting the correlation characteristics of the received pilot signal. Numerical results show that the use of the proposed ACE can provide the receiver performance comparable to that of Wiener CEF without exact a priori information on the operating condition.

In this paper, we proposed a novel two-stage channel estimation (2S-CE) method. In contrast to conventional channel estimation methods, this method makes the maximum use of the information contributed by the known data in every transmission burst. In the first stage, the least-squares (LS) algorithm was used to estimate the channel impulse response (CIR) based on the normal training sequence. Then the maximum channel memory was estimated and used to locate the uncorrupted data in the guard interval. In the second stage, the uncorrupted data together with the normal training sequence were sent to the LS algorithm again to obtain the fine-tuned CIR. To verify the efficiency of the proposed 2S-CE method, both a theoretical analysis and computer simulations have been done. Computer simulation results confirm the analysis results and demonstrate that the proposed 2S-CE method outperforms a conventional single-stage channel estimation method.

An automatic-repeat-request (ARQ) scheme for improving the system throughput efficiency is evaluated in coded multiple-input multiple-output (MIMO) transmissions. Supplementary trellis-coded modulation (TCM) code has been proposed for hybrid ARQ schemes. The free distance of the TCM after code combining can be increased by employing different TCM codes for retransmissions. The MIMO scheme offers additional flexibility in preventing successive frame errors by changing the connections between transmitters and transmit antennas upon retransmission. In this paper, an ARQ strategy employing both TCM reassignment and antenna permutation technique is investigated. It is shown through computer simulations that this ARQ scheme achieves high throughput even in severe conditions of low signal-to-noise ratio and high Rician factor over spatially and temporally correlated Nakagami-Rice fading MIMO channels.

Multi-Carrier Code Division Multiple Access (MC-CDMA) is one of candidates for the next generation wireless communication systems. In an uplink, the MC-CDMA system suffers from the different access timing (asynchronous transmission), the different fading, and the different frequency offsets of each active user. In this paper we analyze the effects of the frequency offset compensation with MMSE-MUD (minimum mean square error based multi-user detection) for MC-CDMA in a quasi-synchronous uplink. We consider the MC-CDMA system with two subcarrier mapping schemes, the continuous mapping scheme and the discrete mapping scheme. From our theoretical analysis and computer simulation, we show that the MMSE-MUD can compensate the different frequency offsets among users. We also show that the MMSE-MUD significantly improves the bit error rate (BER) for the MC-CDMA system with the continuous mapping scheme.

In this article we analyze the performance of a space division output buffered multichannel switch operating in an ATM multimedia environment as follows. Fixed size packets belonging to two classes arrive onto the switch inputs in each time slot. Class-1 packets, representing real time traffic such as live audio and live video communications, are sensitive to delay but insensitive to loss and have their own service time needs. Class-2 packets, representing nonreal time communications such as file transfers, are insensitive to delay but sensitive to loss and have their different service time needs. To respond to the class-1 delay sensitivity, the switch gives class-1 packets higher service priority over class-2. And to respond to the difference in service time needs, the switch operates at two service rates, one for each class. This latter assumption is the major feature of the article, as previous studies have usually assumed that the two classes have the same service needs and thus the same service rate. For the purpose of the analysis, the switch is modelled as a priority, discrete time, batch arrival, multiserver queueing system, with infinite buffer and two geometric service times with two parameters. Performance measures analyzed are system occupancy and packet waiting time.

We had previously proposed a fuzzy logic-based buffer management scheme (BMS) called fuzzy early detection (FED), which was designed to improve transmission control protocol (TCP) performance over the unspecified bit rate (UBR) service of asynchronous transfer mode (ATM) networks. Since a weakness in FED was discovered later, we present a refined version of it named FED+ here. Maintaining the design architecture and the algorithm of FED, FED+ further adopts a specific per virtual connection accounting algorithm to achieve its design goals. The effects of TCP implementation, TCP maximum segment size, switch buffer size and network propagation delay on FED+ performance are studied through simulation. Its performance is then compared with those of pure early packet discard (EPD), P-random early detection (P-RED) and FED. Our evaluations show that FED+ is superior to the others if the issues of efficiency, fairness, robustness, buffer requirement and the ease of tuning control parameters of a BMS are considered collectively.

This paper proposes a novel method of identifying the design parameters for a practical implementation of the fair queueing discipline, which is capable of class-level delay control. The notion of class weight is introduced at first, and then the session weights are determined. This two-phase approach is favorable in terms of the scalability;that is, the overall complexity is dependent upon the number of classes only. We propose a packet scheduler referred to as the DPS (Delay-centric Processor Sharing) scheme which employs those design parameters to deliver class-wise delay bound services. The associated admission policy for delay guarantee is also derived. System analysis and derivation of the parameters have their origins in the understanding of the so-called system equation, which describes the dynamics of the class-level service share. The proposed design parameters are QoS-aware in that they are consistently refined depending on the system status. Several numerical and simulation results show that the DPS scheme is advantageous over other ones in terms of both resource efficiency and the robustness. Concerning the scalability, we show that an alternative tagging process of the DPS scheme is implementable with O(1) complexity with no significant degradation in delay performance.

In the last few years we have seen an explosion in the number of notebook computers and in the growth of the Internet. Mobile users expect to access the Internet's information resources and to communicate with other Internet users. The concept of Mobile IP is proposed to satisfy these demands. Using the base Mobile IP protocol, all datagrams destined for a mobile node are routed through that mobile node's home agent, which intercepts and tunnels each datagram to the mobile node's current location. This tunneling scheme creates a triangle routing problem, causing packets to travel through the home agent. In this paper, we propose that a Proxy Home Router (PHR) for routing performance improvement in a Mobile IP environment. One or more of Intermediate Systems (IS) on the route path between the correspondent node (CN) and the home agent (HA) are established as a PHR which then delivers packet instead of the home agent (HA). This PHR route optimization (PHR-RO) scheme has the following features. It is compatible with the base Mobile IP with existing Internet computers, and applications. Also, it is unnecessary for a CN to be modified in the proposed architecture. We also define Route Optimization messages in order to optimize route to a mobile node. Using these protocol extensions, a PHR may cache the binding information of a mobile node, and then tunnel their datagrams for the mobile node (MN) directly to the care-of address (CoA), bypassing the mobile node's home agent (HA). To analyze the performance of our PHR route optimization (PHR-RO) scheme, we propose a simulation model and show how to improve routing performance. Through this performance analysis, we conclude that the route optimization with a proxy home router (PHR) has better performance in terms of the end-to-end delay and TCP throughput.

TCP (Transmission Control Protocol) is the prevalent reliable transport protocol used for many kinds of the popular Internet services like web browsing and e-mail. Because these services are the dominant applications on the Internet, TCP can be said to control the large majority of today's Internet traffic and we can say the enhancement of TCP performance means the improvement of the most Internet performance. This paper compares various ACK generation TCP strategies that play very important roles in TCP operation and proposes advanced delayed ACK algorithms to mitigate the negative effect of delayed ACK algorithm. Even if delayed ACK has the bright side to save network resources and power consumption that are the very important factors especially for mobile and wireless environment, it has also dark side not to take full advantage of network bandwidth in Slow Start phase. It is reported that delayed ACK timeout of the first and last segment causes severe degradation problem in TCP throughput performance. We suggest new algorithms to solve delayed ACK penalty and to guarantee fairness with other traffics in Slow Start. In wired-cum-wireless network environments,'BADA (Base station Aided Delayed ACK)' with SNDA (Sender Notification Delayed ACK), the combination of our proposed algorithm shows the best performance in terms of throughput as well as in terms of network resource sharing. Our simulation shows significantly improved performance in TCP throughput.

Recent growth in computer communications has led to an increased requirement for high-speed backbone networks. In such high-speed networks, the principle adopted for a time-sensitive flow control mechanism should be that of autonomous decentralized control. In this mechanism, each node in a network manages its local traffic flow only on the basis of the local information directly available to it, although it is desirable that the individual decisions made at each node lead to high performance of the network as a whole. In our previous studies, we have investigated the behavior of local packet flows and the global performance achieved when a node is congested, and proposed the diffusion-type flow control model. However, since we used a simple and homogeneous network model in the evaluation, the results cannot be generalized. In this paper, we propose an extension of the diffusion-type flow control model in order to apply it to networks with inhomogeneous configurations. We show simulation results for two cases: different propagation delays and multiple bottlenecks. Both results show that the proposed diffusion-type flow control achieves high and stable performance even if the network is congested.

Packet loss is a serious problem due to the shortage of optical buffers in all-optical packet switched networks. In order to reduce packet losses, a dynamic routing method called 'deflection routing' has been proposed. Deflection routing, however, requires an optical switch to modify routing tables and packet labels for overflowing packets, so this routing method may also lead to other implementation problems in packet routing and forwarding. This paper proposes a simple routing method called 'reflection routing' which utilizes optical transport links as optical buffers to improve the quality of service in optical packet switched networks in terms of packet loss ratio. We numerically demonstrate the effectiveness and applicability of reflection routing.

Generalized Multi-Protocol Label Switching (GMPLS) is being developed in the Internet Engineering Task Force (IETF). In GMPLS-based wavelength-division-multiplexing (WDM) optical networks, a wavelength in a fiber is used as a label. In the existing GMPLS signaling protocol for bidirectional paths in WDM networks with the wavelength continuity constraint, bidirectional path setup fails with high probability because the upstream label allocated by the previous hop node may not be accepted at the transit node. To solve this problem, this paper proposes an efficient bidirectional label switched path (LSP) setup scheme based on an upstream label set. Called the Upstream Label Set (ULS) scheme, it is an extension of the existing GMPLS signaling protocol. The ULS scheme is consistent with the existing GMPLS signaling procedure and so offers backward compatibility. The numerical results suggest that when the number of the LSP setup retries is limited, the ULS scheme offers lower blocking probability than the existing GMPLS signaling scheme which uses only with the upstream label (UL). In addition, under the condition that the constraint of the number of LSP setup retries is relaxed, the LSP setup time of the ULS scheme is faster than that of the existing scheme. Furthermore, by using our developed prototype of the GMPLS control system, in which the ULS scheme was installed, we demonstrated that the ULS scheme successfully setup bidirectional LSPs.

This paper presents two dynamic multi-layer routing policies for optical IP Networks. Both policies first try to allocate a newly requested electrical path to an existing optical path that directly connects the source and destination nodes. If such a path is not available, the two policies employ different procedures. Policy 1, which has been published already, tries to find available existing optical paths with two or more hops that connect the source and destination nodes. Policy 2, which is proposed in this paper, tries to establish a new one-hop optical path between source and destination nodes. The performances of the two routing policies are evaluated. Simulation results suggest that policy 2 outperforms policy 1 if p is large, where p is the number of packet-switching-capable ports; the reverse is true only if p is small. We observe that p is the key factor in choosing the most appropriate routing policy.

This paper presents a network architecture with a dual interface IP handoff technique that allows smooth node mobility without using any intermediate proxy. The proposed architecture is suitable for low bit-rate time sensitive real time applications, where payload tends to be short and packet header overhead is particularly significant. Connections are established as per permanent addresses of the nodes but are carried on by the IP layer according to the temporary addresses by address translation within the end hosts. The mapping information is maintained by database servers, which can be placed in the Internet in a distributed manner. We describe the architecture and show its mobile capabilities by prototype implementation and performance evaluation. Furthermore a dual-interface handoff suitable to the proposed architecture is also introduced. Preliminary results show that the proposed architecture has significantly low overheads. It is compatible with the existing infrastructure and works fine in both IPv4 and IPv6 environments. Analysis also shows that with dual-interface handoff it is possible to achieve seamless handoff without any packet loss by exploiting overlapping coverage area and speed of the mobile node. Handoff latency is reduced significantly as compare to MIPv6. We believe that with more powerful network interface card drivers our concept of dual interface handoff can be realized.

Applications for transforming money or personal information are increasingly common on the Internet and in mobile communications. These applications require user authentication for confirming legal users. One-time password authentication methods change the verifier every time by sending the present verifier along with the next verifier. However, such methods risk attacks because those protocols use two verifiers every session. The SAS (Simple And Secure password authentication protocol) is a one-time password authentication method that the method uses a hash function five times, but it requires high overhead on low spec machines. In this paper, we propose a new method, SAS-2, which reduces overhead of hash function adaptation by 40%. This method has a mutual authentication phase, which maintains synchronous data communications in its authentication procedure. Moreover, SAS-2 can be applied to key-free systems.

As the Internet proliferates, there has been a growing interest in supporting multiparty collaborative applications. It has led to the emergence of many-to-ma ny reliable multicast. Congestion control is a key task in reliable multicast along with error control. However, existing tree-based congestion control schemes such as TRAMCC and MTCP are designed for one-to-many reliable multicast and have some drawbacks when they are used for many-to-many reliable multicast. We propose an efficient congestion control mechanism, MTRMCC, for tree-based many-to-many reliable multicast protocols. The proposed scheme is based on the congestion windowing mechanism and a rate controller is used in addition. The feedback for error recovery is exploited for congestion control as well to minimize the overhead at the receivers. The ACK timer and the NACK timers are set dynamically reflecting the network traffic changes. The rate regulation algorithm in the proposed scheme is designed to help the flows sharing the same link to achieve the fair share quickly. The performance of the proposed scheme is evaluated using network simulator ns-2. The simulation results show that the proposed scheme outperforms TRAMCC in terms of intra-session fairness and supports responsiveness, TCP-friendliness, and scalability.

This paper investigates the relations between IP network performances and the speech quality of the Voice over IP (VoIP) service through extensive experiments on a test bed network. The aim is to establish an effective and practical methodology for telecommunications operators to manage the quality of VoIP service via the management of IP network performances under their control. As IP network performances, utilization of the bottleneck link in the test bed and the following statistical factors of VoIP packets are examined: the standard deviation of delay variations (jitters), the standard deviation of packet interarrival times, and the packet loss ratio. On the other hand, VoIP speech quality is monitored as the Perceptual Evaluation of Speech Quality (PESQ). To investigate the relations under various network conditions, the experiments are performed by varying the following network related parameters of the test bed: the bandwidth of the bottleneck link, the size of the bottleneck buffer, the propagation delay, and the average of the data sizes transmitted as background data traffic. Statistical analyses of the experimental results suggest that managing the standard deviation of jitters in a network serves as a promising methodology, because its close relation to VoIP speech quality possesses robustness to changes in the network conditions. The robustness makes it practically useful since telecommunications operators can apply it to their networks, which are subject to change. The findings in this paper have opened up new visions for telecommunications operators to manage the Quality of Service (QoS) of VoIP service.

A post-wall waveguide-fed parallel plate slotted array is an attractive candidate for high efficiency and mass producible planar array antennas for millimeter wave applications. For the slot design of this large sized array, a periodic boundary wall model based on the assumption of infinite array size and a parallel waveguide is used. In fact, the aperture is large but still finite (10-40 wavelength) and the TEM-like wave is perturbed due to the narrow walls at the periphery of the aperture as well as the slot coupling; antenna efficiency is affected by the size and the aspect ratio of the aperture. All these observations imply the unique defects of oversized waveguide arrays. In this paper, the aperture efficiency of post-wall waveguide arrays is assessed as a function of size and aspect ratio of the aperture for the first time, both in theory and measurement. An effective field analysis for an electrically large oversized waveguide array, developed by the author, is utilized for determining the slot excitation coefficients and aperture illumination. It is predicted that the oversized waveguide array has a potential efficiency of 80-90% if the aperture is larger than 18 wavelength on a side and the gain is more than 30 dBi. A transversely wide aperture generally provides higher efficiency than a longitudinally long aperture, provided a perfectly uniform TEM wave would be launched from the feed waveguide.

Radars utilizing ultra-wide-band (UWB) pulses are attractive as an environment measurement method for various applications including household robots. Suitable filtering is essential for accurate ranging, which requires an accurate waveform estimation. This paper presents a high-resolution algorithm of estimating target location and scattered waveforms, whose accuracies are interdependent. The technique relies on iterative improvements of estimated waveforms. Description of the algorithm is followed by statistical simulation examples. The performance of the algorithm is contrasted with conventional ones and statistical bounds. Results indicate that our proposed algorithm has a remarkable performance, which is close to the theoretical limit. Next, we clarify the problem of applying HCT to multiple targets. HCT for multiple targets can not be used as an estimated waveform because of interference waves from other targets. We propose an interference suppression algorithm based on a neural network, and show an application example of the algorithm.

This paper takes full advantage of polarimetric scattering parameters and total power to classify polarimetric SAR image data. The parameters employed here are total power, polarimetric entropy, and averaged alpha angle (alphabar). Since these parameters are independent each other and represent all the scattering characteristics, they seem to be one of the best combinations to classify Polarimetric Synthetic Aperture Radar (POLSAR) images. Using unsupervised classification scheme with iterative Maximum Likelihood classifier, it is possible to decompose multi-look averaged coherency matrix with complex Wishart distribution effectively. The classification results are shown using Pi-SAR image data set comparing with other representative methods.

Compressed video bitstream is sensitive to errors that may degrade the reconstructed images severely even the bit error rate is small. One approach to combat the impact of error is error concealment at the decoder without increasing the bit rate and changing the encoder. We propose motion vector based error concealment algorithms to recover the motion vector per pixel instead of that per block according to the relation of neighboring motion vectors. The displacement per pixel can be estimated more accurately by using the tendency of neighboring motion vectors. Besides, we use not only the relation among motion vectors, but also the pixels. The pixels of the error block are divided into different parts according to their consistency with neighboring blocks and the displacement at each pixel of these parts is interpolated by relative motion vectors. From simulation results, the proposed motion vector based methods provide better reconstruction quality for damaged images than other methods.

VoIP is one of the key technologies for recent telecommunication services. The quality of its services should be discussed in subjective terms. Since subjective quality assessment is time-consuming and expensive, however, objective quality assessment which estimates subjective quality without carrying out subjective quality experiments is desirable. This paper discusses the performance of the objective quality measure that was standardized as ITU-T Recommendation P.862 and clarifies the quality factors that can be evaluated with satisfactory accuracy based on it. We found that P.862 can be applied to the evaluation of coding distortion, tandeming of codecs, transmission bit-errors, packet loss, and silence compression in a codec, at least for clean Japanese speech. In addition, we propose a method of estimating the subjective quality evaluation value from objective measurement results and show the validity of this method. We also evaluate the uniqueness of objective quality assessment based on P.862 from the viewpoints of the effect of measurement noise and the variation of test speech samples, and propose how to improve the reproducibility of objective quality assessment.

A new inter-client synchronization framework employing a server-client coordinated adaptive playout and error control toward one-to-many (i.e., multicast) media streaming is discussed in this paper. The proposed adaptive playout mechanism controls the playout speed of audio and video by adopting the time-scale modification of audio. Based on the overall synchronization status as well as the buffer occupancy level, the playout speed of each client is manipulated within a perceptually tolerable range. By coordinating the playout speed of each client, the inter-client synchronization with respect to the target presentation time is smoothly achieved. Furthermore, RTCP-compatible signaling between the server and group-clients is performed to achieve the inter-client synchronization and error recovery, where the exchange of controlling message is restricted. Simulation results show the performance of the proposed multicast media streaming framework.

An RF CMOS active inductor with a novel loss compensation circuit network is proposed. Performance of this active inductor can be improved by adding a novel network, which simultaneously reduces parallel and series losses. Consequently, this technique not only increases Q value, inductance, and operating frequency, but also reduces power consumption and circuit complexity. Simulation results show that better performance indices can be achieved, such as minimum total equivalent loss of 1 mΩ, maximum Q value about 3E5, and inductance value from 20 nH to 45 nH in the RF range of 0.6 GHz to 1.6 GHz. Power dissipation is around 1.76 mW under 2.5 V dc supply voltage.

In this letter, adaptive modulation and coding (AMC) is combined with multiple input multiple output (MIMO) multiplexing to improve the throughput performance of AMC. In addition, a system that adopts selection transmit diversity (STD) in the AMC-MIMO multiplexing system, AMC-STD-MIMO multiplexing system, is proposed. STD in our simulation selects 2 transmission antennas from 4 antennas and AMC-MIMO multiplexing process operates with the selected antennas. The computer simulation is performed in a flat Rayleigh fading channel. The results show that the proposed AMC-STD-MIMO multiplexing system achieves the SNR gain of 4.0 dB, compared to the AMC-MIMO multiplexing system at 3 Mbps throughput.

An upgrade scheme using identical sending lights and local lights for optical amplifier systems based on homodyne detection is described. Optimum receiving loss for maximizing the SNR is analytically derived for the proposed configuration where the local taped lights are low level to avoid a sending power decrease. Derived formulas describing the optimum loss condition and the maximum SNR show that 10 Gbit/s systems can be cost effectively upgraded to 40 Gbit/s systems with the same repeater spacing.

This letter shows the performance comparisons of several different rate scheduling schemes for non-real time data service over the uplink of burst switching-based direct sequence code division multiple access (DS/CDMA) system to support the integrated voice/data service. The closed-form solution of optimal scheduling formulation, which minimizes average transmission delay when all of the active data users are transmitting simultaneously, is presented and mathematical analyses with other rate scheduling schemes, which provide efficiency criterion of transmission delay for rate scheduling schemes, are performed. Numerical results show the analyses explicitly.

Uplink synchronous transmission has been proposed to improve the uplink capacity of DS-CDMA systems by means of canceling interference from the main paths of other intra-cell users. A significant capacity gain has been reported in a single cell environment. This Letter further investigates the uplink capacity in a multiple cell environment, where two crucial factors are taken into account, namely code shortage problem and soft handover. The impacts of the target E_b/I_o and the other-to-own cell interference ratio, together with the number of channelisation codes, are discussed mathematically and then, confirmed through system level simulations with more realistic parameters.

The generalized fading amplitude can be expressed using the Nakagami-m distribution. The probability density function (PDF) for the sum of m-distributions is needed to evaluate performance of diversity combining technique such as equal gain combining (EGC) receiver. The approximated PDF for the sum of m-distributions gives simpler performance expression and reduces the computational complexity in evaluating EGC performance. We investigate the normalized approximation error of EGC performance. From the observed result, even for m > 2, the approximated BER is not accurate. For example, the normalized error is about 32% for m = 2 with 3 identical and independently distributed (IID) fading branches and 10 dB SNR.

We consider a blind estimation of the vector channel for systems with receive diversity. The objective of this paper is to reduce the complexity of the conventional subspace-based method in vector channel estimation. A reduced-complexity estimation scheme is proposed, which is based on selecting a column of the covariance matrix of the received signal vectors. The complexity and performance of the proposed scheme is investigated via computer simulations.

In this letter, we propose an adaptive turbo coded modulation scheme for orthogonal frequency division multiplexing (OFDM), and three power allocation algorithms with transmit power variable over subcarriers, variable over subbands and constant over subcarriers. Our object is to improve the overall throughput under target bit-error-rate (BER). Simulation results show that our fixed power adaptation scheme exhibits a more than 7 dB signal-noise-ratio (SNR) gain relative to non-adaptive turbo coded modulation, and an about 2 dB additional gain can be achieved with our variable power algorithm employed. We also discuss the effect of the number of subbands on throughput performance in our adaptive scheme.

In this letter, a new extended recursive least squares (RLS) algorithm is proposed for the identification of fading channels. We extend the standard RLS algorithm by converting the linear regression model into a state-space model. The unknown terms of the extended model are obtained by estimating the values which minimize the mean squared error (MSE). The proposed algorithm has lower computational complexity than the Kalman filter combined with the hypermodel described in, and exhibits superior performance in simulation than the existing RLS algorithms, namely the exponentially weighted RLS algorithm with a fixed forgetting factor (EW-RLS), and the RLS algorithm with a variable forgetting factor (VFF-RLS).

In this letter, we investigate a decimated selective mapping (SLM) method for the peak-to-mean envelope power ratio (PMEPR) reduction in an OFDM system. Under the condition of the same side information (SI) bits, the SLM can be implemented by decimating OFDM samples, which is less complex compared to the ordinary SLM incurring a slight degradation of the PMEPR performance. The decimated SLM (DSLM) approach can be generalized to a multiple-antenna OFDM system employing a space-time block coding (STBC).

This paper considers the problem of providing relative service differentiation in IEEE 802.11 Wireless LAN by using different Medium Access Control (MAC) parameters for different service classes. We present an analytical model which predicts the saturation throughput of IEEE 802.11 Distributed Coordination Function with multiple classes of service. This model allows us to show that relative service differentiation can be achieved by varying the initial contention window alone. In this case, the saturation throughput of a station can be shown to be approximately inversely proportional to the initial contention window size being used by that station. The simulation results validate our analytical model.

We propose a new data link protocol with an adaptive frame length control scheme for satellite networks. The wireless communication channel in satellite networks is subject to errors that occur with time variance. The frame length of the data link layer is another important factor that affects throughput performance in dynamic channel environments. If the frame length could be chosen adaptively in response to changes in the dynamically varying satellite channel, maximum throughput could be achieved under both noisy and non-noisy error conditions. So, we propose a frame length control scheme that acts adaptively to counter errors that occur with time variance. We model the satellite channel as a two-state Markov block interference (BI) model. The estimation of the channel error status is based on the short-term bit error rate and the duty cycle of noise bursts. Numerical and computer simulation results show that the proposed scheme can achieve high throughput for both dense and diffuse burst noise channels.

In this letter, we investigate the path length priority effect of existing just-enough-time (JET) scheme for optical burst switching (OBS) in the multiple hop network environments. And, we propose a novel hop-by-hop priority increasing (HPI) scheme using the input fiber delay lines (FDLs) at each node. Simulation results showed that the proposed FDL/delayed reservation (DR) with HPI scheme can avoid the path length priority effect and enhance the end-to-end throughput in multiple hop network environments.

In this letter, we propose an effective cooperative caching system under the assumption that each web object is accessed randomly. Under this assumption, the access frequency per unit time is given by Poisson distribution and the probability distribution of the web object in the future is derived. Based on this probability distribution, one can obtain the criterion to allocate the web objects with more access expected to the cache servers closer to clients. It is also shown that there is a tradeoff between the precision to allocate objects and the efficiency of caching.

In General Packet Radio Service (GPRS), combined mobility management is defined to efficiently perform both GSM and GPRS mobility management in a combined manner. In this letter, an enhanced scheme for combined mobility management based on a new mobile station (MS) state model is proposed. The steady state probabilities of the proposed MS state model are derived and the performance of the proposed scheme is analyzed. The results show that the proposed scheme outperforms the conventional scheme, especially for low mobility MSs.

Internet traffic engineering is much important for Internet Service Providers (ISPs) today, since it can be used to fully utilize already deployed network resources. For ISPs, the requirements for traffic engineering should be simple, easy to configure, cost-effective and efficient. Based on these considerations, we propose an algorithm called Web First Adaptive Traffic Engineering (WFATE). Since World Wide Web (WWW) services dominate most of the total Internet traffic and WWW flows are not long-lived, we only apply load balancing to WWW traffic in the algorithm. It can be shown that the number of coexistent WWW flows at an ingress node is almost certainly below a bound, and thus a forward-per-flow mechanism without keeping track of the state of each flow is feasible. This mechanism can balance traffic load at fine granularity and therefore get better performance. Through simulations and performance comparison, it is shown that WFATE is quite efficient, which can improve the network throughput averagely by 26% under the "dense source" traffic pattern and 9% under the "sparse source" traffic pattern.

The Internet and mobile communication systems are being developed, and related applications for managing personal information require user authentication for confirming legitimate users. One-time password authentication methods secure user's authorities by changing the verifier every time. The S/Key is a famous one-time password authentication scheme, which is based on Lamport's scheme. T.-C. Yeh et al. have point out security problems of the S/Key scheme and have proposed a variant of the S/Key scheme, which can be applied to smart cards. However, this method risks certain attacks, too. Those two proposed schemes use counter value, which can easily be modified by an attacker. Herein we discuss security problems of the S/Key and Yeh-Shen-Hwang's password authentication schemes using forgery attacks and stolen-verifier attacks.

Sequential estimation of arrival angles allows us to resolve closely located sources that the standard MUSIC fails to do so. A new sequential estimation method is proposed which utilizes only the signal subspace components of the steering vectors for some estimates of the arrival angles. It is theoretically shown that the asymptotic performance of the proposed method is better than that of the conventional sequential method which exploits both the signal and the noise subspace components. Simulation results show that the former outperforms the latter in correlated sources as well as in uncorrelated sources.

A low cost printed omni-directional spiral-mode mono-pole antenna for wireless communication applications has been designed. This antenna has an integrated open-stub to adjust impedance matching. The 10 dB bandwidth is 336 MHz that achieved 13.7 and radiating characteristics are presented.

This letter describes a dual band planar inverted-F antenna (PIFA) with non-uniform meander-line shaped slot suitable for the mobile environment scenario, which operates at the GSM 900 MHz and GSM 1800 MHz (DCS) bands. This antenna structure overcomes the lack of height of the mobile phone. In a practical mobile handset, the bandwidths of the antenna for return loss -8.5 dB are 240 MHz at 900 MHz and 250 MHz at 1800 MHz. Good impedance bandwidth performance for the dual-band is observed. The advantage of the design suggested in this letter is its simplicity of manufacturing and low cost.

The current letter extends narrow band (NB) local polynomial approximation (LPA) beamforming to wide band (WB) rapidly moving sources. Instead of the conventional beamformer weight in NB LPA, the proposed method adopts the steered minimum variance (STMV) method that can achieve a high resolution with short time observations. The performance of the proposed algorithm is demonstrated via computer simulations.