In the case of personal computers (PCs), interoperability among PCs are sufficiently realized with the advent of Microsoft Windows to take the position of mainstream OS, and major software applications following the mainstream for standardization, for more and more user-friendly human machine interface. Considering the case for PCs as above, it is not surprising to us at all, if the same concept is pursued in a radio communications terminal, which can freely access to different radio systems just by replacing the embedded software. This means that the prospective end user will gain the benefit to be able to change his radio set to one of the desired systems in the field, by installing the software of his choice. Such radio equipment is called Software Defined Radio (SDR), and various kinds of applications are expected for development in many fields. However, for the SDR to be in widespread use, we have many outstanding issues to be solved, which are not limited only in the technical matters. One barrier is interoperability among manufacturers. Namely, even when a technical problem is solved, the appropriate technical solution should be shared for the interoperability among as many manufacturers as possible. If such interoperability is unachievable, that technical solution could only be for internal use within the specific manufacturer, failing to take advantage of the true value of the SDR. Another barrier might be the Radio Law of Japan. Unless overcoming this barrier, the commercial implementation of the SDR is unachievable, resulting in the failure to entertain the real benefit from the SDR implication. Under such a background, this paper first describes the concept of and definition for the SDR to make them clearer for the readers. Then, the interoperability issue, which would be the key to the widespread use of SDR, is taken up as next topic. The last topic is focused on the legal and regulatory issue, to discuss what would be the problem under the Radio Law of Japan.

Software radio promises to bring unparalleled flexibility and reconfigurability to wireless systems, with enormous commercial potential. As the next decade progresses SDR is expected by many to emerge as the dominant design in the commercial wireless marketplace. However, significant practical issues associated with security and regulation exist which, if not adequately addressed, could threaten to result in regulatory hurdles precluding, or at least delaying, its deployment--a regulator could be understandably hesitant about authorising the operation of a handset whose radio emissions can be determined by an end user downloading and using unproven software from an arbitrary source post-purchase. In this article we describe the Reconfiguration Management Architecture--a pragmatic technological approach, developed within the framework of Mobile VCE research, that offers solutions to this and other associated SDR problems. The RMA approach fully acknowledges and builds upon the necessary interaction between the user terminal and the network to allow full validation of a reconfigured user device prior to realtime operational authorisation. Such an architecture allows responsibility for validation to be delegated and assigned by a national regulator to, for example, a mobile network operator. Such a capability can, in turn, facilitate the creation and growth of an open market in downloadable software provision, which itself promises to encourage rapid development of new capabilities, applications and innovation. New business models and revenue streams may be expected to result. This article describes the basic technical concepts associated with the RMA, explaining the key functionalities residing within the terminal and the network and their interrelationships. The RMA is presently being evaluated as part of the SDR Forum's security and architecture work. It promises to provide realistic solutions that could accelerate the successful commercial deployment and rollout of SDR technology to the benefit of the industry, across the whole value chain.

Software Defined Radio is beyond the education and initiation phase. The industry is addressing the needs of reconfigurable radio technology development, implementation, and application in a variety of marketplaces. Regulatory decisions are being formulated to facilitate SDR adoption and deployment. Continued dialog and cooperation among the industry organizations is an important factor in the rate of progress.

To promote the commercial implementation of software download for software defined radio (SDR) terminals, a secure method of download is vital. This paper examines the needs of software download for SDR, and proposes a comprehensive system framework within which secure download can be carried out. The features of the proposed system include unique individual encryption to each terminal and secure exchangeability of any cryptographic components. The main goals of the security system are the following: (i) verification of the identity of the source of the software; (ii) control and verification of the integrity of the downloaded data; (iii) disabling of the ability to run unauthorized software on the software defined terminal; (iv) secrecy of the transmitted data. The proposed system is flexible and in harmony with current requirements regarding the SDR security issues.

In this paper, we focus attention on the development of security techniques using software defined radio (SDR) technologies. We propose a new secure download system which uses the characteristics of the field programmable gate arrays (FPGAs) composing the SDR. The proposed system has the novelty that realization of high security encipherment is possible. This is achieved using the characteristic of FPGAs which allows systems to be arranged in a variety of different layouts, as well as by using the configuration information as the key. This unifies the renewal of the key and the encipherment. In addition the proposed system has the merit that it has high security against illegal acquisition such as a wiretapping, and can also be used in conjunction with any other current cipher algorithm. As an evaluation of the security, we show that the proposed system has high immunity to illegal acquisition of software using replay attack, by verification of the protocol as well as by numerical computation. The proposed system can therefore realize high security software downloads based on SDR.

This paper presents the theoretical analysis of subband adaptive array combining cyclic prefix transmission scheme (SBAA-CP) in multipath fading environment. The exact expressions for optimal weights, array outputs and the output signal to interference plus noise ratio (SINR) are derived. The analysis shows that use of the cyclic prefix data transmission scheme can significantly improve the performance of subband adaptive array (SBAA). An example of implementing SBAA-CP as a software antenna is also presented.

When we use an adaptive array antenna (AAA) with the minimum mean square error (MMSE) criterion under the multipath environment, where the receiving signal level varies, it is difficult for the AAA to converge because of the distortion of the desired wave. Then, we need the equalization both in space and time domains. A tapped-delay-line adaptive array antenna (TDL-AAA) and the AAA with linear equalizer (AAA-LE) have been proposed as simple space-temporal equalization. The AAA-LE has not utilized the recursive least square (RLS) algorithm. In this paper, we propose a space-temporal simultaneous processing equalizer (ST-SPE) that is an AAA-LE with the RLS algorithm. We proposed that the first tap weight of the LE should be fixed and the necessity of that is derived from a normal equation in the MMSE criterion. We achieved the space-temporal simultaneous equalization with the RLS algorithm by this configuration. The ST-SPE can reduce the computational complexity of the space-temporal joint equalization in comparison to the TDL-AAA, when the ST-SPE has almost the same performance as the TDL-AAA in multipath environment with minimum phase condition such as appeared at line-of-sight (LOS).

This paper investigates an accurate channel estimation method using the common pilot channel (CPICH) in addition to a dedicated pilot channel (PICH) when the fading correlation between the dedicated PICH and CPICH is high, and clarifies the area in which the proposed channel estimation method is effective for adaptive antenna array transmit diversity (AAA-TD) in the forward link. Computer simulation results elucidate that although a more precise channel estimation is possible by using the primary-CPICH (P-CPICH) transmitted from an omni-directional antenna in addition to the dedicated PICH for the area where the distance, d, between a base station and a mobile terminal is longer than approximately 200 m, no improvement is obtained for the area where the value of d is shorter than approximately 200 m. Meanwhile, by employing the secondary-CPICH (S-CPICH) transmitted with several directional beams in addition to the dedicated PICH, the required average received E_b/N₀ at the average BER of 10^-3 is decreased by approximately 0.4 (0.2-0.4) dB compared to the channel estimation method using only the dedicated PICH regardless of the value of d when the number of antennas is 4 (8).

Adaptive beamforming, using the Conjugate Gradient Normal Equation Residual problem, is applied to a base station array, in the UTRA-TDD up-link. A Wideband Directional Channel Model is used, characterising specific micro-cell, street-type scenarios. These differ in the number of mobile terminals, grouped and placed along the street axis, and on their distances to the base station. Time- and angle-of-arrival spreads, and on-the-air interference content are the main parameters inherently varied and analysed. The average beamforming gain and signal-to-noise ratio are evaluated, also varying the number of array elements. The high number of arriving correlated and closely correlated signals, together with the composed nature of the correlation matrix in the algorithm's cost function, result in that other than the MMSE solutions may lead to the best interference suppression, for the tested scenarios. Among the several weighted interfering power components, the most relevant is due to the delayed signals from all the other links. The combination of the number of arriving orthogonal codes, time-of-arrival and angle-of-arrival spreads condition beamforming performance: the number of array elements affect performance, depending on the mobile terminal distance to the base station, and on the number of active links; for short distances and large number of users, larger time-of-arrival spread degrades beamformer performance, over the opposing effect of angle-of-arrival spread; the number of active users affects beamforming gain especially in the case that delay spreads are larger, i.e., for shorter mobile terminal distances to the base station.

Computing the Eigen Value Decomposition (EVD) of a symmetric matrix is a frequently encountered problem in adaptive (or smart or software) antenna signal processing, for example, super resolution DOA (Direction Of Arrival) estimation algorithms such as MUSIC (MUltiple SIgnal Classification) and ESPRIT (Estimation of Signal Parameters via Rotational Invariance Technique). In this paper the hardware architecture of the fast EVD processor of symmetric correlation matrices for the application of an adaptive antenna technology such as DOA estimation is proposed and the basic idea is also presented. Cyclic Jacobi method is well known for the simplest algorithm and easily implemented but its convergence time is slower than other factorization algorithm like QR-method. But if considering the fast parallel computation of the EVD with a hardware architecture like ASIC (Application Specific Integrated Circuit) or FPGA (Field Programmable Gate Array), the Jacobi method can be a appropriate solution, since it offers a quite higher degree of parallelism and easier implementation than other factorization algorithms. This paper computes the EVD using a Jacobi-type method, where the vector rotations and the angles of the rotations are obtained by CORDIC (COordinate Rotation DIgital Computer). The hardware architecture suitable for ASIC or FPGA with fixed-point arithmetic is presented. Because it consists of only shift and add operations, this hardware friendly feature provides easy and efficient implementation. In this paper, the computational load, the estimate of circuit scale and expected performance are discussed and the validation of fixed-point arithmetic for the practical application to MUSIC DOA estimation is examined.

In this paper, a digital compensation scheme for coefficient errors of a complex filter bank parallel A/D converter in low-IF receivers is presented. The complex filter bank is employed to suppress DC offset and image signals in the low-IF receivers and relax the requirements on the conversion rate and resolution of A/D converters. The proposed compensation scheme regenerates interference due to coefficient errors and subtracts it from the digital signal converted by an A/D converter. The proposed scheme also improves the effective resolution of A/D converters.

This paper proposes and investigates a coding and decoding scheme to achieve adaptive channel coding using a Finite State Machine (FSM) for Software Defined Radio (SDR). Adaptive channel coding and decoding systems that can switch between different coding rates and error correcting capabilities in order to adapt to changing applications and environments, are effective for SDR. However, in these systems, a receiver cannot always select the correct decoder which causes decoding errors, usually referred to as Decoder-Selection-Errors (DSE). We propose a trellis encoder estimation scheme that compensates for this problem. This scheme uses the circuit of FSM to limit the encoder transition and the Viterbi algorithm for maximum likelihood trellis encoder estimation. Computer simulations are applied for evaluating the DSE rate, the Bit Error Rate (BER) and Throughput of the proposed scheme in comparison with a conventional scheme.

This paper proposes a three-step cell search algorithm utilizing a synchronization channel (SCH) and common pilot channel (CPICH) in the forward link for OFCDM (Orthogonal Frequency and Code Division Multiplexing) broadband packet wireless access, and evaluates the cell search time performance by computer simulation. In the proposed three-step cell search algorithm, the OFCDM symbol timing, i.e., Fast Fourier Transform (FFT) window timing is estimated employing SCH or guard interval (GI) correlation in the first step. Then, the frame timing is detected by employing the SCH and the cell-specific scrambling code (CSSC) is identified by the CPICH in the second and third steps, respectively. Computer simulation results elucidate that the proposed three-step cell search algorithm achieves fast cell search time performance, i.e., cell detection probability of 90% within approximately 50 msec, assuming the number of CSSCs of 512 in a 19 hexagonal-cell model. We also clarify that there is no prominent difference in cell search time performance between the two employed SCH structures, time-multiplexed and frequency-multiplexed, assuming that the total transmit power of the SCH is the same. Based on the comparison of four substantial cell search algorithms, the GI-plus-SCH correlation method, in which FFT windowing timing detection, frame timing detection, and CSSC identification are performed by GI correlation, frequency-multiplexed SCH, and CPICH, respectively, exhibits the cell search time of approximately 44 msec at the detection probability of 90% with an optimized averaging parameter in each step.

This paper investigates the impact of inter-carrier interference (ICI) due to Doppler spread on the packet error rate (PER) performance in Orthogonal Frequency and Code Division Multiplexing (OFCDM) packet wireless access employing turbo coding in a multipath fading channel, and describes the optimization of the sub-carrier spacing, Δ f, i.e., the number of sub-carriers, N_c, with an approximate 50-100 MHz bandwidth. Simulation results show that although the uncoded OFCDM in a 1-path flat Rayleigh fading channel is affected by the ICI caused by the Doppler spread when the maximum Doppler frequency, f_D, becomes more than 5% of Δ f, OFCDM employing turbo coding in a 24-path Rayleigh fading channel is robust against Doppler spread and the degradation is not apparent until f_D reaches more than 10% of Δ f. This is because the turbo coding gain and the frequency diversity effect compensate for the degradation due to ICI. Meanwhile, the PER performance with a larger N_c is degraded, since the effect of the error correction capability becomes smaller due to the larger variance of the despread OFCDM symbols associated with the narrower spreading bandwidth in the frequency domain. Consequently, along with the packet frame efficiency for accommodating the guard interval to compensate for the maximum multipath delay time of 1 µsec, we clarify that the optimum number of sub-carriers is approximately 512-1024 (the corresponding Δ f becomes 156.3-78.1 kHz) for broadband OFCDM packet wireless access assuming a 50-100 MHz bandwidth.

A software defined radio (SDR) prototype based on a multiprocessor architecture (MPA) is developed. Software for Japanese personal handy phone system (PHS) of a 2G mobile system, and IEEE 802.11 wireless LAN, which has much wider bandwidth than the 2G systems, is successfully implemented. Newly developed flexible-rate pre-/ post-processor (FR-PPP) achieves the flexibility and wideband performance that the platform needs. This paper shows the design of the SDR prototype and evaluates its performance by experiments that include PHS processor load and wireless LAN throughput characteristics and processor load.

In this paper, we newly developed a small-size software radio terminal that can realize global positioning service (GPS) navigation system, vehicle information and communication system (VICS), electronic toll collection system (ETC), AM/FM radio broadcasting services on middle wave (MW) and very high frequency (VHF) bands, FM multiplex broadcasting system, and several modulation schemes such as BPSK, ASK, QPSK, GMSK, and π/4QPSK by downloading software to realize each system from wired and wireless networks. The developed terminal realizes simultaneous multiple services when users would like to use several radio communication services in the driving situation by using our proposed multitask algorithm. The developed terminal has a size of 17.5 cm wide, 19.0 cm deep, and 5 cm high and worked at DC-12.0 V and around 2 A. The size and electrical power consumption are quite small and low and acceptable for consumers such as car drivers. In this paper, we introduce the configuration and proposed key technologies in our developed terminal and measure the software configuration time.

A new approach to build up a real-time multiprocessing system that is configuration flexible for evaluating space-time (ST) equalizers is described. The core of the system consists of fully programmable devices such as digital signal processors (DSPs), field-programmable gate arrays (FPGAs), and reduced instruction set computers (RISCs) with a real-time operating system (RTOS). The RTOS facilitates flexibility in the multi-processor configuration for the system conforming with ST processing algorithms. Timing jitter synchronization caused by use of the RTOS-embedded system is shown, and an adjustable frame format for a transmission system is described as a measure to avoid the jitter problem. Bit error rate (BER) performances measured in uncorrelated frequency-selective fading channels show that an ST equalizer provides a significantly lower BER than an array processor does.

A basestation power amplifier requires high linearity and stable operation for common amplification of a multiplexed wideband signal. Nonlinearity compensation techniques with the feedforward or predistorter are useful for this purpose. This paper presents a predistortion amplifier with automatic control at RF band for application at a basestation. In this method, the predistorter distorts an RF input signal by referring to a look-up table (LUT) corresponding to the input power. Out-of-band radiation power are directly monitored at IF band to determine the LUT. A DSP with an iterative algorithm updates the content of the LUT to minimize the out-of-band radiation power. Computer simulation experiment is carried out. The use of this proposed method promises a highly linear wideband and high power-efficiency amplifier.

A broadband RF front-end having a direct conversion architecture has been developed. The RF front-end consists of two broadband quadrature mixers, a multi-band local oscillator, and a broadband low-noise variable gain amplifier (LNVGA). The mixer achieves broadband characteristics through the incorporation of an in-phase power divider and a 45-degree power divider. The in-phase power divider achieves broadband characteristics through the addition of a compensation capacitor. The 45-degree power divider achieves broadband phase characteristics through the addition of a compensation capacitor and a compensation resistor. The local oscillator, which is composed of two VCOs, two frequency dividers, and four switches, can cover three systems including one FDD system. The LNVGA achieves its broadband characteristics without the use of reactance elements, such as inductors or capacitors. In a trial demonstration, when the RF frequency was between 900 MHz and 2.5 GHz, the mixer for a demodulator experimentally demonstrated an amplitude balance of less than 1.6 dB and a quadrature phase error of less than 3 degrees. When the RF frequency was between 900 MHz and 2.5 GHz, the mixer for a modulator demonstrated an image ratio of less than -30 dBc. The local oscillator demonstrated multi-band characteristics, which are able to cover the target frequencies for three systems (PDC, PHS, 2.4 GHz WLAN). From 900 MHz to 2.5 GHz, the amplifier shows a noise figure of less than 2.1 dB and a gain of 28 1.6 dB.

This paper discusses a design methodology suitable for the development of software defined radio platforms. A flexible digital receiver was designed and implemented using a multi-port direct converter and an FPGA-based platform. The design starts with a hardware-oriented top-level system model. The model is built based on basic signal processing blocks connected together in a graphical tool. Carrier symbol timing recovery is implemented in the discrete-time (digital) domain with an interpolator-based synchronizer. Carrier phase and frequency are recovered using a feedback synchronization algorithm (a second-order type-II digital PLL). Experimental results of the platform and its simulation results demonstrate the effectiveness of the proposed design methodology.

This paper evaluates the bit error rate (BER) performance of high rate data transmission such as at 64 and 384 kbits/s (kbps) with high quality (average BER is below 10^-6) using turbo/convolutional coding associated with Rake time diversity, antenna diversity, and fast transmission power control (TPC) in multipath fading channels for W-CDMA mobile communications. Laboratory experiments using multipath fading simulators elucidate the superiority of turbo coding over convolutional coding when the channel interleaving length is 40 msec. The required average transmission power for the average BER of 10^-6 using turbo coding is decreased by approximately 1.1-1.5 dB and 1.5-1.6 dB for 64 and 384 kbps data transmissions, respectively, compared to that using convolutional coding for a two-path Rayleigh fading channel with the fading maximum Doppler frequency of f_D = 5-200 Hz. Furthermore, field experimental results elucidate that the required transmission power for the average BER of 10^-6 employing turbo coding is decreased by approximately 0.6 dB and 2.0 dB compared to convolutional coding for 64 and 384 kbps data transmissions, respectively, without antenna diversity reception, while that with antenna diversity reception exhibits only an approximate 0.3-0.5 dB decrease. This decrease in improvement with antenna diversity reception indicates that in an actual fading channel in the field experiments, the impact of the error in path search for Rake combining and SIR measurement for fast TPC diminishes the performance improvement of the turbo coding due to a very low received signal power.

This paper presents an adaptive burst-mode M-QAM modem architecture suitable for variable rate broadband wireless packet data networks. The core signal processing functions for the modem are common to all constellations resulting in an efficient hardware architecture for field programmable gate array (FPGA) implementation.

In this paper, we analyze the convergence and steady-state behavior of the least mean-square (LMS) adaptive filtering algorithm for a finite-length phase-splitting hybrid-type decision feedback equalizer (H-DFE). With some approximations, we derive an iterative expression for the excess mean-square error (MSE) of the H-DFE, which is composed of three statistically dependent excess MSEs; that is, the excess MSEs of the feedforward filter (FFF), intersymbol interference predictive feedback filter (ISI-FBF), and noise predictive feedback filter (NP-FBF) taps. Computer simulation and analytical results show that the average eigenvalue of the input signal for the NP-FBF taps of the H-DFE is time-varying, whereas those for the FFF and ISI-FBF taps are fixed. Nevertheless, the H-DFE can be implemented with fixed step sizes that ensure the convergence of the LMS algorithm without performance degradation from the standpoint of convergence speed, as well as steady-state performance for digital subscriber line (xDSL) applications.

We investigate dispersion compensation using dispersion-compensating fibers (DCFs) for ultrashort light pulse code division multiple access (CDMA) communication systems in a multi-user environment. We employ fiber link that consists of a standard single-mode fiber (SMF) connected with two different types of DCFs. Fiber dispersion can be effectively decreased by adjusting the length ratios of DCFs to SMF appropriately. Some criteria for dispersion compensation are proposed and their performances are compared. We theoretically derive a bit error rate (BER) of ultrashort light pulse CDMA systems including the effects of the dispersion and multiple access interference (MAI). Moreover, we reveal the mutual relations among BER performance, fiber dispersion, MAI, the number of chips, a bandwidth of a signal, and a transmission distance for the first time. As a result, we show that our compensation strategy improves system performance drastically.

An adaptive modulated orthogonal frequency division multiplexing (OFDM) radio transmission scheme that enables efficient data transmission in multipath fading environments is newly proposed. This scheme can be used in standardized multimedia mobile access systems such as ETSI-BRAN, and ARIB-MMAC. It is based on estimating the delay spread and the carrier-to-noise power density ratio (C/N₀). The estimation is done using channel estimation symbols that are inserted into the frames of the standard OFDM radio transmission scheme. Computer simulations show that the estimation method results in an average BER close to those when propagation characteristics are perfectly estimated. Furthermore, when the adaptive OFDM transmission scheme is based on BPSK, QPSK or 16 QAM, the average BER is almost close to that when BPSK-OFDM is only used, and the average transmission rate is 1.8 times as high. Using an error-correction code based on convolutional code results in an average BER lower than that with the BPSK- and QPSK-OFDM schemes.

In this paper, a new approach is proposed to improve the channel estimation accuracy with channel tracking capability for adaptive multicarrier equalization systems under time-variant multipath fading channel. The improvement is carried out based on the assumption that the channel is static over a transmitted block period, and slowly linearly changing over several block periods. By applying IFFT to the concatenated channel transfer function derived from different blocks, the noise-averaging improvement is achieved, and a better estimation of the channel coefficients with some delay can be obtained. A multi-step channel predictor and a smoothing filter is utilized to compensate for the delay and make the system more robust in terms of channel tracking performance. Adaptive time domain equalization is jointly performed with this approach to avoid the channel invertibility problem found in the frequency domain approach. A short period of training sequences is utilized resulting in more efficient use of available communication capacity. The effectiveness of the proposed approach is evaluated through simulation for multicarrier systems in time-variant multipath fading channels. Results show improvement over previous channel estimation schemes.

A receiver structure, which has linear computational complexity with the number of users, is proposed for decoding multiuser information data in a convolutionally coded asynchronous DS-CDMA system in multipath fading channels. The proposed receiver architecture consists of a multiuser likelihood calculator followed by a bank of soft-input soft-output (SISO) channel decoders. Information is fed back from SISO channel decoders to multiuser likelihood calculator, and the processing proceeds in an iterative fashion analogous to the decoding of turbo codes. A simplification to the above receiver structure is given too. Simulation results demonstrate that for both receiver structures at high signal-to-noise ratios (SNR) both multiple-access interference (MAI) and inter-symbol interference (ISI) are efficiently suppressed, and single-user performance is approached. Furthermore, the proposed iterative receiver is near-far resistant.

In OFDM-CDMA down link (base-to-mobile) transmissions, each user's transmit data symbol is spread over a number of orthogonal sub-carriers using an orthogonal spreading sequence defined in the frequency-domain. The radio propagation channel is characterized by a frequency- and time-selective multipath fading channel (which is called a doubly selective multipath fading channel in this paper). Frequency-domain equalization is necessary at the receiver to restore orthogonality among different users. This requires accurate estimation of the time varying transfer function of the multipath channel. Furthermore, the noise enhancement due to orthogonality restoration degrades transmission performance. In this paper, pilot-aided threshold detection combining (TDC) is presented that can effectively suppress the noise enhancement. If the estimated channel gain is smaller than the detection threshold, it is replaced with the detection threshold in the frequency equalization. There exists an optimal threshold that can minimize the bit error rate (BER) for a given received E_b/N₀. The average BER performance of OFDM-CDMA down link transmissions using the TDC is evaluated by computer simulations. It is found that TDC using optimum detection threshold can significantly reduce the BER floor and outperforms DS-CDMA with ideal rake combining.

As demonstrated by many studies, measured wide-area network traffic exhibits fractal properties, such as self-similarity, burstiness, and long-range dependence (LRD). In order to describe long-range dependent network traffic and to emphasize the performance aspects of descriptive traffic models with additive and multiplicative structures, the multifractal wavelet model (MWM), which is based on the binomial cascade, has been shown to match the behavior of network traffic over small and large time scales. In this paper, using appropriate mathematical and statistical analyses, we develop the MWM proposed in [14], which provides a complete description of long-range dependent network traffic. First, we present accurate parameters of the MWM over different time scales. Next, a marginal stable distribution of MWM network traffic data is analyzed. The accuracy of the proposed MWM compared to actual data measurements is confirmed by queuing behavior performance through computer simulations.

The modern network service of finding the optimal path subject to multiple constraints on performance metrics such as delay, jitter, loss probability, etc. gives rise to the multi-constrained optimal-path (MCOP) QoS routing problem, which is NP-complete. In this paper, this problem is solved through both exact and heuristic algorithms. We propose an exact algorithm E_MCOP, which first constructs an aggregate weight and then uses a K-shortest-path algorithm to find the optimal solution. By means of E_MCOP, the performance of the heuristic algorithm H_MCOP proposed by Korkmaz et al. in a recent work is evaluated. H_MCOP only runs Dijkstra's algorithm (with slight modifications) twice, but it can find feasible paths with a success ratio very close to that of the exact algorithm. However, we notice that in certain cases its feasible solution has an unsatisfactorily high average cost deviation from the corresponding optimal solution. For this reason, we propose some modified algorithms based on H_MCOP that can significantly improve the performance by running Dijkstra's algorithm a few more times. The performance of the exact algorithm and heuristics is investigated through computer simulations on networks of various sizes.

In ATM networks, call processing on switches can be greatly simplified by using the concept of virtual path (VP); and good resource management strategies ensure that virtual channel connections (VCC) can be rapidly and efficiently established. In order to have good system performance, several methods of constructing virtual paths and strategies of allocating and managing resources should be considered. In this paper, several multicast strategies with dynamic routing are used and applied to the metropolitan LATA network. For the VP-based network, dynamic routing is also applied, and those strategies are discussed and investigated to show the versatility of the approach. Some results using dynamic multicast routing, such as call blocking rate, VP utilization, and VP adjustment rate, are obtained for the different strategies by simulation experiments.

For real-time systems, multiprocessor support is indispensable to handle the large number of requests. Existing on-line scheduling algorithms such as Earliest Deadline First Algorithm (EDF) and Least Laxity Algorithm (LLA) may not be suitable for scheduling hard real-time tasks on multiprocessors. Although EDF has a low context switching overhead, it can produce arbitrarily low processor utilization. LLA has been shown as suboptimal, but has the potential for higher context switching overhead. We propose new on-line scheduling algorithms Earliest Deadline/Least Laxity (ED/LL) and Earliest Deadline Zero Laxity (EDZL) for identical multiprocessors. We show that ED/LL is suboptimal for multiprocessors and EDZL is suboptimal for two processors. Experimental results show that ED/LL and EDZL have low context switching overhead and low deadline miss rate.

In this paper, we propose a packet-scheduling algorithm, called the Class-level Service Lagging (CSL) algorithm, that guarantees multiple delay bounds for multi-class traffic in packet networks. We derive the associated schedulability test conditions, which are used to determine call admission. We first introduce a novel implementation of priority control, which has a conventional and simple form. We show how the efforts to confirm the logical validity of that implementation are managed to reach the definition of the CSL algorithm. The priority control is realized by imposing class-level unfairness in service provisioning, while the underlying service mechanism is carried out using the notion of fair queueing. The adoption of fair queueing allows the capability to maintain the service quality of the well-behaving traffic even in the presence of misbehaving traffic. We call this the firewall property. Simulation results demonstrate the superiority of the CSL algorithm in both priority control and firewall functionality. We also describe how the CSL algorithm is implementable with a computational complexity of O(1). Those features as well as the enhanced scalability, which results from the class-level approach, confirm the adequacy of the CSL algorithm for the fast packet networks.

The network duplicating can achieve significant improvements of the Local Area Network (LAN)'s performance, availability, and security. For LAN duplicating, a Dual-Path Ethernet Module (DPEM) is developed. Since a DPEM is simply located at the front end of any network device as a transparent add-on type independent hardware machine, it does not require sophisticated server reconfiguration. We examine the desirable properties and the characteristics on the Dual-LAN structure. Our evaluation results show that the developed scheme is more efficient than the conventional Single-LAN structures in various aspects.

In this paper, we propose an efficient quality-providing scheme to satisfy delay bound and loss ratio requirements for real-time video applications. To utilize network resources more efficiently while meeting service requirements, the network resources are dynamically allocated to each video connection based on the predicted traffic and currently provided quality of service degree. With the proposed bandwidth allocation method, a fair quality of service support in terms of packet loss ratio and maximum packet transfer delay to each video source can be achieved. To avoid possible quality violation by incoming new video connections, we present a connection admission control based on the provided QoS for existing connections and the measured traffic statistics. Simulation results show that our proposed dynamic method is able to provide accurate quality control.

Asymmetric networks, which provide asymmetric bandwidth or delay for upstream and downstream transfer, have recently gained much attention since they support popular applications such as the World Wide Web (WWW). HTTP (Hypertext Transfer Protocol) is the basis of most WWW services so, evaluating the performance of HTTP on asymmetric networks is increasingly important, particularly real-world networks. However, the performance of HTTP on the asymmetric networks composed of satellite and terrestrial links has not sufficiently evaluated. This paper proposes new formulas to evaluate the performance of both HTTP1.0 and HTTP1.1 on asymmetric networks. Using these formulas, we calculate the time taken to transfer web data by HTTP1.0/1.1. The calculation results are compared to the results of an existing theoretical formula and experimental results gained from a system that combines a VSAT (Very Small Aperture Terminal) satellite communication system for satellite links (downstream) and the Internet for terrestrial links (upstream). The comparison shows that the proposed formulas yield more accurate results (compared to the measured values) than the existing formula. Furthermore, this paper proposes an evaluation formula for pipelined HTTP1.1, and shows that the values output by the proposed formula agree with those obtained by experiments (on the VSAT system) and simulations.

This paper proposes a novel method to correct numerical phase velocity errors in FDTD meshes with nonuniform step size. It enables the complete compensation of the phase velocity errors introduced by the mesh grading for one frequency and one arbitrary direction of propagation independently of the mesh grading. This permits the usage of the Total-Field-Scattered-Field formulation in connection with electrically large nonuniform FDTD meshes and allows a general reduction of the grid dispersion errors. The capabilities of the proposed method are demonstrated with the help of two examples: (1) the fields in a dielectric sphere illuminated by a plane wave are calculated and (2) a patch antenna simulation demonstrates that the uncertainty in determining its resonance frequency can be reduced by about 50%.

The problem of the super-resolution time delay estimation of the real stationary signals is addressed in this paper. The time delay estimation is first converted into a frequency estimation problem. Then a MUSIC-type algorithm to estimate the subsequent frequency from the single-experiment data is proposed, which not only avoids the mathematical model mismatching but also utilizes the advantages of the subspace-based methods. The mean square errors (MSEs) of the time delay estimate of the MUSIC-type method for varying signal-to-noise (SNR) and separation of two received signal components are shown to illustrate that they approximately coincide with the corresponding Cramer-Rao bound (CRB). Finally, the comparison between the MUSIC-type method and the other conventional methods is presented to show the advantages of the proposed method in this paper.

Fundamental EMI source that generates common-mode radiation from printed circuit boards (PCBs) is investigated here. It is done by modelling the ground lines of PCBs as imperfect ground. The radiation emission in the far zones from PCBs is obtained by regarding interconnects on PCBs as transmission lines and the far field emission is evaluated based on the current distribution of the lines. The finite size ground trace is defined as an imperfect ground, that can be viewed as an inductive reactance which, in turn, causes the ground return path to radiate as a wire antenna. For the accurate analysis of imperfect ground effect, we divide the equivalent circuit into N sections. In addition, based on transverse electromagnetic (TEM) assumption, we estimate the electromagnetic interference (EMI) of three typical PCB geometries, namely, coplanar strips, parallel-plate strips and microstrips. The quantitative value of induced current distribution along the ground return path depends on the physical size, geometry and length of ground traces. Measured data are presented to confirm the result of numerical analysis. A knowledge of EMI source mechanisms and their relationship to layout geometries is necessary to determine the essential features that must be taken into account to estimate emissions and provide direction for reducing EMI due to interconnects on PCBs.

In this paper, a new method is proposed for supervised classification of ground cover types by using polarimetric synthetic aperture radar (SAR) data. The concept of similarity parameter between two scattering matrices is introduced for characterizing target scattering mechanism. Four similarity parameters of each pixel in image are used for classification. They are the similarity parameters between a pixel and a plane, a dihedral, a helix and a wire. The total received power of each pixel is also used since the similarity parameter is independent of the spans of target scattering matrices. The supervised classification is carried out based on the principal component analysis. This analysis is applied to each data set in image in the feature space for getting the corresponding feature transform vector. The inner product of two vectors is used as a distance measure in classification. The classification result of the new scheme is shown and it is compared to the results of principal component analysis with other decomposition coefficients, to demonstrate the effectiveness of the similarity parameters.

We clarify the effectiveness of receiver-side compensation in offsetting fiber Bragg grating (FBG) dispersion induced-electrical signal-to-noise ratio (SNR) degradation in a 10 Gb/s 8-channel wavelength-division multiplexing (WDM) 6,400 km transmission system. The receiver-side compensation greatly improves the SNR degradation. The allowable accumulated FBG dispersion is -400 1000ps/nm for the worst arrangement, a single FBG at the transmitter, which is about half the accumulated fiber dispersion permissible with receiver-side compensation.

We propose an enhanced packet access scheme for IMT-2000/UMTS random access channel (RACH). In the proposed scheme, 2-level preamble detection threshold and 2-level message transmission power are used to mitigate the power imbalance in RACH. Simulation results demonstrate that the proposed scheme improves the interference characteristics of the conventional RACH and makes wider range of the detection threshold available.

This paper proposes a new algorithm named WGF (Weight-Gap First) scheduling. It not only supports the maximum throughput of a cell, but also provides the fairness of a terminal in HDR. The WG-based new scheduling algorithm is determined by served data of mobile terminals to guarantee requested data rate. WGs are renewed at each timeslot to improve throughput and fairness. Simulations confirm the better performance of the proposed algorithm in terms of throughput and fairness.

This letter derives low-complexity frame and coarse frequency-offset synchronization techniques for orthogonal frequency division multiplexing (OFDM)-based HIPERLAN (HIgh PErformance Radio LAN) system. We first propose a frame detector structure directly based on the correlation method and a reduced complexity structure having the similar performance compared with conventional correlation method. We then propose a coarse frequency-offset synchronization technique and show the performance of the proposed techniques by simulation.

The optimization of path bandwidth allocation in large-scale telecommunication networks is studied. By introducing a decomposition-coordination scheme to global optimization of the path bandwidth allocation which aims at minimizing the worst case call blocking probabilities in the network, the spatial and time complexities are both reduced, while the accuracy is almost the same as that given by direct optimization.

An attempt to derive the lethal dose for mice was made at 2.45 GHz for whole body exposure. Based on a numerical dosimetry result and an experimental death rate investigation, the lethal dose was estimated to be a whole body averaged specific absorption rate (SAR) with a level at double the mouse's basal metabolic rate.

In this letter, we address geometry coding of 3-D mesh models. Using a joint prediction, the encoder predicts vertex positions in the layer traversal order. After we apply the joint prediction algorithm to eliminate redundancy among vertex positions using both position and angle values of neighboring triangles, we encode those prediction errors using a uniform quantizer and an entropy coder. The proposed scheme demonstrates improved coding efficiency for various VRML test data.

Among the wireless personal area networks, much attention has been placed on the HomeRF system due to the simple hardware complexity. In this letter, we propose several techniques for the HomeRF system. First, a DC-offset compensation technique is considered for HomeRF system. In addition, a decision directed channel estimation technique is proposed. The proposed techniques require no additional training sequence and a little additional hardware. And finally, a turbo coding technique is considered as a improved coding scheme. It is shown that the performance of the proposed algorithm is significantly improved in comparison with the conventional HomeRF system.