This paper outlines the perspectives on Software Defined Radio (SDR) technology in viewpoint of the standardization of the future mobile communication systems. The activities of ITU-R SG8 Working Party 8F (WP8F) and mITF (mobile IT Forum) of Japan for systems beyond IMT-2000 (B3G) or 4-th generation mobile systems are firstly summarized. The latest discussions relating to SDR technology in the both parties are reported. It is followed by consideration on both expectations and technical issues on SDR in order to realize the technology in the future mobile communication systems. They are clarified in the viewpoint of standardization activity on B3G. Also some regulation issues are lastly summarized.

The Communications Research Laboratory (CRL) started a new project named the New Generation Mobile Network Project in April 2002. The target of this project is the development of new technologies to enable seamless and secure integration of various wireless access networks such as 3rd and 4th generation cellular, wireless LAN, high-speed mobile wireless, wired communications, and broadcasting networks. This paper presents an overview of CRL's new generation mobile communication system that is called The Multimedia Integrated Network by Radio Access Innovation Plus (MIRAI₊), as well as details the role of Software Radio Technology (SDR) in MIRAI₊.

A framework of dynamically adaptive hardware mechanism on multicontext reconfigurable devices is proposed, and as an example, an adaptive switching fabric is implemented on NEC's novel reconfigurable device DRP (Dynamically Reconfigurable Processor). In this switch, contexts for the full crossbar and alternative hadware modules, which provide larger bandwidth but can treat only a limited pattern of packet inputs, are prepared. Using the quick context switching functionality, a context for the full crossbar is replaced by alternative contexts according to the packet inputs pattern. If the context corresponding to requested alternative hadware modules is not inside the chip, it is loaded from outside chip to currently unused context memory, then replaced with the full size crossbar. If the traffic includes a lot of packets for specific destinations, a set of contexts frequently used in the traffic is gathered inside the chip like a working set stored in a cache. 4 4 mesh network connected with the proposed adaptive switches is simulated, and it appears that the latency between nodes is improved three times when the traffic between neighboring four nodes is dominant.

Interest in the regulatory issues for Software Defined Radio (SDR) is spreading worldwide since the Federal Communications Commission (FCC) recently recognized SDR and created a new category for SDR authorization. SDR technology will bring enormous benefits to the field of wireless services. However, in order to ensure such benefits, revisions of the radio law and/or related ordinances are required regardless of standardization of the software downloading and other implementation details. In order to define the issues peculiar to SDR and to investigate how conformity evaluation should be conducted for radio equipments whose RF characteristics can be altered by software changes in the field, "Study Group on Software Technology for Radio Equipment" was organized by the Telecom Engineering Center (TELEC) in 2000. This paper summarizes a report of the Study Group that was published in March 2003 including the proposal for "Technical regulation conformity evaluation system," the principal output of the study, which proposes how to prevent unauthorized changes to radio equipment in the field.

This paper presents the summarized achievements of "Study Group on Software Technology for Radio Equipment" held at TELEC from April 2000 to March 2003. The Study Group specified the essential issues on Software Defined Radio (SDR), and discussed desirable methods to evaluate conformity to technical regulations in radios that can change RF characteristics only by changing software. The biggest objective in SDR is to build the architecture to allow users to install software exclusively in the combination of hardware and software that have passed the certification test. The Study Group has reached a solution by introducing the idea of "tally." This paper explains the concept of tally, and proposes two types of systems to use tallies in checking adaptability in combinations of hardware and software.

In relation to the Software Defined Radio (SDR) concept, an experimental simulation system was developed. Likewise, verification tests were performed in order to validate the envisaged SDR certification processes including its development, certification, distribution, and software installation assuming the future possibility of exchanging the software in the field.

A new spectrum management architecture for a flexible software defined radio (SDR) is proposed. In this architecture, the SDR hardware and software are certified separately so as not to destroy the SDR flexibility, but to ensure that any combinations of hardware and software are compliant to the radio regulations even at the system (vertical) handover, global (horizontal) handover, and upgrade (forward) or downgrade (backward) handover. This architecture is based on automatic calibration & certification unit (ACU), built-in GPS receiver, and radio security module (RSM). The ACU is a hardware embedded RF manager that dynamically controls the output power spectrum to be compliant to the local radio regulation parameters. This local radio regulation parameters are securely downloaded to the hardware as an electronic label of the SDR software and stored in the RSM which is a security manager of the hardware. The GPS position check is used, especially during roaming, to keep the compliancy of the terminal to each local radio regulations managed by the geographical region. The principle parties involved in this architecture are telecommunication certification body (TCB), SDR hardware maker (HW maker), SDR software maker (SW maker), and SDR user. The roles and relationships of these four parties in the proposed architecture are clarified in this paper.

We have proposed two types of software download methods for software radio (SR) based intelligent transport systems (ITS): (1) broadcasting-type software download method and (2) communication-type software download method. In this paper, we study their feasibility of their employment in a newly developed prototype. We give tangible examples of method (1) using the vehicle information and communication system (VICS) and method (2) using the dedicated short range communication (DSRC) system. We describe the download formats and procedures for both methods and use the experimental prototype to evaluate the basic software download time and configuration time. Moreover we also propose architecture of SR-based multimode terminal that can reduce download time and utilize over-the-air software download services by VICS and DSRC links.

Many carriers are introducing multi-media services to satisfy customer demands for these services. In order to provide such services, carrier must increase their system capacity. It is well known that space division multiple access (SDMA) improves system capacity and is compatible with existing access systems. In order to evaluate the performance of SDMA, we developed an SDMA test bed. The test bed maintains the personal handy phone systems (PHS). The PHS adopts time division multiple access (TDMA). Aiming to compare the performance of SDMA and TDMA using the same analog hardware, the SDMA test bed employs a software-defined radio (SDR) technique. This paper shows the outline and performance of the test bed. The results of laboratory tests indicate that the bit error rate (BER) of the test bed operated in the SDMA mode at under 10^-3 when the carrier-tointerference ratio (CIR) was larger than approximately -22 dB. Antenna patterns measured in an anechoic chamber show that the SDMA test bed produces correct antenna patterns when there are three desired signals and one interference signal. The results of the four field tests confirm that the test bed operated while two-multiplex SDMA mode doubled of the traffic and decreased the interference level as compared with the TDMA mode. Furthermore, the test bed operated while threemultiplex SDMA mode improves the traffic about 2.4 to 2.7 times. The SDMA test bed decreased the impact of the adjusted TDMA base station (BS). Therefore, we confirmed that the SDMA improves system capacity without any degradation.

A novel reconfigurable architecture has been proposed for a mobile terminal receiver that can drastically reduce power dissipation dependant on adjacent channel interference. The proposed design can automatically scale the number of filter coefficients and word length respectively by monitoring the in-band and out-of-band powers. The new architecture performance was evaluated in a simulation UTRA-TDD environment because of the large near far problem caused by adjacent channel interference from adjacent mobiles and base stations. The UTRA-TDD downlink mode was examined statistically and results show that the reconfigurable architectures can save an average of up to 75% power dissipation respectively when compared to a fixed filter length of 57 and word length of 16 bits. This power saving only applies to the filter and ADC, not the whole receiver. This will prolong talk and standby time in a mobile terminal. The average number of taps and bits were calculated to be 14.98 and 10 respectively, for an outage of 97%.

This paper proposes a new algorithm to control an adaptive duplexer for multiband software radio. It uses a wideband low isolation device combined with a two-tap/two-loop adjustable canceller to eliminate the need for multiple switched high isolation duplexers. The taps are adjusted to provide isolation peaks in the transmit and receive bands. The algorithm is based on the superposition of squared errors and achieved 66 dB isolation of the transmit signal and a 37 dB cancellation of the transmitter noise in the receiver band.

Software Radio has been proposed in the 1990s as the solution to flexible transceiver design for future wireless systems. Potential advantages and drawbacks of this approach have been described and analysed in verbose format in many articles. However, a mathematical perspective of the software radio design problem is to be found in the literature only once. Despite this attempt to develop a sound formal description the conclusions do not reach beyond algorithm design. Open issues in system design are often mentioned, but remain unresolved hitherto. We develop a novel mathematical perspective of software radio, and we formulate the design problem accordingly, by means of an integer linear programming (ILP) representation. This type of problem is well-known in computer science and operations research, but it has never been linked to software radio design before. In a first approach to solve the ILP problem we reduce it to a scheduling problem with processor constraints. In the remainder of the theoretical section we introduce the notions of granularity G and speedup s to assess the quality of modular implementations. A random runtime argument leads the way to a system-theoretic approach to modular design issues such as maximizing speedup over a great number of different implementations. For the special case G = 1 we deduce the speedup potential of a primitive graph in analytical form. In the experimental section we compare simulation results to our theory, and we extend the experiments to a more complicated graph which stems from a real software radio design project. The paper concludes with a discussion and a brief outlook to future research issues.

This paper presents a transmit diversity scheme that uses space-time block codes (STBC) in space-spreading code dimensions for time-direction spreading or two-dimensional spreading orthogonal frequency-division multiplexing code-division multiplexing (OFDM-CDM) downlink transmission. The STBC output symbols in two adjacent time slots are spread by two distinctive spreading codes and multiplexed in the same spreading segment. At a receiver, the received subcarrier signals are despread with the two spreading codes in the direction of time, space-time decoded, and then combined in the direction of frequency. Simulation results demonstrated that the proposed scheme provided high tolerance to Doppler spread and outperformed space-time transmit diversity (STTD) for high-mobility users.

Space-Time Turbo code is an effective method for the enhancement of link capacity and maximizing the link-budget by balancing the coding gain obtained via Turbo codes and the diversity gain obtained through multiple antenna transmission. A study on an antenna selection scheme for Space-Time Turbo code for OFDM systems is presented in this paper. In the proposed method, the systematic bits and the punctured parity bits are sent from the selected antenna for each sub-carrier, while data transmission is suspended from the antenna experiencing poor channel conditions at the receiver. Simulation results show that the proposed method yields a 2.2 dB gain in the required TxE_b/N₀ relative to the conventional method, and makes the channel estimation accuracy more robust. Moreover, the proposed method reduces transmission power by about 4 dB compared to the conventional method.

In this paper, as an important technology for the software-defined radio, a novel scheme of adaptive array antenna utilizing bandpass sampling technique is proposed. For adaptive signal processing, it is necessary to convert the radio frequency signal received by the antenna that is given by real number into baseband region, i.e., complex number region. Then, the method for dividing the bandpass sampled signal to in-phase and quadrature components is analyzed. The sampling scheme is called the IQ-division bandpass sampling. An adaptive array antenna based on the IQ-division bandpass sampling is characterized by the signal processing at the bandpass sampled signal stage, namely, intermediate frequency stage, not baseband. Finally, we will confirm the validity of the proposed scheme through an experiment in a radio anechoic chamber.

In radio surveillance systems we consider the problem of identifying interferers or illegal radios in licensed communication channels. The systems considered are receivers using arrays of antennas for spatial processing. At the output of each antenna, we have a mixture of communication signals. The mixture will depend on the distance of the source radios and the propagation environment. These signals may or may not have the same modulation type. The main four tasks in the radio surveillance system are: Separation of the source signals contained in the data mixture at the array antenna, modulation recognition to identify the illegal radio, direction of arrival estimation to pinpoint the location of the illegal radios, and demodulation to intercept the information contained within the illegal transmission. In this paper we deal with the application of the Fast ICA algorithm to a uniform linear array. Our interest is to separate the independent source signals from the mixture of signals obtained at the sensors. Since the target system operates in the HF domain, where analog modulations dominate, the impinging signals are assumed analog modulated communication signals.

The software-defined radio technique translates the traditional hardware radio platform to a flexible software radio platform that can support multiple air interface standards. This work proposes an efficient IF processing architecture based on software-defined radio for 2G GSM/IS-95 and 3G W-CDMA systems. Hardware complexity is estimated by fixed-point simulation. IF processing architecture should be highly flexible and minimally complex. Firstly, a carrier channel is selected from a wide frequency band using a high-resolution numerically controlled oscillator (NCO). Wide-range interpolation/decimation is performed by the cascaded integrator comb (CIC) filter that involves no multiplier nor stores filter coefficients. Both the desired narrowband and the desired wideband signals can be extracted. The look-up table (LUT), based on the distributed arithmetic (DA) algorithm is used to implement the finite impulse response (FIR) filter. Therefore, a small area and high speed can be achieved. The errors caused by truncation, quantization, rounding-off and overflow are predicted using a fixed-point simulation. These predictions will help to evaluate the word-length for VLSI implementation. Finally, ALTERA APEX20KE is used as a target device. One hundred thousand gates are used for the implementation. Thus, the proposed architecture has high processing flexibility and small area.

This paper proposes complex form BandPass Sampling (BPS) that is suitable for the software radio. This BPS utilizes offset frequency sampling and quadrature component interpolation. Three types of BPS techniques are first reviewed, which shows effectiveness of the proposed BPS technique. The major advantages over the conventional BPS techniques are: i) free from the DC offset that is caused by the leak of the sampling clock harmonics into the received signal, and ii) reduction of alias by the complex number processing in the signal detection. Next, detailed description of the BPS operation shows that it requires real-time interpolation for the time alignment of the sampled quadrature component. Finally, computer simulation shows that the misalignment generates distortion, and that effective interpolation techniques can reduce the distortion level less than -60 dB even for wideband signals.

A flexible symbol-timing synchronization met-hod is a one that uses a common sampling clock to find synchronization points for radio communication systems that have different symbol rates. This method estimates synchronization points from state patterns calculated using the symbol rate, sampling clock, and number of observed symbols. Decreasing the number of state patterns is one of best ways to reduce the amount of device resources needed to store the patterns. In this paper, we propose a new pattern generation method in which the number of generated patterns does not increase when the sampling clocks of the communications systems are different. To show the feasibility of this method for symbol-timing synchronization, we analyzed a relationship between the number of samples and the number of state patterns and calculated the BER (bit error rate) in AWGN (additive white Gaussian noise) and one-path flat Rayleigh fading environments by computer simulation.

We consider the routing and wavelength assignment (RWA) problem for large-scale WDM optical networks where each transmission request is served by an all-optical lightpath without wavelength conversion. Two heuristic RWA algorithms are proposed in order to minimize the number of wavelengths required for a given set of connection requests. The proposed algorithms are evaluated and compared with the existing algorithms for two realistic networks constructed based on the locations of major cities in Ibaraki Prefecture and those in Kanto District in Japan.

Recently, wireless networks have become a major sector in the telecommunication industry. More and more applications seek to become wireless. However, a major obstacle in adapting wired applications to wireless is the quality of service problem. Although the wireless bandwidth is improving at a fast pace, it still is not enough for modern multimedia applications. Even if we solve the bandwidth problem, the mobility of users also poses challenges for QoS provision. If the user moves randomly, how and where can resources be reserved in advance for roaming users to move smoothly and seamlessly? In this paper, we propose a method for predictive resource reservation in wireless networks. Resources reserved but not used will seriously affect the system performance. Therefore, we also have mechanisms to release the reserved resources when it is not used within a time limit and allow resources to be used temporarily by another mobile user. We compare the performance of our method with those of fixed allocation scheme and shadow cluster scheme. The results indicate its effectiveness and feasibility.

This paper presents the approximate error rates of M-ary phase shift keying (MPSK) for optimum combining (OC) with multiple interferers in a flat Rayleigh fading channel. The approximations, which have been used to evaluate the performance of binary PSK for OC, are extended to the performance analysis of MPSK for OC in the presence of arbitrary numbers of antennas and interferers. The mean eigenvalues of interference-plus-noise covariance matrix are analyzed to compare the approximation techniques, i.e., first-order approximation and the orthongal approximation. Using the moment generating function (MGF)-based method, the approximate error rates of MPSK for OC are derived as the closed-form expressions in terms of the exact error rates of MPSK for MRC. The approximate analytical results show the simple and accurate way to assess the average symbol error rate of MPSK for OC with arbitrary numbers of antennas and interferers.

For the channel estimation in the pilot channel aided CDMA systems which can support a multi-code scheme, we consider a linear prediction using both pilot and traffic channels. After deriving a new form of the optimal Wiener filter which requires less computational load, for its practical implementation, we propose the decision-directed adaptive linear prediction filter (DD-ALPF). To prevent from falling into the false lock, the proposed DD-ALPF uses the conventional channel estimate obtained only from pilot channel as a baseline for checking the reliability of the filter output. It will be shown through computer simulation that the proposed method can improve the receiver performance and performs better in the fast fading environments, compared with the existing ones.

In ad hoc wireless networks, wireless terminals can autonomously construct and can maintain the network. They communicate with some neighbor terminals, exchange network information and determine routes for packets on the multi-hop wireless network. Flexible Radio Network (FRN), one of the ad hoc wireless network systems, adopts a proprietary protocol that provides a multiple routes management and a packet retransmission mechanism against packet transmission errors. This system is a commercial product that has been in use in a recent few years. In this paper, we first evaluate the performance through simulations for data-link protocol and routing protocol of the FRN to clarify its basic properties. Furthermore, we propose some techniques that enhance its performance and solve problems on the protocols. We show how they improve the system performance through simulations and analyses.

We propose a new protocol, Clustered Group Multicast (CGM), for multicasting in ad-hoc mobile networks. In CGM, there is a set of forwarding nodes (called multicast backbone) which are responsible for forwarding multicast datagrams. Unlike the multicasting protocols in wired networks (e.g., Internet) which construct and maintain a shortest path tree for every multicast {source, group} pair, CGM is a mesh-based multicasting protocol in which the connectivity among the nodes in the backbone is of no longer importance. Thus, there is no tree maintenance overhead, but there are more connectivity than trees and yet it can prevent long-term or permanent routing loops from occurring. A key feature of CGM is the use of the advertising agent to reduce advertising traffic to the system. An advertising agent acts as both a server and a client for the purpose of advertising join requests on behalf of its local clients. Because in CGM multicast traffic is only allowed to be delivered over the backbone, CGM restricts the amount of hosts participating in the backbone to decrease the impact of multicast traffic to the system. From the simulation results, the multicast group management traffic and multicast datagram traffic are much less than the other protocols. This is particularly important for wireless networks which lacks bandwidth.

Vertical stacking is a novel technique for building switching networks, and packing multiple compatible connections together is an effective strategy to reduce network hardware cost. In this paper, we study the crosstalk-free permutation capability of an optical switching network built on the vertical stacking of optical banyan networks to which packing strategy is applied. We first look into the nonblocking condition of this optical switching network. We then study the crosstalk-free permutation in this network by decomposing a permutation evenly into multiple crosstalk-free partial permutations (CFPPs) and realizing each CFPP in a stacked plane of the network such that a crosstalk-free permutation can be performed in a single pass. We present a rigorous proof of CFPP decomposability of a permutation and also a complete algorithm for CFPP decomposition. The possibility of a tradeoff between the number of passes and the number of planes required for realizing a crosstalk-free permutation in this network is also explored in this paper.

Because most link-state routing protocols, such as OSPF and IS-IS, calculate routes using the Dijkstra algorithm, which poses scalability problems, implementors often introduce an artificial delay to reduce the number of route calculations. Although this delay directly affects IP packet forwarding, it can be acceptable when the network topology does not change often. However, when the topology of a network changes frequently, this delay can lead to a complete loss of IP reachability for the affected network prefixes during the unstable period. In this paper, we propose the Cached Shortest-path Tree (CST) approach, which speeds up intra-domain routing convergence without extra execution of the Dijkstra algorithm, even if the routing for a network is quite unstable. The basic idea of CST is to cache shortest-path trees (SPTs) of network topologies that appear frequently, and use these SPTs to instantly generate a routing table when the topology after a change matches one in the caches. CST depends on a characteristic that we found from an investigation of routing instability conducted on the WIDE Internet in Japan. That is, under unstable routing conditions, both frequently changing Link State Advertisements (LSAs) and their instances tend to be limited. At the end of this paper, we show CST's effectiveness by a trace-driven simulation.

This study examines a slitted parallel plate waveguide (PPW) from the perspective of diffraction and equivalent circuit representation for a narrow slit and radiation, including the surface wave effect, from a wide slit. The fundamental differences between the diffraction and equivalent admittance properties of the slit discontinuities in typical microstrip and waveguide structures are considered by comparing how the waveguide heights of the PPW and dielectric constants filling the inside of the PPW correspond to those of the two structures, respectively.

By adding an auxiliary transformer to a single-stage single-switch power-factor-corrected converter (S⁴PFCC), the storage capacitor voltage and its range of voltage change against line voltage change are reduced. In addition, this transformer provides a direct power transfer path for input line to output load to increase the conversion efficiency. High power factor is maintained due to the elimination of dead angle of the input current. This paper presents detailed analysis and optimal design of a discontinuous conduction mode (DCM) boost-flyback S⁴PFCC with the auxiliary transformer. Experimental results for a 15 V/60 W prototype and with comparison to a S⁴PFCC without the auxiliary transformer are given to show the proposed approach effective.

The effects of hybrid concatenated space-time (HC-ST) codes applying iterative a posteriori probability (APP) decoding are investigated. The bit error rate (BER) and frame error rate (FER) performance of the iterative decoded hybrid Tarokh, Seshadri, Calderbank space-time (TSC ST) coded system under flat Rayleigh fading is analyzed. At the FER 10^-2 level the results show that the serially concatenated space-time (SC-ST) codes provide a coding gain of 3 dB compared to the TSC ST codes, where an additional coding gain of 1 dB beyond the SC-ST code performance can be obtained applying the HC-ST coding topology.

A new method for computing precise estimates of the motion vectors of moving objects in a sequence of images is proposed. The proposed method is based on dynamic programming matching applied along chain-coded binary contours of images. This significantly reduces the computational complexity of the correspondence matching applied to the 2-D optimization problem. Computer simulation and experimental results demonstrate a good performance of the method in terms of dynamic motion analysis.

We propose a mathematical model to analyze the performance of TD-CDMA/TDD systems in terms of call blocking probability and then find the optimum time-slot switching-point at the smallest call blocking probability considering asymmetrical traffic load distribution for various kinds of service applications.

The IEEE 802.11 WLAN standards adopt CSMA/CA protocol with a backoff algorithm as medium access control technique. When the number of stations which attempt to access a network increases, the throughput efficiency of the standard goes down. In this paper, we propose a modified backoff algorithm which adaptively selects the Contention Window (CW) size according to the variation of the number of contending stations and present the results of simulation analysis.

In this letter, a concise formula for the SNR degradation of OFDM caused by carrier frequency offset is derived by approximations over a shadowed two-path channel, which explicitly shows the sensitivity of SNR degradation to various parameters including the frequency offset. It is shown that, for small frequency offset, the SNR degradation is proportional to the square of the frequency offset and the square of the number of subcarriers. It is also shown that, if E_s/N₀ is reasonably large, the SNR degradation becomes insensitive to E_s/N₀, which is contrary to the case of the AWGN channel.

This letter examines a very simple iterative chip-by-chip multiuser detection strategy for spread spectrum communication systems. An interleaving-based multiple-access transmission technique is employed to facilitate detection. The proposed scheme can achieve near single-user performance in situations with very large numbers of users while maintaining very low receiver complexity.

In this letter, iterative sequence estimation technique based on expectation-maximization (EM) algorithm is considered for quadrature amplitude modulation (QAM)-orthogonal frequency division multiplexing (OFDM) signals. For QAM-OFDM signaling, the optimal EM algorithm requires high computational complexity due to the inversion of complex matrix executed at each iteration. To avoid this problem, we propose a sub-optimal iterative sequence estimation algorithm with some approximations, which results in reduced computational complexity for QAM-OFDM signals. Moreover, we use two different approaches to obtain initial estimate for beginning iteration of proposed algorithm. One is for less time-dispersive but fast fading channel and the other is for highly time-dispersive but relatively slow fading channel. The bit error rate (BER) performances of the proposed algorithm are evaluated using computer simulations. The results show that the proposed algorithm performs nearly as well as the optimal EM algorithm.

Two important lemmas on the determinant of random matrixes are deduced in this paper. Then based on these results, expression for the mean capacity of MIMO system over Rayleigh fading channels is obtained. This expression requires little calculation and is simple and efficient compared with conventional methods, and furthermore, it gives an explicit relation on the mean capacity of MIMO systems with antenna numbers and the relation of mean capacity with signal to noise ratio (SNR). Accuracy of this theoretic formula has been verified by computer simulation.

Among the many scheduling algorithms which can guarantee delay bounds of sessions, the EDF scheduler with regulators has received wide attention since it can admit a very high number of sessions. However, due to regulators, EDF with regulators has difficulty in scalable implementation. To implement a regulator, a buffer and a timer are needed to restrict the incoming traffics. Given N number of sessions, N regulators are required. Moreover, due to regulators, the entire scheduling algorithm is not work-conserving. To enforce work-conserveness, it is known in the literature that additional buffers and a complex mechanism are required to bypass regulators. Thus, scalable implementation becomes much more difficult in the case of the work-conserving EDF with regulators. In this paper, however, we show that the buffers and timers used to implement regulators are unnecessary to guarantee delay bounds of admitted sessions in fixed-sized packet networks. Then, we can remove those unnecessary buffers and timers. By the removal, the resulting scheduling algorithm can be implemented in a scalable way and becomes work-conserving for free.

The current congestion control mechanism for TCP with ECN suffers from the oscillation of the window size and the unfair sharing of network resources. These problems are caused by the insufficient information about the congestion level of a network and the different round trip time (RTT) among the connections respectively. In this paper, we propose a new window control algorithm for TCP with ECN to avoid the oscillation of the window size and to achieve the fairness among the connections.

The aim of this paper is to raise utilization of resource and handover success rate at handover time. The proposed method takes advantage of the moving history of a mobile host at connection admission control time. We demonstrate the benefit of the proposed method over previously proposed reservation schemes by simulation.

In a content-based image retrieval (CBIR) system, both the retrieval relevance and the response time are very important. This letter presents the condensed two-stage search method as a new fast image retrieval approach by making use of the property of Cauchy-Schwarz inequality. The method successfully reduces the overall processing time for similarity computation, while maintaining the same retrieval relevance as the conventional exhaustive search method. By the extensive computer simulations, we observe that the condensed two-stage search method is more effective as the number of images and dimensions of the feature space increase.

Chain codes were developed for storing contour information for shape matching. The traditional chain codes are highly susceptible to small perturbations in the contours of the objects. Therefore, traditional chain codes could not be used for image retrieval based on the shape boundaries from the large databases. In this paper, a histogram based chain codes are proposed which could be used for image retrieval. The proposed chain codes are invariant to translation, rotation and scaling transformations, and have high immunity to noise and small perturbations.