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The purpose of this paper is to speed up handover processing. The handover process is first analyzed. Based on the result of analysis, an extension of the protocol is proposed, in which adjustment of the timer value, a mechanism to report the link layer information to the network layer, and a speed-up of DAD processing are considered. The extended protocol is implemented. A problem in the operating system is identified by experiments, and solutions are presented. This paper also presents the tool constructed for the handover experiment, which produces a network shift at the desired time and timing. The handover time was measured and evaluated using the tool in an experimental environment, and it was shown that the handover time is drastically reduced compared to the system based on the standard IPv6 specification. The effect of the handover time on the application was evaluated, and it was demonstrated that the proposed method can realize handover with little effect on the application, although the environment is prepared by emulating wireless communications by cable communications. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 88(11): 44–53, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.20213 [ABSTRACT FROM AUTHOR]

ABT (ATM block transfer) has been considered an efficient protocol for high-burst data transmission in ATM networks. The characteristics of ABT in the ATM service class have already been investigated. For practical applications, however, it is necessary to consider an upper layer having its own congestion-control mechanism, apart from the ABT. This paper analyzes the characteristics of a TCP (transmission-control protocol) for the Internet which has become very popular recently, when this is used for the upper layer of the ABT. It is shown to be possible to separate the congestion control and the error-recovery service, since the congestion control is carried out effectively in the ABT by the retransmission mechanism of TCP, without using the congestion control of TCP. However, for high performance it is necessary to tune the retransmission timer of the TCP as discussed in this paper. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 86(1): 97–104, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.1150 [ABSTRACT FROM AUTHOR]

This paper discusses a cell loss recovery method using FEC. The technique applies VP-based FEC and recovers discarded cells that were lost because of buffer overflow at the ATM nodes in the transport network. This paper presents the FEC code format and analyzes the performance, with emphasis on the design of coding matrix and FEC code. Next, an improved structure is presented in which FEC is applied only to the paths with higher quality requirements. This improves loss recovery performance at higher traffic load and reduces coding matrix size. The analyses also include the aggregate processing delay in the cell multiplexer and FEC codec. © 2000 Scripta Technica, Electron Comm Jpn Pt 1, 84(3): 77–88, 2001 [ABSTRACT FROM AUTHOR]

Recently a few studies have been made of distributed coupled-line type digital frequency multipliers which multiply the repetition frequency of a square input wave. The transfer functions of an n-section distributed coupled-line constant-resistance network and an n-section distributed-constant bridged-T network have the form of a rational function of order n with respect to delay operator Z-1. We develop our discussion on the basis of this transfer function. The theory is developed for the coefficients of the rational function and the output response in a stationary state can be determined by the coefficients of both the denominator and numerator. The obtained results can be applied to all networks having the transfer functions treated in this paper, but the detailed realization condition for each individual network is not discussed. The experiments and calculations shown in Figs. 4 - 7 are conducted using the distributed coupled-line constant-resistance network as an application example. The network described here can be a digital frequency multiplier for two incident impulse trains, 1111 and 1-11-1. The behaviors of the multiplier in the frequency domain as well as in the time domain are investigated and it is shown that a faster impulse train can be obtained at the output in the time domain by adequately thinning out the discrete frequency components of the incident impulse train. [ABSTRACT FROM AUTHOR]

This paper discusses the channel assignment problem in the multichannel, high-speed packet network. It is assumed that multiple channels exist between a pair of switches in the network. It is expected that by regarding those channels as a single channel group and executing the packet transmission, a more efficient high-speed and large-capacity transmission will be realized. This paper newly introduces, in contrast to the virtual circuit in the multichannel network, the concept of a channel group-virtual circuit (CG-VC) for the multichannel network, where the packets are assigned to the channels based on the concept of the overflow traffic. To realize the CG-VC system in the high-speed environment, a simple and practical switch structure is proposed. The switch uses optical fibering as the basic module and the system is constructed by superposing the basic modules. Each packet automatically selects an appropriate channel from the set of output channels belonging to the same channel group. It is shown as a result of performance evaluation that, compared to the conventional VC packet switching system, the proposed system greatly improves the transmission delay and utilizes efficiently the channel. [ABSTRACT FROM AUTHOR]

Transmission channels in mobile communication are extremely unstable due to the effect of multipath fading. Therefore the realization of high-reliability communication necessitates the introduction of error control in the form of error-correction coding. This paper considers the decoding of error-correcting codes that are adapted to digital mobile communication operating under Rayleigh fading conditions. A soft decision Viterbi decoding using the received carrier level is studied for the case of convolutional codes. This soft decision method is based on the level of the received carrier by adaptively determining the metric used in the decoding. In this way a metric is obtained for digital mobile communication. This paper also presents an estimation method using error exponent which provides effective means to realize such a soft decision method. If the channel can be considered memory less due for example to interleaving, then this method provides an easy way to estimate: (1) the figure of merit for quantizing thresholds, and (2) the gains of the soft decision. First, it is shown that using this kind of estimation in the case of channels which can be considered sufficiently memoryless due to interleaving, extremely large gain is obtained by the soft decision method discussed in this paper. Next, simulations are performed and the efficiency of the estimation method is demonstrated. Finally, the bit error rate characteristics are shown for the case when the channel is not made sufficiently memoryless by interleaving. [ABSTRACT FROM AUTHOR]

The demand for video service in mobile communications systems is increasing. In response to this demand, wireless P-MP (Point-to-MultiPoint) video transmission, in which a given video is transmitted to several moving terminals via a single channel, has been studied. This scheme can make effective use of bandwidth, power, and spreading codes. In this paper, as wireless P-MP video transmission schemes effective in the CDMA system, we propose MCAP (MultiCode Assignment algorithm for P-MP transmission) and the P-MP video transmission system. In MCAP, the mobile terminals receiving the identical video are divided into two groups, high quality and low quality, based on the channel quality. By performing transmission power and transmission rate control for each group, the transmission rate can be improved. Also, by using multicode transmission, transmission rate control and the channel allocation are carried out in a unified fashion. In this paper, by means of simulations, it is demonstrated that the proposed scheme can substantially improve the transmission rate without increasing the number of spreading codes and the power. Several schemes are compared as power control base selection schemes needed for MCAP. An optimum scheme is determined from the comparative evaluation results. © 2006 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 89(8): 14–25, 2006; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.20280 [ABSTRACT FROM AUTHOR]

This paper discusses methods of sending messages from a source that created a mobile agent (MA) to the MA, which has moved. These methods include the Home-Proxy method and Broadcast method. However, with these techniques, it is possible that the transmitted messages may not be delivered to an MA that is frequently moving around on the network. A method for resolving this problem has been proposed by Murphy and Picco (hereafter termed the Murphy method). Although the Murphy method can reliably deliver messages to the MA, it does not take into consideration the time that elapses until the messages are delivered to the MA. The authors propose a method for delivering messages in the shortest possible time (hereafter termed the clustering method) by partitioning the entire network into multiple areas (clusters) to improve this Murphy method. This paper, proposes the clustering method precisely and in detail. In addition, the validity of this method is certified by performing a logical verification. Simulation experiments are performed to compare the method and Murphy method to prove the effectiveness of the method by verifying that the time required to deliver messages to the MA decreases if the time that an MA stays at a node is not short and that the number of messages decreases if the time that an MA stays at a node is short. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 89(1): 67–80, 2006; Published online in Wiley InterScience (www.interscience.wiley. com). DOI 10.1002/ecja.20184 [ABSTRACT FROM AUTHOR]

In multicast delivery in a network environment with various bandwidth limitations, it is effective to apply the layered multicast technique, which realizes delivery at a reception rate matched to each receiver, although the rate is set discretely. This paper assumes a multimedia session such as a multichannel broadcast or a video conference, composed of multiple streams. In such an application, it is desirable from the viewpoint of efficient use of bandwidth that not only should the streams be transmitted with stable quality, but that the rate should be assigned according to requests, such as attention to the speaker. However, it is difficult to achieve the above goal by directly applying the conventional layered multicast rate control technique, since each stream controls the rate independently. From such a viewpoint, this paper presents a session level mechanism covering multiple layered streams. A multiple streams controller (MSC) is introduced into each receiver, and each MSC autonomously controls the reception rate with interstream coordination in the session. The MSC monitors the reception state of the multiple streams simultaneously and estimates the bandwidth information on the bottleneck link and the set of transiting streams, based on the detected congestion data. Based on the estimation, the learning efficiency of the reception rate is improved and rate assignment control over the streams is applied. An evaluation is performed by computer simulation, and it is shown that the reception quality of the stream is improved by applying the proposed method, and that the reception rate can be coordinated among the streams. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 88(8): 18–31, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.20142 [ABSTRACT FROM AUTHOR]

This paper considers the selective-repeat ARQ scheme, which is a typical error control scheme used in data communication systems, and analyzes the performance of the scheme in which the number of frame retransmissions is limited. In the proposed scheme, messages such as image for which information loss can be tolerated to some extent are transmitted after being divided into multiple frames. If the retransmission of a frame reaches the preset limit, the receiver accepts that frame even if an error is present, and the error is processed as noise or distortion. By this scheme, an upper limit can be defined for the time from the start to the end of the message transmission. This paper analyzes the proposed scheme using the discrete-time queuing model and derives the distribution of the message transmission delay. The throughput and the required receiver buffer capacity are also derived. It is shown by numerical examples that the throughput can be improved and the receiver buffer capacity can be reduced. If the strict analysis is continued, however, the state space of the Markov chain is enlarged with the increase in the number of parameters such as the round-trip propagation delay and the number of message divisions. Thus, the numerical calculation of the transmission delay becomes difficult. The approximate analysis of the message transmission delay and its properties are discussed in Ref. 1. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 87(6): 1–11, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10181 [ABSTRACT FROM AUTHOR]

In this paper, we propose a multichannel Medium Access Control (MAC) protocol based on the Point Coordination Function (PCF) stipulated in the IEEE 802.11 standard in order to guarantee the end-to-end bandwidth in a multihop wireless network. In the proposed protocol, each wireless terminal is equipped with multiple wireless interfaces and assigns a wireless channel to each wireless interface. Each wireless interface autonomously determines whether to operate in the master mode as the parent terminal, in the slave mode as a child terminal, or in the silent mode as a transmission disabled terminal, and builds a parent-child relationship with each wireless link. Since this enables the use of the PCF at each wireless link and guarantees the bandwidth, the end-to-end bandwidth can be guaranteed in a multihop wireless network. We evaluate the performance of the proposed protocol in computer simulations and demonstrate improved throughput, delay, and packet loss rate compared to the conventional Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA) protocol. We also demonstrate that the bandwidths for bandwidth-guaranteed packets are guaranteed even when traffic mixing conventional bandwidth-guaranteed packets and best-effort packets are present on the network. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 87(4): 16–25, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10139 [ABSTRACT FROM AUTHOR]

Recently, the spread of multicast communication is expected from the perspective of the efficient utilization of the network. Since the connection state of multicast communication clearly differs from that of conventional unicast communication, the optimum network topology is believed to change if multicast communication spreads in the future. The authors studied the network topology used when unicast communication and multicast communication are mixed. It was found that the Delta-Star topology was optimal compared to the Star topology currently used in the lowest level of the network. In this paper, we studied the performance of each topology from the perspective of the required capacity of the equipment and the costs required to construct the network when a circuit switching or a connection service was assumed and the multicast traffic increased. Furthermore, we proposed a network extension method that changed the topology from the Star to the Delta-Star topology and evaluated the performance. We found that by using a relatively simple extension method to set the locations of the transmission route link extensions, adequate cost performance was obtained. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 86(6): 69–83, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.1166 [ABSTRACT FROM AUTHOR]

Since the fundamental wave of TV broadcasts has very high power, its second harmonic wave also has sufficiently high power to interfere with radio terminal signals close to a broadcast tower. In this paper, we propose an interference canceler for a digital wireless system operated at the same frequency band as that of the harmonic of TV broadcasting wave. The proposed canceler can eliminate such interference by making use of the replica of harmonic wave, which is generated based on the fundamental wave received by another subsystem. Computer simulation has evaluated the performance of the proposed canceler in a receiver with PSK modulation, and it is confirmed that the proposed system is effective in improving the efficiency of frequency use. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 86(2): 14–22, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10038 [ABSTRACT FROM AUTHOR]

Subsurface radar to detect buried objects from the ground surface by using electromagnetic waves has been used widely for detection of buried pipes since a shallow subsurface zone of up to 2 to 3 m can be visualized with high resolution at high speed. On the other hand, deeper subsurface detection is required in searching for archaeological ruins. In this paper, a delay correlation-type subsurface radar using monocycle pulse signals modulated by M-sequence codes is proposed with a view to expanding the search range by increasing the average energy of the transmitted signals. With a test setup, a steel pipe buried 2.5 m below the ground surface was detected with a transmitting power of 1 mW. In regard to the maximum detectable range of the subsurface radar using this type of coded signals, this paper carries out a study from the point of view of both the time side-lobe and S/N. © 2001 Scripta Technica, Electron Comm Jpn Pt 1, 85(2): 8–16, 2002 [ABSTRACT FROM AUTHOR]

Recently, for the improvement of reliability of digital communication systems, various kinds of error-correcting codes are being used. To improve system reliability through the use of more capable error-correcting codes it is extremely important to simplify the decoder construction and to reduce the decoding delay. This paper studies a decoding algorithm for a class of simple iterated codes (SI codes) constructed by simply repeating I times a codeword (called block) in a given code (called subcode). Moreover, this paper discusses the decoding delay when this decoding algorithm is applied to decoding the Reed-Muller codes constructed by the superimposing method. [ABSTRACT FROM AUTHOR]

With the improvement of the speed of computers in recent years, there is required a local area network (LAN) that can realize a high-speed] broadband communication. For this purpose, asynchronous transfer mode (ATM) LANs bassed on ATM technique is considered important. This paper considers the bandwidth reservation control as a means of traffic control in ATM LAN and evaluates the performance by simulation. The evaluation as made for the following methods of congestion control: ACK/NACK method,based on the ACK/NACK signal which is results when the bandwidth reservation succeeds/ fails; the dynamic ACK/NACK method, which is an improvement so that the control can S adapted to the traffic load; and control method based on the link information from the switch. It is shown for the single-stage switching network that the dynamic ACK/NACK methode has a better performance when the traffic is low but the control method based on the link information has the highest performance when the traffic is high. It also is shown for the three-stage switching network that the control method based on the link information has the highest performance. [ABSTRACT FROM AUTHOR]

The dual-frequency antenna with the same beam- width is well suited to the mobile communication base station antenna. One candidate is a corner reflector antenna whose elements are connected to a parallel feed line. However, no design equations or analysis have been published. This paper corrects this deficiency through analysis of the characteristics after integration of the two factors, spatial coupling and transmission line, for this antenna. A two dual-frequency corner reflector antenna is designed. One uses co-phase feedlines and is a wide band sector-beam antenna with a 3 dB beamwidth difference of 300 in the field pattern of horizontal plane and a return loss of less than -15 dB over a wide frequency range (from 900 MHz to 1.5 GHz). The other uses antiphase feed lines and its beamwidths can be varied by changing the corner angle while the beamwidths are coincident at two frequencies. [ABSTRACT FROM AUTHOR]

This paper proposes a flexible parallel optical data transmission system as follows. The data transmission itself is executed in parallel by utilizing the super- parallelism of the Light. In addition, the address information which is necessary for the transmission is superposed on the transmitted data. By those schemes, the capacity of the individual transmission channel as well as the number of transmission channels in the system are made variable. This paper describes first the principle of parallel optical data transmission. It is shown that the symbolic substitution (SS) is useful in the superposition and separation of the address information, which are the basic operations of the system. SS is a parallel and position-invariant Logical operation. An experimental system is constructed by the SS technique using the computer- generated Fourier hologram (CGFH). A result of experiment is shown for the single-channel parallel transmission of two-dimensional data. Thus, it is shown that the flexible transmission system, where the capacity of the individual transmission channel as well as the number of transmission channels are variable, can be realized by the proposed transmission scheme. [ABSTRACT FROM AUTHOR]

In the multichannel frequency division multiplex (FDM) optical transmission, the intermodulation distortion generated by the nonlinearity of the LD is a problem. In this paper, it is analyzed that fin the number of transmission channel, N, and the optical modulation index per channel, m, the CSO and CTB are constant for a different value of N if the effective optical modulation index, m√N, is maintained. The results are compared with experiments. In the present analysis, two types of nonlinearity of the current-to-optical conversion characteristics are considered. One of them is the nonlinearity near the bias current of the LD and is predominant in the region of a small effective optical modulation index. Another is the nonlinearity at the threshold value of the LD and is dominant when the effective optical modulation index is large. It is shown by means of the probability density function that the CSO and CTB degrade suddenly if the effect of the nonlinearity of the threshold value appears. In the FM-FDM optical transmission. many distortions in the transmission bandwidth behave as the white noise due to frequency modulation when the carrier is frequency modulated by a video signal. Hence, the CN ratio is effectively degraded. In this paper, it is further presented that the CN ratio degraded by the distortions can be derived from the CSO and CTB by taking into consideration the frequency allocation of the FM signal. [ABSTRACT FROM AUTHOR]

This paper presents a BCMP-type closed queuing network which models the error recovery defined as HDLC based protocols such as LAPB, LAPD, and LLC type II. Throughput and average end-to-end delay against the I-frame loss probability owing to bit errors are obtained by using the queuing network. With the I-frame length independence assumption, agreement between analytical and simulation results is shown to be fairly good. [ABSTRACT FROM AUTHOR]

Intensive efforts have been observed in recent years on the development of data communication technology, aiming at the effective utilization of such existing transmission tines as the in-house distribution line. The automatic consumption meter reading, for example, uses the low-voltage distribution line as the transmission channel. This paper aims at the improvement of the receiver demodulation performance m the data transmission using the low-voltage distribution line and proposes the phase-diversity reception using two phases (red and black) of the single-phase, three-wire system. In the data transmission using the low-voltage distribution line, stray signals are produced between the two phases of the single-phase, three-wire lines and two signals with different signal-to-noise (SN) ratios are received. If a simple scheme such as the addition is employed, a problem arises in that the reliability of the demodulated data is deteriorated. To cope with this problem, this paper proposes a phase-diversity reception, where time-varying phase difference is adjusted to zero using an adaptive phase shifter and the SN ratio is improved by synthesizing the two-phase received signals. The proposed scheme is implemented on a digital signal processor (DSP), and a robustness against the channel noise and the periodic attenuation is realized by the feedback (FB) control. As a result of an experimental evaluation, it is shown that the SN ratio is improved by 2.7 dB at the maximum for the stationary phase error, compared to the traditional simple addition and the addition/subtraction selection schemes. Thus, it is verified that the receiver demodulation performance of the low-voltage distribution line data transmission can be improved. [ABSTRACT FROM AUTHOR]

Traffic generated by B-ISDN services is difficult to predict because the fluctuations in normal traffic patterns are large. As a result, it is difficult to provide network equipment to avoid traffic unbalance. The advantage of using asynchronous transfer mode (ATM) in BISDN networks is that path bandwidth control facilities can be used to dampen the unbalance in traffic. This paper proposes a new algorithm, based on the hatch modification method, for controlling traffic fluctuations. The batch modification method is used to control path bandwidth from a centralized control center. The virtual path bandwidth is calculated using optimization calculations based on information from the entire net-work. Hence, it is necessary to develop an algorithm to perform the optimization calculations. The algorithm uses the worst case blocking probability and the total amount of traffic. The algorithm is verified with numerical examples that compare traffic patterns in a model net work. [ABSTRACT FROM AUTHOR]

An asynchronous transfer mode (ATM) switching network handles not only traffic of various bandwidths, but multirate bursty traffic including constant bit rate and variable bit rate traffic as well. Consequently, to simplify the design and operation of the facility, it is desirable for the ATM network to be nonblocking at the connection level. The nonblocking issue when the network must handle variable bit rate traffic is related closely to the connection admission control (CAC) method. While the non blocking feature for a linear CAC function has been studied in the past, the relation between the switching network cell-level quality and non-blocking characteristic for a specific CAC premising variable bit rate traffic has not been analyzed. This paper examines a nonlinear CAC, evaluates the cell loss probability and statistical multiplex gain when it is applied and proposes the definition for a non-blocking feature called "(ρ0, p, q) nonblocking." Next, a cell distribution-type, three-stage ATM switching network is shown that ensures (ρ0, p, q) nonblocking for traffic including multicast connections without expanding the internal link capacity either in terms of space or speed. [ABSTRACT FROM AUTHOR]

Intensive research and development efforts have been made on the intelligent network (IN) to realize diversified and advanced telecommunication services and the early introduction of new services under a multi-vendor environment. Earlier research and development, however, assume the conventional telephone service and narrowband ISDN services and gave little consideration to the broadband ISDN. The study of the IN from the viewpoint of the broadband ISDN is urgent considering the systematic realization of the services (i.e., multimedia services) that are not yet sufficient as well as the increase of the demand for the development of new services. This paper proposes the call model, which provides the basis for realizing the IN architecture incorporating the broadband ISDN. First, the basic concepts are defined clearly. Then the call model is proposed which can realize the general connection control for various connection types as well as the flexible control for the multiple virtual channels in the multimedia environment. The model also facilitates the control of the interactions among related services. Finally, the service control characteristics of the proposed call model are investigated. [ABSTRACT FROM AUTHOR]

This paper investigates an adaptive equalizer with career frequency offset compensation for digital mobile radio transmission over several tens kbit/s. The performance of conventional adaptive equalizers for mobile radio severely degrades for receiving signals having carrier frequency offset which is mused by local oscillator frequency offset due to misadjustment, temperature dependence, and aging. Carrier frequency offset compensation for TDMA burst signaling requires both accurate and short-time measurements. However, both requirements have a trade-off relationship and must be optimized. This paper proposes the carrier-acquisition-during-training (CAT) algorithm for transmissions allowing a training signal. First, two basic configurations and their equivalence are discussed. Next, computer simulations and real-time experiments with one-chip DSP show the effectiveness of CAT by demonstrating training convergence, optimum parameter characteristics, and average BER performance under two-path selective fading with one-symbol delay difference and independent Rayleigh components. [ABSTRACT FROM AUTHOR]

Recent construction of corporate communication networks with the use of high-speed digital links, etc., is becoming popular. Economical and reliable networks must be designed. Few papers have been published on the design of corporate communication networks. Moreover, some need improvement. This paper proposes a new topological design algorithm for efficiently constructing economical and reliable corporate communication networks. This algorithm achieves effective network design by utilizing the correct network information. The triangle network configuration, which provides higher reliability than two-connected networks with only a small additional cost, is proposed. Moreover, this paper proposes a new idea at a lower cost (cost lower bound) that evaluates the results obtained by various heuristic algorithms. Superiority of the proposed algorithm is verified by means of this cost lower bound. [ABSTRACT FROM AUTHOR]

The integrated services digital network (ISDN), which is under construction at present, will enlarge the range of communication service by developing radio access systems, with the digital mobile communication systems as the core. The purpose of such a radio access system is to realize the communication capability, which is as flexible as that of the fixed system, while utilizing the features of the radio system. The following problems arise in the traditional access schemes, however, when multiple types of calls with various information characteristics and rates are to be handled. The immediate (circuit switching) system cannot contain efficiently the multiple type of traffic, while the waiting (packet switching) system may produce a delay with a long variation. This paper proposes a new scheme for radio channel access, aiming at a positive integration of the multiple type of traffic. In this scheme, the blank of information is compressed to construct a block (cell) with a constant length, and the physical channel is controlled to be assigned to each cell. The blank is inserted on the receiving side to ensure the time-transparency. Then, the flexibility to various types of immediate and waiting calls, as well as the efficient operation of the channels, are realized. This paper describes also the concept of the scheme and fundamental composition. The channel capacity and the channel occupancy are evaluated by a computer simulation, and it is shown that the proposed scheme has the desirable property that the statistical multiplexing effect is utilized. [ABSTRACT FROM AUTHOR]

Methods for evaluating the cross-polarization power ration (XPR) in mobile communication include the one using a crossed dipole, the one using a dipole antenna and a turnstile antenna, and that using a dipole antenna and a loop antenna. However, measurement errors in these methods have not been investigated. In this paper, the measurement error characteristics of various XPR measurement methods are analyzed theoretically. By finding the factors causing degradation of measurement accuracy, this paper proposes a method with more accuracy and clarifies its measurement error characteristics. it is found that the measurement error depends on the XPR and the measurement accuracy id degraded significantly because the measuring antenna does not have isotropic characteristic and the antenna for horizontal polarization has a vertical polarization pattern. To improve this deficiency, a method is proposed in which a cylindrical slot antenna is used for the horizontal polarization because the sensitivity of this antenna to the vertical polarization can be made sufficiently low. it is pointed out that the measurement error with this antenna can be improved significantly. [ABSTRACT FROM AUTHOR]

This paper discusses a simple go-back-N- ARQ scheme in which the satellite is provided not with the repeater function but also with the processing function. The satellite detects the error in the data block sent from the transmitter. It returns ACK/NACk to the transmitter and relays the correctly received date block to the receiver. The receiver detects the error in the received data block and returns ACK/NACk to the transmitter. The transmitter examines two acknowledgement signals returned from the satellite and the receiver, and transmits the data block based on the basic go-back-N ARQ protocol. This paper analyzes the scheme in which the hardware complexity of the satellite is minimized. It is shown that the throughput can be increased and the average transmission delay can be decreased. [ABSTRACT FROM AUTHOR]

This paper considers the block demodulation method with carrier phase synchronization without preamble, and presents a proposal and evaluation for the new carrier phase synchronization method based on the sequential regression estimator. The black demodulation has a feature in that the traditional way of real-time demodulation is not adhered to, and the preamble is not required. This improves greatly the through-put especially in a short packet communication. The traditional system is based on the least-square scheme, noting that the cumulative carrier phase variation between data symbols is represented in a linear form. It has a problem in that the linearity is lost due to noise in deriving the cumulative value for the phase variation, which may greatly deteriorate the demodulation performance. The sequential regression estimator proposed in this paper is based essentially on the least-square principle. However, the cumulative value is not determined for each time as in the traditional method but the estimation is updated based on the difference between the observed value and the value estimated from the data up to the previous sample. Consequently, the problem in the traditional system essentially is eliminated. Because of this elaboration, the computation in the proposed system is very small compared with traditional systems. It is seen as a result of simulation that the tolerable noise margin for 80 percent synchronizing probability, which has been used as the evaluation criterion, is improved by 11 dB for BPSX. In the satellite packet communications network such as VSAT (very small aperture terminal), the throughput is improved remarkably. Applying the proposed principle to the design of the present link, the time- ratio, where the burst synchronization error is 1 percent or less in the Ku band, is 0.03 percent. In VSAT, a random access scheme is employed. Considering the burst collision probability of the access, it follows that the burst synchronization error probability of 1 percent does not affect the overall performance, and the proposal in this paper can be considered as sufficiently practical. [ABSTRACT FROM AUTHOR]

This paper proposes a new ARQ scheme that can reduce the influence of backward channel errors for bi-directional data transmission systems. The main feature of the proposed scheme is that both present and past acknowledgement signals are utilized for the decision as to whether or not the data signals should be retransmitted. Throughput performance is analyzed in both CBN and SR ARQ schemes. A small number of returned past acknowledgement signals is required to improve the throughput efficiency. For an ideal SR scheme with an infinite buffer, increasing the number of returned past acknowledgment signals makes the throughput efficiency asymptotically close to the upper bound. [ABSTRACT FROM AUTHOR]

One of the factors determining the performance of the digital radio-relay system is the interference from the existing FDM- FM systems. Traditionally, the FDM-FM interference characteristics have been evaluated by analyzing the effect on the error rate by the interference wave which is approximated by a sinusoidal wave. The method tends to underestimate the effect on the error rate in a digital system when the analysis is compared with the case where an interference wave is actually given. This paper proposes a method for evaluation in which the FDM-FM interference wave is divided into the carrier and the sidehand components. Based on the new method, the interference characteristics have been analyzed. It is found from a laboratory experiment that the present evaluation method is valid. [ABSTRACT FROM AUTHOR]

The following is a method to eliminate an interference signal. The interference signal is obtained from the source, and the amplitude and the phase of the interference signal is adjusted to the same amplitude and the opposite phase of the interference component, and is added to the received signal. One important problem is how to control automatically the amplitude and the phase of the interference signal. This paper proposes a vector correlation detection-type interference canceller for the digital system. A orthogonal error signal detector is employed which produces the in-phase and quadrature error signals. Combining the circuit and the given interference signal, the correlation detection is performed between the two to perform the automatic control of the amplitude and the phase of the interfering signal. In the first part of this paper, the operation principle is described theoretically. Then, the configuration is shown. An experiment was carried out where the desired signal is 16 QAM or 256 QAM, and the interference signal is an FM signal (telephone 3600 ch.). the improvement factor was 26 dB for 16 QAM and 21 dB to 140 ms. The result indicated the validity of the theory and the effectiveness of the proposed canceller. [ABSTRACT FROM AUTHOR]

When the window-flow control scheme is applied to the packet communication network, one of the important problems is the determination of the window size. In past studies on the window-size allocation problem, the network was described by a simple queue model, and an approximate analysis is made. This paper proposes an approximate analysis method for a mathematical model which is closer to the actual window-flow control network. In the proposed model, there is more than one logical channel (LC), and the input queue for each LC has a buffer with finite capacity. The interference among LCs is also taken into consideration. The window-size allocation problem is discussed, using the proposed method of analysis, and it is shown that the window size,which is 1.5 times the number of transmission stages, is the optimum under the condition of uniform traffic. When the traffic in the network is not distributed uniformly, bias toward either or any LC may occur. By taking such an uneven situation into consideration, not only the delay and the throughput but also the fairness among LCs are examined. It is shown that the window-size allocation equalizing the input regulation degree among LCs is more effective. [ABSTRACT FROM AUTHOR]

In the digital radio system using dual polarizations, an interference canceller is needed that cancels excessive interference. In this paper, a fully digital cross-polarization interference canceller is proposed in which digital signal processing (DSF) technology and LSI technology are applied. By means of DSP technology, the cancellation characteristics are determined mainly by the quantization accuracy and the number of taps. The only degradation factor is the rounding-off error of the operations. High accuracy and large cancellation effects are expected. Also, an application of an advanced control algorithm is possible. Flexible applications are possible for any symbol rate. In this paper, the estimation method for the fully digital cross-polarization interference canceller is presented. Based on this method, the relationships among the cancellation characteristics, the input signal quantization accuracy, tap coefficient quantization accuracy, and number of taps are analyzed theoretically for the case where the method is applied to the 256 QAM system. Next, the configuration of the canceller is presented. Prom the results of the experiments using the symbol rate of 12.5 MBaud and 256 QAM signals, good agreement with the theoretical prediction is obtained and, hence, the validity of the present estimation method is demonstrated. It is demonstrated that a high-performance cross-polarization interference canceller is realized which can be applied to a multilevel modulation system for a high-speed region, such as 12.5 MBaud. [ABSTRACT FROM AUTHOR]

In the control system using a digital computer with finite word-length, the control performance is degraded by the quantization error of the signal and the control parameter, as well as the round-off error in the computation. Consequently, one of the important problems is to establish the design method which can reduce the effect of quantization error. This paper considers the deadbeat control system with an observer, and analyzes the round-off error in the floating-point computation. The effect of the floating-point computation. The effect of the round-off error on the control system is evaluated quantitatively. Finally, a numerical example is presented demonstrating the validity of the result in this paper, as well as the effectiveness of the parameter determination method for the observer to reduce the effect of the error. [ABSTRACT FROM AUTHOR]

This paper deals with a design of digital transmitter and receiver IIR Nyquist los-pass filter pairs for data transmission. The transmitter and receiver filters are required to limit the spectrum bandwidth of a transmitting signal and a received signal, respectively, and to have zero intersymbol interference when the filters operate in a cascade. Due to the bandlimiting requirements, the sampling frequency F6 = 1/T6 for the filters must be larger than twice the data transmission rate 1/T. This paper supposes the receiver filter followed by the time domain equalizer. The filter design proceeds as follows: first, the transmitter and receiver filters are designed to ahve the same transfer function with a stop-band equi-ripple attenuation, considering the amplitude characteristics with the preassigned roll-off factor. The method does not require any iterative approximation. Secondly, the output of the receiver filter is decimated to the data symbol repetition rate T, then equalized in the time domain. The time domain equalizer consists of a linear or a decision feedback circuit, using an MS adaptive algorithm. The amount of delay given to the training signal supplied tot he equalizer affects the residual erros. Let the order of the equalizer (numerator and denominator has the same order) denoted by n. Then an optimum delay is found to be equal to nT for the linear feedback circuit and (n/2) T for the decision feedback circuit. [ABSTRACT FROM AUTHOR]

At present, the vehicular electronic system is composed if several independent subsystems. By contrast, this paper proposes an integrated system where the same functions are realized in a more reasonable and economical way, utilizing the digital multiplex communication and distributed control. The system aims at optimization of the connections of the components, considering only the distances among the subsystems and components, independently of their kinds and functions. To improve the economy, reliability and flexibility, the system is composed of a ring transmission system and a distributed processing system using the microprocessors and optical fibers. This paper discusses the functions and structures of the transmission and processing devices composing the system. The optimum number of transmission and processing devices as well as the processing ability of the system are examined. [ABSTRACT FROM AUTHOR]

The quasi-optical diplexer presently is considered to be an interesting as a special-purpose filter, which divides or mixes the electromagnetic waves above microwave frequency in free space. This paper reports on the new types of diplexers: the gridded square, gridded square frame and gridded double square types. The feature of these new types is that various frequency characteristics from broad to sharp, can be realized by adjusting independent parameters, and the symmetry or linearity around the resonant frequency can also be realized. This has been difficult in past diplexers. By a systematic experiment, the frequency characteristics of the new types of diplexers are examined. Empirical formulas are derived for the resonant and anti-resonant frequencies, as well as fractional bandwidth as functions of dimensional parameters. Based on these expressions of the characteristics, the design protocols are presented to meet various requirements. Although there still exists an error in the sense of reproducibility, which is less than 15 percent, the design for the proposed diplexers is established. Thus, the diplexers proposed in this paper have desirable characteristics, which have not been observed in the past. With the detailed design formulas for a wide range of frequencies from roughly 1 to 100 GHz, the diplexers can cope with various requirements, and a diversity of applications can be expected. [ABSTRACT FROM AUTHOR]

This paper aims at the total design of synchronizing clock phase quality in a master-slave synchronization network and studies the clock path phase errors. In master-slave synchronization, reference clock distribution paths are a major source of phase errors. It is useful to examine the phase errors using Time Interval Error (TIE), specified in units of time. In this paper, the TIE of clock paths containing multiplexers, digital transmission lines and digital synchronous terminals is analyzed. The measurements taken for multiplexers are presented by varying a stuffing ratio. Field data taken for a 7-link clock path, the longest effective link path in Japan, shows that in practice a TIE of less the 100 ns over a 10² - 10³ s mesurement period is measured, which agrees well with theoretical data. [ABSTRACT FROM AUTHOR]

FM demodulators using Digital Signal Processing (DSP) are smaller and more reliable than conventional analog FM demodulators. The authors have already proposed an improved tangent-type FM demodulator (I-TAN) capable of extending the demodulation range without introducing distortion. This paper proposes a new range-extended TAN-type FM demodulator (RE-TAN) capable of extending the demodulation range for wideband FM signal without increasing the sampling rate. The range can be extended incrementally by increasing the number of stages of the range extender proposed in this paper. [ABSTRACT FROM AUTHOR]

A new double-sided electromagnetic wave absorber allowing transmission is proposed. In the conventional permeable wave absorber, a resistive film is attached to a lossless dielectric layer. However, this is not nonreflective for an incident wave on the resistive film side. In the present paper, a basic structure is considered in which there is no reflection on both sides of the absorber. A simple structure made of two resistive films inserted between three dielectric layers is proposed. The nonreflection condition of the present wave absorber is derived by means of the equivalent circuit, and the reflection and transmission characteristics are derived for given permittivities of the dielectric layers. It is found that the bandwidth of the reflection coefficient does not significantly depend on the permittivity of the inner layer and that the permittivity of the outer layers is increased and that of the inner layer is decreased in order to make the transmission coefficient smaller. The wave absorber is fabricated and measured for confirmation of the validity of the theory. © 2007 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 90(6): 1–8, 2007; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.20369 [ABSTRACT FROM AUTHOR]

For pulse compression processing, a range sidelobe suppression method previously reported for the derivation of a pulse compression filter window function realizes a zero range sidelobe on the time constraint points and simultaneously minimizes the S/N loss corresponding to the signal loss. The present paper presents an evaluation by simulation of the pulse compression processing performance based on this design method. It is found that filter design by about fourfold oversampling is adequate and that characteristics better than those achieved by the conventional window function scheme can be obtained in terms of the peak sidelobe level, S/N loss, and compressed pulse width, indicating that the present design procedure is effective. The performance deterioration due to the Doppler shift is similar to that in the conventional method and thus is acceptable for practical purposes. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 89(1): 45–53, 2006; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.20233 [ABSTRACT FROM AUTHOR]

It is proposed to use the minimum product distance as the design criterion for a space-time trellis code (STTC) in a fast fading environment. Reference 1 discusses the minimum product distance for all error event paths. Reference 2, on the other hand, discusses the minimum product distance for the minimal error event path. This paper shows, as the first step, that by itself, the minimum product distance for the minimal error event path is not sufficient to evaluate correctly the frame error rate performance of STTC. Then it is shown that the minimum product distance for the error event paths that affect the error rate under consideration is fairly effective as a design criterion. It is further proposed to use the product distance distribution and the average trace of code word distance matrix for the minimal event path as design criteria for the STTC. A computer simulation demonstrates that the proposed design criterion is more effective than the conventional design criterion. A computer search based on the proposed design criterion shows that an STTC that achieves a better frame error rate performance than the conventional STTC can be discovered. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 88(11): 54–64, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.20181 [ABSTRACT FROM AUTHOR]

As the frequency of operation of recent mobile communication devices has increased and their bandwidth has expanded, the behavior of electromagnetic waves inside the devices has attracted attention. In the microstrip lines used for these devices, T-junctions are used for branching in general. This structure is more likely to be affected by the transmission line orientation. In contrast, the asymmetric branch is less restrictive. On the other hand, the transmitted power is divided in an unbalanced way in high-frequency regions. In this paper, the unbalance of the transmitted power at the asymmetrical branch is quantitatively evaluated and a solution is discussed. It is found that a notch provided near the branch is effective. FDTD analysis reveals that the transmitted power unbalance of 16.2% at 10 GHz is reduced to 2.3% by means of a notch at the asymmetrical branch. Measurement confirms the effectiveness of our method. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 88(5): 20–27, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.20164 [ABSTRACT FROM AUTHOR]

In this paper, we propose a demodulation method using maximum likelihood sequence estimation (MLSE) after coherent detection for partial response signaling single sideband (SSB) signals and present the bit error rate performance achieved by this configuration. Coherent detection is implemented by a pilot carrier method. The power of the pilot carrier inserted in the SSB modulated wave and the noise bandwidth of the phase locked loop (PLL) that extracts the base carrier cause the degradation in the power efficiency of the SSB signal. We then evaluate the equivalent carrier-to-noise power ratio (CNR) degradation of the pilot carrier method. The partial response demodulated signal can be MLSE decoded since the signal is time correlated. From the characteristics of an SSB demodulated signal, first-order MLSE decoding using only the in-phase channel generally results. Here, we also study a complex MLSE decoding method that considers the quadrature channel. Based on this configuration, we show that the SSB signal power efficiency was improved by about 5 dB compared to the conventional coherent detection and instantaneous identification and demodulation method. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 88(4): 44–53, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.20167 [ABSTRACT FROM AUTHOR]

With the rapidly spreading use of the Internet, many systems for connection to the Internet using various media and methods have appeared. One is a system using a satellite link as a unidirectional link, which is expected to provide broadband Internet service to areas in which a ground Internet environment has not yet been developed. However, most of the protocols which are regarded as standard on the Internet assume that the links composing the Internet can provide bidirectional communications. Consequently, when a unidirectional link is used as a part of the network, it is difficult to make direct use of the protocols which are considered standard on the Internet. Consequently, this paper considers the situation in which Internet connection service is provided by using a unidirectional link, and proposes and implements the required dynamic assignment of network resources and the associated management scheme. The system is also evaluated in the actual environment. It is shown that by applying the proposed method, it becomes possible to utilize a unidirectional link combined with a bidirectional link on the ground, resulting in an efficient Internet connection environment. It is also shown that the connection to the network is provided without constraints on its use. [ABSTRACT FROM AUTHOR]

In this paper, a typical mobile communication environment is assumed and the eigenvalue distributions of the correlation matrix of a MIMO channel with the spatial correlation of the multipath taken into account are studied by simulation. Based on this study, a simple method is proposed for accurately deriving the probability distribu- tion of the largest eigenvalue of the correlation matrix. The proposed approximation equation is compared with simu- lation results and good agreement is obtained. In order to understand the information transmission capability of MIMO in a multipath fading environment with spatial correlation, the proposed equation is used for studying the effect of the spatial correlation on channel capacity. Finally, the digital transmission characteristics (relationship be- tween the BER and the CNR) with correlation taken into account are obtained. [ABSTRACT FROM AUTHOR]

It is known that OFDM transmission systems give better performance than single-carrier systems in a multipath environment. However, the communication quality degrades rapidly when the delayed wave arrives beyond the guard interval. In order to solve this problem, various systems that try to suppress the undesired wave by using an adaptive array have been proposed. This paper considers fixed reception and investigates an MMSE adaptive array for OFDM that does not suppress waves arriving within the guard interval, only waves arriving beyond the guard interval. More precisely, the SMI optimization method is proposed in which a window function is used to extract only the wave components beyond the guard interval. In order to demonstrate the effectiveness of the proposed system, an evaluation is made by computer simulation. It is shown that the system operates well as an MMSE adaptive array in a fixed reception environment, exhibiting high suppression power for delayed waves beyond the guard interval. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 87(9): 1–8, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10221 [ABSTRACT FROM AUTHOR]

As a means of flexibly providing large-capacity access infrastructure at a low cost, the broadband fixed wireless access (BFWA) system using millimeter-wave and quasi-millimeter-wave frequencies is promising. In general, wireless channel quality variations are significant due to rainfall in wireless systems using millimeter-wave and quasi-millimeter-wave frequencies. Hence, deterioration of the channel working rate is a problem. In this paper, a multihop wireless access system with a mesh configuration is proposed that can deploy a millimeter- and quasi-millimeter-wave BFWA system at a high quality and a high density. In the proposed system, deterioration of the channel working rate due to wireless channel quality variations is avoided. In addition, the capacity is maximized by adaptively assigning the network resources for variations of the traffic demand and wireless channel transmission capacity. Further, as an adaptive control used in the proposed system, wireless channel assignment for a wireless link and load distribution among several paths within the mesh network are discussed. The problem is formulated as a mixed nonlinear optimization problem. In addition, a heuristic solution is proposed for solving the problem within a realistic time for a network of the anticipated size. The proposed algorithm is applied to a mesh network and the behavior and the performance of the algorithm are evaluated to prove its effectiveness. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 87(6): 22–33, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10133 [ABSTRACT FROM AUTHOR]