A field trial was conducted to evaluate the technical performance of land mobile message communication in different environments. The OmniTRACS system and the Ku-band JCSAT satellite were utilized as the mobile communications system and the satellite, respectively. The trial took place in September 1990 at different areas in Japan. Data collected correspond to about 65 hours of operation, during which a large number of messages were sent via the satellite. Two land mobile terminals operated simultaneously, each terminal having a function of generating messages automatically which simulates a large volume of traffic corresponding to about 50 terminals. Thus, the system was evaluated under the condition that 100 mobile terminals were in operation. Obtained data have been analyzed with a particular focus on the message transmission correlating with actual environments. The analysis was done by classifying environments into five categories: overall condition, type of roads, terrain, areas and weather conditions. The average transmission count per message experienced under all conditions is equal to 1.432 for forward messages transmitted from the hub station to mobiles, and 1.157 for return messages transmitted from mobiles to the hub station. With respect to the classification by the type of roads, for enample it becomes obvious that the performance is generally good except along roads of North-South orientation through dense urban areas. It is concluded that the message communications from/to mobiles are feasible in a wide range of environments, with the performance of success essentially depending on the visibility of satellite.

This paper analyzes the performance of the capacity controlled radio system under a frequency selective fading environment. This system controls the number of modulation levels according to the number of active subscribers. In the analysis, we consider the capacity controlled system either with or without adaptive equalizer. As the results of analysis, it is clarified that the system is superior to the conventional fixed capacity system, and can be considered as a good countermeasure for multi-path fading. And it is found that there exists a synergistic effect due to capacity controlling and adaptive equalizing.

Progress of speech recognition based on the hidden Markov model has made it possible to realize man-machine dialogue systems capable of operating in real time. In spite of considerable effort, however, few systems have been successfully developed because of the lack of appropriate dialogue models. This paper reports on some of technology necessary to develop a dialogue system with which one can converse comfortably. The emphasis is placed on the following three points: how a human converses with a machine; how errors of speech recognition can be recovered through conversation; and what it means for a machine to be cooperative. We examine the first problem by investigating dialogues between human speakers, and dialogues between a human speaker and a simulated machine. As a consideration in the design of dialogue control, we discuss the relation between efficiency and cooperativeness of dialogue, the method for confirming what the machine has recognized, and dynamic adaptation of the machine. Thirdly, we review the research on the friendliness of a natural language interface, mainly concerning the exchange of initiative, corrective and suggestive answers, and indirect questions. Lastly, we describe briefly the current state of the art in speech recognition and synthesis, and suggest what should be done for acceptance of spontaneous speech and production of a voice suitable to the output of a dialogue system.

In this paper, we investigate the discrepancy between a serial version and a parallel version of zero-knowledge protocols, and clarify the information "leaked" in the parallel version, which is not zero-knowledge unlike the case of the serial version. We consider two sides: one negative and the other positive in the parallel version of zero-knowledge protocols, especially of the Fiat-Shamir scheme.

Optical fiber transmission systems have advanced rapidly with the advent of highly advanced electronic and optical devices. This paper introduces several IC technologies required for ultra-high-speed optical transmission and overviews current IC technologies used for the existing and developing optical fiber trunk transmission systems. Future trends in device technologies are also discussed.

This paper describes the linguistic procedure of our speech dialogue system. The procedure is composed of two processes, syntactic analysis using a finite state network, and discourse analysis using a plan recognition model. The finite state network is compiled from regular grammar. The regular grammar is described in order to accept sentences with various styles, for example ellipsis and inversion. The regular grammar is automatically generated from the skeleton of the grammar. The discourse analysis module understands the utterance, generates the next question for users and also predicts words which will be in the next utterance. For an extension number guidance task, we obtained correct recognition results for 93% of input sentences without word prediction and for 98% if prediction results include proper words.

Theoretical and actual experimental investigations are made on a Loop Antenna System (LAS) consisting of N-gap loop antennas. A general formula for the LAS response to an external EM field is derived by using both the method of moments and techniques for transmission-line analysis. Numerical evaluation verifies that the LAS has favorable characteristics, such as invariable response within a wide frequency range. In addition, differences between the LAS method and the present open-field method in the results of measurement are evaluated. It should be noted that the LAS gives a much lower output for an electrical source than for a magnetic source, even if both sources produce the same value with the open-field method. It is concluded that conversion factors for the results from the LAS method to the open-field method strongly depend on the orientation of the source as well as its characteristics. Consequently, new conversion factors are proposed to obtain practical EMI measurements.

In this paper, we investigate the round complexity of zero-knowledge interactive proof systems of possession of knowledge, and mainly show that if a relation R has a three move blackbox simulation zero-knowledge interactive proof system of possession of knowledge, then there exists a probabilistic polynomial time algorithm that on input x{0,1}*, outputs y such that (x,y)R with overwhelming probability if xdom R, and outputs "" with probability 1 if x dom R. The result above can not be generalized to zero-knowledge interactive proof systems of possession of knowledge with more than four moves, because it is known that there exists a "four" move blackbox simulation perfect zero-knowledge interactive proof system of possession of knowledge for a nontrivial relation R.

Given an integer N, it is easy to determine whether or not N is prime, because a set of primes is in LPP. Then given a composite number N, is it easy to determine whether or not N is of a specified form? In this paper, we consider a subset of odd composite numbers +1MOD4 (resp. +3MOD4), which is a subset of odd composite numbers consisting of prime factors congruent to 1 (resp. 3) modulo 4, and show that (1) there exists a four move (blackbox simulation) perfect ZKIP for the complement of +1MOD4 without any unproven assumption; (2) there exists a five move (blackbox simulation) perfect ZKIP for +1MOD4 without any unproven assumption; (3) there exists a four move (blackbox simulation) perfect ZKIP for +3MOD4 without any unproven assumption; and (4) there exists a five move (blackbox simulation) statistical ZKIP for the complement of +3MOD4 without any unproven assumption. To the best of our knowledge, these are the first results for a language L that seems to be not random self-reducible but has a constant move blackbox simulation perfect or statistical ZKIP for L and without any unproven assumption.

Noise properties in cascaded erbium-doped fiber amplifiers used in subcarrier multiplexed analog video distribution systems are experimentally examined. The noise dependency on signal wavelength is measured for a 4 stage EDFA cascade, and it is shown that an optical narrow bandpass filter is not necessary after each fiber amplifier for signal wavelength of 1.5511.560µm and that optical bandpass filters are necessary for shorter wavelength than 1.551µm to avoid the noise degradation by spontaneous-spontaneous beat noise. Finally, the attainable distribution loss is estimated for AM and FM video distribution systems.

The security problems of object-oriented database system are investigated and security level assignment constraints and an access control mechanism based on the multilevel access control security policy are proposed. The proposed mechanism uses the Trusted Computing Base. A unique feature of the mechanism is that security levels are assigned not only to data items (objects), but also to methods and methods are not shown to the users whose security level is lower than that of the methods. And we distinguish between the security level of a variable in a class and that in an instance and distinguish between the level of an object when it is taken by itself and it is taken as a variable or an element of another complex object. All of this realizes the policy of multilevel access control.

This paper describes design issues of a speech dialogue system, the evaluation of the system, and the data collection of spontaneous speech in a transportation guidance domain. As it is difficult to collect spontaneous speech and to use a real system for the collection and evaluation, the phenomena related with dialogues have not been quantitatively clarified yet. The authors constructed a speech dialogue system which operates in almost real time, with acceptable recognition accuracy and flexible dialogue control. The system was used for spontaneous speech collection in a transportation guidance domain. The system performance evaluated in the domain is the understanding rate of 84.2% for the utterances within the predefined grammar and the lexicon. Also some statistics of the spontaneous speech collected are given.

We studied the manner of clarification and verification in real dialogs and developed a spoken dialog system that can cope with the disambiguation of meanings of user input utterances. We analyzed content, query types and responses of human clarification queries. In human-human communications, ten percent of all sentences are concerned with meaning clarification. Therefore, in human-machine communications, we believe it is important that the machine verifies ambiguities occurring in dialog processing. We propose an architecture for a dialog system with this capability. Also, we have investigated the source of ambiguities in dialog processing and methods of dialog clarification for each part of the dialog system.

This paper presents two methods for securely realizing caller-authenticated and callee-specified calls over telecommunication networks with terminals that accept IC cards having KPS-based cryptographic functions. In the proposed protocols, users can verify that the partner is the proper owner of a certain ID or a certain pen name. Users' privacy is protected even if they do the caller-authenticated and callee-specified calls and do not pay their telephone charge in advance.

In mobile communication systems using Dynamic Channel Assignment, channels are possible to be rearranged so that blocking probability can be made low. The smaller the number of cells where channels are rearranged, the smaller the load on the base stations in the cells. Also, we can reduce the deterioration of communication quality caused by reassingning a new channel to a call instead of the channel already assigned. In this paper, we consider not only how to rearrange channels but also which channel should be rearranged and assigned to a new call in rearrangement, and propose very simple but effective methods for rearrangement. The ways to select a candidate channel to be rearranged and assigned to a new call in the new methods make the number of cells where a channel is rearranged smaller. We also examine the relations between characteristics and the number of cells where a channel is rearranged. Using computer simulation results, the properties of the new rearrangement methods are compared with those of the traditional methods.

The rising demand for mobile communication is increasing the importance of efficient use of limited radio frequency resources. The assignment of radio channels to the cells of current cellular mobile radio systems, specifically, to each base station, has been much studied to increase efficiency in radio frequency use. Dynamic Channel Assignment (DCA) is one approach to this problem. This paper compares the basic characteristics of DCA with Fixed Channel Assignment (FCA) and describes the main DCA strategies. The most important current research topics on DCA are discussed, focusing on micro-cellular systems, which are considered indispensable in meeting the huge demand for future mobile communications.

Video service is slated to be a major application of emerging high-speed communications networks of the future. In particular, full-motion video is designed to take advantage of the high bandwidths that will become affordably available with the advent of B-ISDN. A salient feature of compressed video sources is that they give rise to autocorrelated traffic streams, which are difficult to model with traditional modeling techniques. In this paper, we describe a new methodology, called TES (Transform-Expand-Sample) , for modeling general autocorrelated time series, and we apply it to traffic modeling of compressed video. The main characteristic of this methodology is that it can model an arbitrary marginal distribution and approximate the autocorrelation structure of an empirical sample such as traffic measurements. Furthermore, the empirical marginal (histogram) and leading autocorrelations are captured simultaneously. Practical TES modeling is computationally intensive and is effectively carried out with software support. A computerized modeling environment, called TEStool, is briefly reviewed. TEStool supports a heuristic search approach for fitting a TES model to empirical time series. Finally, we exemplify our approach by two examples of TES video source models, constructed from empirical codec bitrate measurements: one at the frame level and the other at the group-of-block level. The examples demonstrate the efficacy of the TES modeling methodology and the TEStool modeling environment.

For the connection request procedure in mobile communication systems, a previous study had shown that the 3-channel systems provide the haighest maximum of stable per channel throughput. In this paper, we propose and study a new algorithm, called the Parallel Collision Resolution Algorithm, which can be implemented in a Q-channel connection request environment, where Q3. For the implementation, the channels are arranged in R groups, where R is a positive integer. The collision resolution scheme distributes the collided messages over all the groups so that throughput and delay measures can be improved. At any point in time, there can be a maximum of R collision resolution schemes operational irrespective of the channel or the group number over which collisions occurred. The performance measures are estimated by computer simulation. Under the new algorithm, almost the same level of the perchannel stable throughput measure of a 3-channel network can be achieved in networks for which Q3. This feature allows freedom to the network designer to employ a higher number of connection request channels without forfeiting high channel utilization rates. When Q is an integral multiple of 3, the maximum stable per channel throughput level achieved can be the same as that achieved by the 3 channel system, if the grouping of channels is such that each group consists of 3 channels. When Q is not an integral multiple of 3, the intuitive strategy of organizing the channels in such a way that Q/3 groups consist of 3 channels each and one group consists of (Q mod 3) channels, may result in much degraded performance. It is found that, if the channels are so organised that no group is composed of (Q mod 3) channels, the performance levels can be substantially enhanced. Also, under the new algorithm, the delay measure is significantly improved, particularly in schemes like the mobile satellite systems with high propagation delays. We conclude that the new scheme presents a promising collision resolution methodology for connection request procedures.

In this paper, we propose a new network organizing method for packet radio networks, a layered self-organizing method. In the layered self-organizing network, whole service area is divided into multiple sub-areas and one base station is settled in each sub-area. Communication links are settled in shorter time than the conventional self-organizing method. We evaluate the network organizing performance of the method by using simulations.

A location registration/paging procedure which is free from location registration area design is proposed. Each base station (BS) broadcasts a "responsibility area", composed of its own and neighboring cells' identification (ID). A mobile station (MS) makes a new location registration request when the current BS's responsibility area does not include the MS's registered location. Each BS is allowed to decide its own responsibility area autonomously based on route information, which is composed of neighboring cells' ID and reported from MSs. Therefore, the responsibility area can be adaptively changed based on MSs' moving characteristics. Moreover, this procedure solves the problems of registration traffic concentration and excess registration request on boundary BSs.