The Distributed Queue Dual Bus protocol (DQDB) has been adopted as the metropolitan area network (MAN) standard by IEEE802.6 committee. Recently, the unfairness problem in the DQDB protocol, by which head stations benefit, has been pointed out. Although a fair bandwidth distribution among the stations is obtained by adding the so-called bandwidth balancing mechanism into the DQDB protocol (DQDB/BB), the DQDB/BB protocol leaves a portion of the available bandwidth unused, and it takes a considerable amount of time to converge to fair channel assignment. In this study, to overcome the drawbacks in DQDB and DQDB/BB, we introduce a new media access control protocol which is based on assigning each station a level according to some traffic information such as the queueing length, delay time etc. Only the station with the highest level is allowed to transmit. Through the operation of level assignment or the choice of level function, the transmission can be easily controlled in a distributed manner. This protocol is simple compared with DQDB/BB and can be implemented in the DQDB architecture. The simulation results show that the new protocol obtains not only fair throughput regardless of the distance between the stations, but also fair delay performance. In addition, the new protocol can easily provide preempty priority service through level assignment. The new protocol converges to fair distribution of the channel in the time required for only one or two round-trips. This is very fast compared with the DQDB/BB protocol.

This paper discusses two approaches to statistical multiplexing: rate envelope multiplexing, allowing resource sharing with small delays for low peak rate connections, and rate sharing, based on the use of large multiplexer buffers to ensure high link utilization for high speed data traffic. We argue that the weighted fair queueing scheduling algorithm provides an efficient means for combining both kinds of multiplexing in the B-ISDN. A feasible implementation known as Virtual Spacing is outlined. We illustrate the flexibility of the proposed scheme by showing how different service categories could be provided.

A Hierarchical Cubic Network (HCN) is a hierarchical hypercube network proposed by Ghose. The HCN is topologically superior to many other similar networks, in particular, the hypercube. It has a considerably lower diameter than a comparable hypercube and is realized using almost half the number of links per node as a comparable hypercube. In this paper, we propose the shortest routing algorithm in HCN(n, n) and show that the diameter of HCN(n, n) with 22n nodes is n(n1)/31 which is about 2/3 of that of a comparable hypercube. We also propose the optimal routing algorithm in HCN(m, n) where mn and obtain that its diameter is n(m1)/31. Typical parallel algorithms run in HCN(m, n) with the same time complexity as a hypercube and the hypercube topology can be emulated with O(1) time complexity in it.

Over the last decade, the Internet has been extremely successful by distinguishing between overlaying applications and underlying networking technologies. This approach allows rapid and independent improvement in both networking and application technologies. The internetworking layer that divides applications and the network enables the Internet to function as a general and evolving infrastructures for data communications. The current Internet architecture offers only best-effort data delivery. However, recent emerging computer and networking technologies, demand the Internet guaranteed performance. In particular, audio and video applications have more rigid delay requirement than those applications which the current Internet supports. To offer guaranteed services in addition to best-effort services, both a new service model and a new architecture are necessary in the Internet architecture. The paper surveys researches and experiments conducted in the Internet community to accommodate a wide variety of qualities of services.

Calls using different media which require different transfer quality will arrive at ATM networks. Therefore it is important to develop a method for allocating network resources efficiently to individual calls by judging admission of calls. Various call admission control schemes have been already proposed, and these schemes assume that users specify values of traffic descriptors when they originate calls. However, it is sometimes difficult for users to specify these values accurately. This paper proposes a new ATM call admission control scheme based on cell transfer state monitoring which does not require that users specify values of traffic descriptors in detail when they originate calls. In this proposed scheme, the acceptance or rejection of calls is judged by comparing the monitored cell transfer state value with a threshold prepared in advance. This threshold must be adjusted according to changes in the characteristics of traffic applied to ATM networks. This is one of the most serious problems in the control scheme based on the monitoring of cell transfer state. Herein, this paper proposes neural network application to the control scheme in order to resolve this problem and improve performance. In principle, the threshold can be adjusted automatically by the self-learning function of the neural network, and the control can be maintained appropriately even if the characteristics of traffic applied to ATM networks change drastically. In this paper, the effectiveness of the application of a neural network is clarified by showing the configuration of this proposed control scheme with the neural network, a method for deciding various parameter values needed to implement this control scheme, and finally the results of a performance evaluation of the control scheme. Inputs required by the neural network are also discussed.

This paper reports an overload control method for the Intelligent Network (IN). The IN, which is being investigated as a future communication network, facilitates both rapid introduction of new services and easy modification of existing services. In the IN, the call processing functions and data needed to achieve IN services are distributed over several nodes. Therefore, traffic demand for the various services may cause varying patterns of node overloads. It is therefore important to develop effective overload control methods and to evaluate their characteristics. We propose an overload control method and evaluate its characteristics in comparison with other methods under various overload traffic patterns with a network simulator that models all nodes and their relationships in the IN. In particular, we focus on three aspects of overload control: how can high throughput be maintained, how can an overloaded node be stabilized, and how can fair access be guaranteed.

We propose a new concept of network dimensioning, which is based not only on the grade of service but also on profit. In traditional network dimensioning methodology, the number of circuits on links is designed under a cost-minimization concept with grade of service constraints. Recently, telecommunication markets have become very large and competitive; therefore, we believe that a profit viewpoint is now essential. However, it is difficult to calculate profit in almost all the dimensioning methods currently used, because they mainly employ peak-hour traffic data, while profit depends on all the hourly traffic data which contain both peak and off-peak data. In this paper, we propose using all the hourly traffic data in network dimensioning. From these data and telephone charges for each hour, revenues will be estimated. On the other hand, facility costs will be estimated from the number of circuits. Finally, we can estimate profit from the difference between revenues and facility costs. Focusing on both quality and profits in network dimensioning leads to more advanced quality management and quality control in telecommunications networks than with traditional methodology. This paper outlines a dimensioning method based on profit, and describes its properties, some applications of it, and summarizes further studies.

Switched-capacitor chaotic neurons fabricated in a full-custom integrated circuit are used to investigate the behavior of 2- and 3-neuron chaotic neural networks. Various sets of parameters are used to visualize the dynamical responses of the networks. Hysteresis of the network is also demonstrated. Lyapunov exponents are approximated from the measured data to characterize the state of each neuron. The effect of the finite length of data and the rounding effect of data acquisition system to the computation of Lyapunov exponents are briefly discussed.

We study the correspondence between problem descriptions and requirements specification documents derived from them. Based on the results of this investigation, a model that integrates the problem space and the requirements specification space is developed. This integration is based on a semantic network representation. We also propose a model of the requirements elicitation process that is consistent with our empirical studies of traceability in requirements documents. In this process, analysts derived requirements specifications from incomplete and ambiguous problem descriptions given by customers, identify missing information, completed it, and then decide the system boundaries that define which part of the problem descriptions to implement as the target system. The model can be used to complete problem descriptions given by customers and determine the system boundaries.

We investigated performance of routing schemes in B-ISDN, for heterogeneous traffic flows under various bandwidths. In particular, we compared the simulated performance of these schemes by evaluating their blocking probabilities. To achieve high performance, these schemes use special kinds of routing algorithm, one which is pre-selection algorithm and one which is cyclic algorithm. We investigated the efficiency of the pre-selection algorithm and the robustness of the cyclic algorithm for nonuniform traffic and network resources. We found that these routing algorithm schemes can compensate for errors in resource design.

In this paper, the ATM Mini-Bar System (AMBS) which is a future information providing service infrastructure is proposed. The purpose of AMBS is to provide a multi-media environment in which a user can (1) select and get quickly any needed information, in low cost, at any time, among very large amount of different media information provided by a variety of providers, (2) be charged only for the information which is selected and used, (3) edit or process informations into users' individually requested style or format before using them. The basic concept and configurations of AMBS are also addressed. This system is basically a center-end oriented one-way information providing system. The information center broadcasts its contents to all user equipments based on a user request forecast, and every user equipment stores the delivered contents in its large storage. A user can select one's needed informations from the storage, and may edit or process them within the user equipment. The charge is only on the read informations from the storage, not on all contents in it. The key points of this system are the following three. (A) Introduction of a broadcast (or multicast) media for economical information delivery (exactly speaking, it is a predelivery which means a delivery before request) to user equipments. (B) Introduction of a 1 to 1 communication network for selective charging and control of each user equipments. (C) Introduction of the user equipment storage for Quick response to user information request in most cases with the broadcast (or multicast) information delivery media described above, Separation of information delivery speed and replay speed to increase system flexibility, Local user information processing or editing. As an example of technical solutions, a memory architecture, which is based on hierarchical architecture, is described. AMBS is expected to give some impacts to information industries because it can integrate many kinds of services into the same platform, but some standerdization items are needed to realize it.

Next generation personal communication systems will provide a range of different services to moving users. In parallel, packet switching is being proposed as a way to statistical multiplexing and hence to better resource exploitation. The co-existence of different services may prove difficult due to the different requirements on quality of service parameters like packet loss, delay, and so on. This requires a careful design of Call Admission Control policies, which are to be quite different from those used in fixed network, due to two phenomena which are typical of mobile systems, namely co-channel interference and handovers. In this paper we address these complex topics, and propose some basic rules for Call Admission Control policies suitable in this context.

We propose an asymmetric neural network which can solve inequality-constrained combinatorial optimization problems that are difficult to solve using symmetric neural networks. In this article, a knapsack problem that is one of such the problem is solved using the proposed network. Additionally, we study condition for obtaining a valid solution. In computer simulations, we show that the condition is correct and that the proposed network produces better solutions than the simple greedy algorithm.

Circuit access control is a traffic control technique of rejecting calls arriving at a group of specified circuits to make the group free at a target scheduled time so that the capacity may be dynamically reallocated to serve other traffic demand. This technique plays an important role for resource allocation control in state-of-the-art capacity reconfigurable networks as well as for switching calls on a reserved basis in the ISDNs. In this paper, we present a novel adaptive scheme for circuit access control in order to overcome the inefficiency of the conventional deterministic scheme. The presented scheme is based only on knowledge about service time and bandwidth characteristics of calls. The transitional behavior of the circuit group under the scheme is analyzed, and the gain in utilization achieved by the adaptive scheme is examined. We treat a model of the circuit group shared by multi-slot calls with different service times, and describe the results of the transient analysis and the approximation method for evaluating the gains.

The performance of a nonblocking switch with two kinds of correlated input calls is analyzed. We define two kinds of calls as the waiting call and the immediate call, and assume that the immediate call has the priority over the waiting call. If the traffic density of one kind of calls is larger than maximum throughput, the ratio of the corresponding kind of calls to the total traffic must be restrained in some range. We derive the maximum ratio of the waiting call by using two approximate methods. The effects of traffic densities and transition probabilities of two kinds of calls on the maximum ratio of the waiting call are also considered. It is shown that, if the traffic density of the immediate call is smaller than that of the waiting call, our approximate methods are useful to derive the maximum ratio of the waiting call to the total traffic.

This paper deals with network performance of 64kbit/s-based services supported in an ATM network and an ATM interworking network with 64kbit/s-based networks. It first clarifies network performance issues giving a model and objectives of study and experiment. A result of computational analysis is then presented, where a cell loss ratio of an order of 10-4 or 10-5 is obtained to give a performance equivalent to that currently used as objective values in existing networks for a 64kbit/s digital level. In order to capture the impact of cell loss and associated performance for application levels, an experimental test has been carried out using typical applications. Test results show that the cell loss ratio needs to be approximately ten times better than the bit error ratio for comparable performance for application levels. A cell loss ratio of better than 10-5, or an order of 10-6 considering an interworking situation, seems to be necessary according to the test results. It is further clarified by the test that a single cell is more valuable than a multiplexed cell for providing better cell loss resilience characteristics. Although not all applications of 64kbit/s networks have been tested, it is expected that the test results can be used as guidance for considering the support of 64 kbit/s services in an ATM and interworking networks.

Telecommunication services are expected to be upgraded from POTS to B-ISDN services in the future. This means that the conventional metallic access networks should be upgraded to optical fiber access networks because of providing high bit-rate services. Therefore, it is very important to clarify upgrade strategies in access networks. This paper proposes a dynamic evaluation method that can support decision-making on the upgrade strategy from the viewpoint of economy. This method can determine the most promising future access network and upgrade timing. Moreover, viability of various upgrade strategies can be evaluated by this method.

Connectivity (of node-to-node) is generally used to examine the robustness of graphs. When telecommunication network switches are integrated into logical switching areas, we should examine node-to-area connectivity rather than node-to-node connectivity. In a previous paper, we proposed node-to-area (NA) connectivity using area (subset of nodes) graph. In this paper, we consider a further constraint: "there is a path that does not include other nodes in the source node area." We call this property, directly NA-connected. Application of this constraint makes telecommunications networks robust against locally striking disasters. The problem of finding the maximum number of edge deletions that still preserves the direct NA-connection is shown to be NP-hard. It was shown in our previous paper that an NA-connected spanning tree is easily found; this paper shows that the problem of finding a directly NA-connected spanning tree is also NP-hard. We propose an O(|E||X|) approximation algorithm that finds a directly NA-connected spanning subgraph with an edge nummber not exceeding 2|V|3 for any NA-connected area graph that satisfies a described simple condition. (|V|,|E|,and |X| are the numbers of nodes, edges, and areas, respectively.)

This paper describes, based on generation mechanism of conductive noise, that the real conductive noise on AC-mains can't be measured by LISN and 50 Ω-input impedance instrument specified by regulations such as CISPR. Second, it is pointed out that one of the causes of poor reproducibility in radiated emission measurement is the difference among line impedances of AC-mains. Finally, it is insisted that the apparatus such as LISN is necessary for stable measurement of radiated emission, and what improvement on LISN characteristics for higher frequency range should be done is introduced.

This paper describes an advanced rule-embedded neural network (RENN+) that has an extended framework for achieving a very tight integration of learning-based neural networks and rule-bases of existing if-then rules. The RENN+ is effective in pattern recognition with ill-posed conditions. It is basically composed of several component RENNs and an output RENN, which are three-layer back-propagation (BP) networks except for the input layer. Each RENN can be pre-organized by embedding the if-then rules through translation of the rules into logic functions in a disjunctive normal form, and can be trainded to acquire adaptive rules as required. A weight-modification-reduced learning algorithm (WMR) capable of standard regularization is used for the post-training to suppress excessive modification of the weights for the embedded rules. To estimate the effectiveness of the proposed RENN+, it was used for pattern recognition in a radar system for detection of buried pipes. This trial showed that a RENN+ with two component RENNs had good recognition capability, whereas a conventional BP network was ineffective.