This paper is a study on the behavioral aspects of the input buffer limiting scheme whose basic feature is to award priority to the transit messages over the input messages so that congestion does not develop in the network. The numerical method employed in the analysis is that proposed in Ref.(7). The performance aspects are studied for different buffer capacities, different message handling capacities and different levels of reservation for transit traffic. The numerical method indicates that thrashing occurs at low levels of reservation for the transit messages, irrespective of the buffer size or the processor capacities of the node. This observation is supported by simulation results. With reference to the state-space of the model of our study, the congestion aspects are related to two Liapunov functions. Under the domain of one of the Liapunov functions, the evolution of the perturbed system is towards a congested state whereas, under the domain of the other Liapunov function, the evolution is towards a congestion-free state. Regardless of the configuration, it is found that the fundamental characteristic of the congestion under the input buffer limiting scheme is the characteristic of a fold catastrophe. In the systems with insufficient level of reservation for the transit traffic, the performance degradation appears to be inevitable, irrespective of the capacities of the nodal processor and output channel processor, and the size of the buffer pool. Given such an inevitability, the active life of a node under a typical node configuration is studied by simulation. A suitable performance index is suggested to assess the performance of deadlock-prone nodes.

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.

This paper describes voice communication connection controls in digital public land mobile networks (D-PLMNs). Voice communications in the D-PLMNs are carried at about 10 kbit/s over narrow-band TDMA channels with highly efficient cellular voice encoding schemes. Extensive research is being carried on half-rate voice encoding schemes that will effectively double radio resources. We first outline the configuration of voice communication connection between a cellular phone in the D-PLMN and a telephone in a fixed network, and we describe the optimum position for the CODECs that transform cellular voice codes to the conventional voice codes used in the fixed network, and vice versa. Then we propose a CODEC-bypassed communication control scheme that improves the quality of voice communication between cellular phones. And we propose a cellular voice code negotiation scheme in the D-PLMN which supports different cellular voice encoding schemes. We also propose an efficient channel reassignment scheme for effectively assigning TDMA channels to voice calls with two different bitrates (full-rate and half-rate), and we analyze this scheme's traffic capability. Finally, we describe a dual-tone multiple-frequency (DTMF) signal transmission scheme and estimate the number of DTMF signal senders required in the D-PLMN.

This paper describes a perspective on Personal Communicatoins Services (PCS) and technological trends. It takes into consideration rules pertaining to the use of PCS for mobile radio communication and countermeasures to cope with the huge increase in PCS subscribers. In this paper, PCS network structures, inter-regional roaming, microcell structure, radio access and channel access methods are also covered as PCS technologies. Furthermore, trends in domestic and international standards are also described. Although these technologies present many difficulties, we believe that they will be overcome and PCS services will be introduced in the near future.

We discuss fault tolerance of an information disseminating scheme, t-disseminate on a network with N processors, where each processor can send a message to t directions at each round. When N is a power of t+1 and at most tlogt+1N-1 (at most t) processors and/or edges have hailed, logt+1N+(f1)/t rounds (logt+1N+2 rounds) suffice for broadcasting information to all destinations from any source by t-disseminate. For a arbitrary N, logt+1N2f/t1 rounds (logt+1N+2 rounds) suffice for broadcasting information to all destinations from any source by t-disseminate if at most t(logt+1N1)/2 (at most t/2) processors and/or edges have failed.

This paper proposes a new fault tolerant intercommunication scheme for real-time operations and three new interconnection networks to construct a fault tolerant multi-processor system for pipeline processings. The proposed intercommunication scheme using bank memory switching technique has an advantage to make a fault tolerant pipeline system so that it can detect any failure caused in a processing element of the system. In addition, it can overcome conventional problems caused in interconnection circuits to flow data with one way direction such as a pipeline processing.

This paper presents a new architecture of a copy network which employs the principle of recursive generation of copy cells. The proposed architecture achieves high utilization of the links and buffers of the copy network, and preserves the cell sequence. The architecture lends itself modularity so that large multicast ATM switches can be fabricated by employing the proposed copy network. Two different modular structures - one for reduced latency of the unicast cell and the master cell from which copies are made, and the other for reduced hardware overhead - for realizing large multicast ATM switches are configured. The hardware of functional elements of the copy network are the same as those of the functional elements of a modular point-to-point switch proposed earlier, thereby resulting in the modularity of functional elements as well. Simulation studies show that the proposed copy network achieves high throughput and low cell loss probability, and the required buffer sizes are small. The delay of cells is found to be very small for traffic loads up to 90%.

This article discusses a synthesis procedure of a discrete-time asynchronous neural network whose information is a limit cycle. The synthesis procedure uses a novel connection matrix and can be reduced into a linear epuation. If all elements of desired limit cycles are independent at each transition step, the equation can be solved and all desired limit cycles can be stored. In some experiments, our procedure exhibits much better storing performance than previous ones.

Rate-based congestion/flow control is a promising way to achieve high throughput in high speed packet-switching networks. We consider a rate-based congestion control to aim at obtaining high throughput and fair sharing of the communication resources. In the scheme, each intermediate node informs its congestion status to the source node. Two kinds of control packets are used for this mechanism. One (a choke packet) is to throttle the rate and another (a loosen packet) is to allow increase of the rate. The source node initiates transmission with a low rate and increases the rate slowly to avoid a rapid increase of the packet queueing at an intermediate node. When the source node receives a choke packet, it decreases the rate rapidly to relieve congestion as soon as possible. The source node upon receipt a loosen packet increases the rate slowly again. We develop a queueing model to investigate the parameter settings to provide a good performance via simulation. The increasing and decreasing parameters of the rate control function are first investigated in various load conditions. We next examine the effect of the queue-length threshold value for the indication of congestion at the intermediate node. The numerical results indicate that the threshold value should be small to obtain a good performance. We finally introduce a technique which accurately recognizes congestion and inhibits an acceptable queueing of the packets at intermediate nodes.

In this paper, the software structure for telecommunication user support are discussed, and it is proposed to apply knowledge processing technology to the software. Capabilities of telecommunications networks are becoming quite complicated, and the number of service items and parameters which have to be selected and memorized will become too large for telecommunications end users to make full use of the network capabilities. As such, more effort should be focused on assisting telecommunications end users to use the network and providing user friendly human interfaces of the network. However, this kind of software has additional type of requirements other than those for protocol handling software and call control software, and the realization of such support software has not yet been fully studied. To realize such support software, this paper stressed the realization of the user-system interface. Especially identified in this paper are meaning-based interpretation of user inputs to permit the handling of synonyms and multivocations, and a method to access the database in the support system without consideration of its data schema. To satisfy these objectives, this paper has proposed that the application data should be represented in both a character string and a meaning representation, and that the thesauruses should have the attribute-value relation. In line with these studies, an experimental system called CAPRIS (CAlling PRocedure Instruction System) was developed. It is used to assist the calling party in a telecommunications network to find an appropriate contact point depending on the purpose of the communication. Implementation of CAPRIS is completed and it was confirmed that all the functions described in this paper were actually realized. Some functional experiments were performed on CAPRIS, and the system was concluded to realize satisfactory user-friendliness.

In order to make the implementation of network components flexible and cost effective, it is required to use widely available technologies as the implementation platform. The distributed operating systems can be adopted as such a platform, because they allow to implement a network component using multiple computers connected through a local area network. In this paper, we focus on the Intelligent Network (IN) whose network components are modelled as Functional Entities (FEs), and describe an implementation method of FEs using distributed operating systems. Our method is summarized as follows: The remote procedure call (RPC) is used for the access transparent inter-process communication. The lightweight process mechanism is used for handling concurrent requests. CCF/SSF (Call Control Function/Service Control Function) and SDF (Service Data Function) are implemented as an SSF server and an SDF server, respectively. SCF (Service Control Function) is composed of a Service Dispatcher and a set of Service Executors. The Service Dispatcher accepts all the requests for IN call processing and dispatches them to appropriate Service Executors. Service Executors are created for the individual IN services and execute the service logics. SDF server and Service Executor may be replicated for load partitioning.This paper has also described the implementation of experimental system supporting "Freephone" service based on our method, and showed the performance evaluation of the experimental system in terms of the real-time and concurrent call processing of IN services. We used Mach and SUN OS as a platform for implementing the servers for FEs. The experimental system using four workstations shows that it can handle up to 170IN calls in one second with the additional response time of less than 200msec, which is small enough compared with the response time for the basic connection control. Those results prove that our method is feasible for implementing practical FEs.

This paper describes a VLSI processor architecture designed for a back-propagation accelerator. Three techniques are used to accelerate the simulation. The first is a multi-processor approach where a neural network simulation is suitable for parallel processing. By constructing a ring network using several processors, the simulation speed is multiplied by the number of the processors. The second technique is internal parallel processing. Each processor contains 4 multipliers and 4 ALUs that all work in parallel. The third technique is pipelining. The connections of eight functional units change according to the current stage of the back-propagation algorithm. Intermediate data is sent from one functional unit to another without being stored in extra registers and data is processed in a pipeline manner. The data is in 24-bit floating point format (18-bit mantissa and 6-bit oxponent). The chip has about 88,000 gates, including microcode ROM for processor control, the processor is designed using 0.8-µm CMOS gate arrays, and the estimated performance at 40 MHz is 20 million connection updates per second (MCUPS). For a ring network with 4 processors, performance can be enhanced up to 90 MCUPS.

The essential functions of the passive double star (PDS) system are clarified by comparing them to the functions of the single star (SS) and the active double star (ADS) system. A layered structure describing the functional characteristics of the PDS system is proposed for flexible transport capability. The functions of the optical network unit (ONU) on the customer premises are systematically partitioned into four layers. The functions of the optical subscriber unit (OSU) in the central office are described using five layers. Call by call activation and deactivation techniques are described on the basis of a layered architecture. The reduction of ONU power consumption by adopting activation and deactivation control is also discussed.

Fiber-optic subscriber loops are beginning to be constructed worldwide as an infrastructure for the 21st century. Making the most of the superior transmission capabilities of optical fiber cable, high-speed and broadband services such as so-called Broadband-ISDN (B-ISDN) services are expected to be achieved. We review current trends in fiber-optic systems and discuss how fiber-optic subscriber loops should be implemented from the points of view of services and technology. First, prospective services offered over fiber-optic subscriber loops and resulting requirements for such loops are presented. Various kinds of services produced through fiber-optic subscriber loops will further develop the information-oriented society and have a major influence not only on the industrial world but on family life as well. Next, basic concepts of totally fiber-optic subscriber loops and proposals for shifting from the existing metallic cable network to totally fiber-optic subscriber loops are discussed. In this regard, we must consider the advantages of optical fiber cables and improve upon the disadvantages of the existing metallic cable network. In particular, a Digital Loop Carrier (DLC) system or a Passive Double Star (PDS) have been proposed to shift from the existing metallic cable network to totally fiber-optic subscriber loops effectively and economically. Finally, line configuration topology and latest fiber-optic technologies such as high-count cable, construction techniques and operation systems for use in achieving totally fiber optic subscriber loops are shown. We believe a single star configuration is the most appropriate for totally fiber-optic subscriber loops.

We report on 1N branching devices for fiber-optic networks. A novel Y-pattern mask for loss reduction in branching waveguide has been investigated based on simulation. 18 branching waveguides fabricated by ion-exchange proved that the scattering loss at the branching region made with novel Y-pattern is reduced considerably. Pigtailing technique has been developed using glass fiber arrays (FAs). The fibers in FA are aligned precisely, so that all fibers can be easily and precisely put into alignment with the output ports in a branching waveguide chip at the same time. A glass plate is used for reinforcement which improves mechanical and thermal stability of the device. Reliability of packaged devices is being strongly demanded. Various reliability tests have been performed with packaged 18 branching devices and those results are presented.

We study the most appropriate network architecture for constructing FTTH (Fiber To The Home) networks. We conclude that the Single Star network is the most advantageous for supporting the various services required by subscribers who use broadband signal such as B-ISDN (Broadband-Integrated Services Digital Network). We also study high fiber count cable and low-loss connector which are needed for this network and clarify their requirements. We also show the cable structure, connector structure and connection loss reduction method, which satisfy these requirements. We describe a 4,000-fiber cable with a new structure, a 1,000-fiber one-touch connector, a 40-fiber unit connector and TEC-fiber (Thermally-diffused Expanded Core fiber) which reduces connection loss.

A novel harmonic retrieval algorithm is proposed in this paper based on Hopfield's neural network. Frequencies can be retrieved with high accuracy and high resolution under low signal to noise ratio (SNR). Amplitudes and phases in harmonic signals can also be estimated roughly by an energy constrained linear projection approach as proposed in the algorithm. Only no less than 2q neurons are necessary in order to detect harmonic siglnals with q different frequencies, where q denotes the number of different frequencies in harmonic signals. Experimental simulations show fast convergence and stable solution in spite of low signal to noise ratio can be obtained using the proposed algorithm.

This paper describes the integration of a qualitative method and a quantitative method by Bottleneck Diagnosis/Improvement Expert Systems for Synchronized queueing network (BDES-S and BIES-S). On the basis of qualitative reasoning, BDES-S can carry out parameter tuning in order to diagnose and improve bottlenecks of synchronized queueing networks. BDES-S can produce several alternative qualitative improvement plans for one bottleneck server. BIES-S can produce quantitative improvement equations for each qualitative improvement plan. Our method using BDES-S and BIES-S can integrate both quantitative and qualitative methods for parameter tuning on complicated queueing synchronized networks.

In this paper we examine the architectural and operational design issues of a practical network management system using the Simple Network Management Protocol (SNMP) in the context of a large-scale OSI-based campus-network TAINS. Various design aspects are examined and the importance of time-management is elicited. In the proposed design, intelligent, time-synchronised agents are deployed to collect information about the network segments to which they are attached. The manager talks to the agents and gathers relevant network information. This information is used by the expert network manager, in conjunction with a network knowledge base (NKB) and a management information knowledge base (MIKB) , to reconstruct the overall network-traffic characteristic, to evaluate the status of the network and to take/suggest some action. This model is particularly useful in networks where some global control, monitoring and management is desired and installing agents on all elements, connected to the network, is impossible. The use of time labels and narrow time windows enables the manager to obtain a reasonably accurate picture of the network status. The introduction of time-labelled composite objects in the Management Information Base (MIB) provides a means of reducing the load of management-related traffic on the network. The MIKB containing a logical description of the behaviour of the managed objects defined in the MIB, drives the expert system and provides the knowledge of general nature that a human expert has about networks. The proposed MIKB concept provides a very convenient schema for building the knowledge base in an expert network management system. Further since the MIKB is MIB-specific, it can be used in network management systems for managing similar MIB's.

As the size and complexity of network increases, the distributed systems management (DSM) will be vital in order to improve the availability of network management, to reduce the complexity of network operations, and to provide the high reliability. In this paper, we (1) describe general requirements required for systems management in distributed environment, (2) introduce the basic structure of distributed network management system (DNMS) for efficient network management, and also especially illustrate the concrete design of system management application process which is an important element among them, (3) propose the connectionless CMIP to accomplish for effectively managing the distributed management system, and indicate its efficiency; this protocol is available to manage dynamically changing DSM environment, to negotiate among the managing systems, to handle the dynamic informations etc. Finally, (4) the behavior and software structure of a suggested management system during negotiation to execute the received request for implementation are presented.