When communication network planning-design is performed, especially in a short-term case, it is important to utilize existing facilities in the construction of the new network. In this paper, link capacity assignment problem (CA problem) for packet-switched networks is investigated with the consideration of the existing network. To deal with this, per-unit cost of existing link capacity is thought to be less than that of newly installed capacity and a link cost function is modeled by a non-linear, non-differentiable one which is composed of two portions of capacity cost. After formulating the CA problem, two optimum algorithms derived from Lagrange multiplier method are presented and a modified algorithm is used for solving the CA problem in order to reduce the computation time. Some numerical results show that according to the values of link traffic flows, there will be links whose capacities must be set equally to the existing values. Moreover, when link cost difference is introduced in the CA problem, the number of links that the capacities of which have to be changed from existing values is less than that of linear cost function case, i.e., the case without consideration of the cost difference in link capacity.

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.

Managed Objects (MOs) can be specified by the combination of a static information model and a dynamic process model. First, this paper presents a mapping of attributes from a process model diagram to an information model diagram. Then, it introduces a concept of topology into these two models and proposes a hypothesis about the relationship of topology in these two models. To explicitly explain the hypothesis, it can be stated that all attributes of incoming or outgoing data related to a process in a process model are mapped to an information model where these attributes are interconnected by an explicit relationship which corresponds to a specific meaning, such as physical containment or logical connection. From an intuitive perspective, it can be said that if two attributes have a close topological relationship in a process model, the mapped attributes also have a close topological relationship in an information model. This hypothesis provides clues for determining whether there is an error in an attribute either in the process model or in the information model. By examining the way attributes of incoming or outgoing data related to a process are mapped to an information model, we can detect whether there is an error with respect to the process. The error correction is performed with the assistance of probability analysis. The method of error detection and correction can be implemented in a computer aided tool. Then, error detection on the attribute level becomes automatic, and error correction on the attribute level becomes interactive through the computer aided tool. Finally, the validity of the hypothesis is confirmed by analyzing ITU-T Recommendation M.3100. The specification of the fabric object class defined in M.3 100 is transformed into these two models and the hypothesis is validated for the analysis of the mapping between these two models.

This paper compares three typical system-sharing configurations for FTTH networks that provide narrowband and video distribution services and proposes a remote node locating strategy for each configuration. Two new evaluation factors, required land space and service provisioning effort, are included in the calculation, in addition to facility cost and maintenance effort. By considering these factors together, the total network cost is calculated and the sensitivity to the number of remote nodes is evaluated. Finally, the most economical system-sharing configuration is identified on the basis of the evaluations for two typical service areas in Japan, for both present and future cost environments.

In this paper, we propose the architecture of BALANCE (Better Adaptive Load-balancing through Acquiring kNowledge of Characteristic of an Environment) in which users can submit their jobs without acquiring either a status of an environment or characteristics of jobs and servers even in a widely connected heterogeneous network. The architecture of BALANCE includes three types of information bases and two types of daemons. Information bases, namely job, resource, and environment information base, manage the knowledge of job characteristics, available resources for CPUs, and status of the environment, respectively, as a proxy for users. The dispatching daemon selects an adequate server for each job using knowledge stored in the information bases. A service daemon executes each job. On completing each job, a service daemon gets a statistic of the job and returns it to the dispatching daemon where the job came from so that the statistic will be available at the next dispatching time. BALANCE enables an environment (1) to balance the load, (2) to share software functions as well as hardware facilities, and (3) to learn a user's job characteristics. We have implemented a prototype with more than 50 heterogeneous UNIX workstations connected by different networks. Two simple experiments on this prototype are presented. These experiments show a load balancing scheme that takes the characteristics of each job into account.

The concept of flexible system is long being used by many researchers, aiming to solve some particular problem of adaptation. The problem is viewed differently in different situations. In this paper, we first give a set of definitions and specifications to generalize this concept applicable to any system and in particular to communication networks. Through these definitions we will formalize, what are the conditions a system should satisfy to be called as a Flexible Communication System. The rest of the paper we formalize the concepts of flexible information network, and propose an agent oriented architecture that can realize it.

With the arrival of B-ISDN, widespread usage of multicast services such as TV broadcasting and video 900 services will increase the possibility of network congestion unless efficient multiple destination routing (MDR) algorithms are used. Current MDR algorithms using link cost based on bandwidth usage or distance to compute the minimum cost routing tree do not take into account the vast amount of information collected by intelligent network (IN) databases. In this paper, we propose a multicast routing algorithm which modifies the way the cost is calculated by using the predicted traffic statistics collected by IN databases. We also show that the traffic handling characteristics vastly improved over conventional MDR algorithms.

To provide personal, intelligent, and multimedia services through a mobile communications network, a Mobile Service Control Point (M-SCP) was developed, which performs both the location register and service control functions. The M-SCP was constructed on a common platform to allow quick introduction of new services. Software techniques to reduce the frequency of process-switching, assign the highest priority to real-time tasks, and operate a multiple-CPU structure provide faster real-time processing. This is confirmed by computer simulation and research in the field.

The throughput performance of a slotted, non-persistent Idle-Signal Casting Multiple Access (ICMA) protocol under the effects of various combinations of Rayleigh fading, lognormal shadowing, and spatial distribution of mobile users is studied. The opposing effects of propagation impairments on the performance of the protocol through simultaneously increasing the probability of receiver capture and attenuation of the received signal power level are demonstrated.

In this paper, we propose DSVMA (Data Steal into Voice Multiple Access) scheme for integration of voice and data in wireless information networks. By using speech activity detectors and effective downstream control signals, DSVMA enables data terminals to transmit multi-packet messages when voice terminals are in silent periods. The S-G (throughput versus offered load) performance of the DSVMA system and the blocking probabilities of both the second generation systems and the DSVMA systems are evaluated by the static analysis. A dynamic analysis of a system with finite number of terminals is also presented using an approximate Markov analysis method. Some numerical examples are given in the paper. As a result, it is shown that DSVMA can improve the channel utility efficiency of a circuit-switched TDMA (Time Division Multiple Access) wireless communication system and is directly applicable for second generation wireless information systems.

This paper discusses the potential benefits of H0-based ISDN. By providing widespread H0 (384kb/s) call services, much like today's telephone systems, it should be possible to bring multimedia public communication services to the general public. Small distributed switching systems and reuse of existing metallic loops will contribute to its economical construction and rapid expansion into general use. System-on-a-chip technology, expected to appear soon, will make the necessary small switching systems cost-effective. The short loops resulting from the distribution will allow Hl-band (1.5/2.0Mb/s) metallic loops to be used. In terms of function, the distributed switching nodes will permit node-by-node customization, which will make switching systems simple and flexible. The proposed node-by-node customization will allow subscribers in each different node to choose they prefer. H0-based ISDN will provide various type of nodes, supporting various types of subscriber loops, whereas present public networks enforce standardized subscriber loops. Furthermore, when customized subscriber switching systems accommodate star-LAN interfaces as subscriber loops, the public network will be able to provide LAN services in a closed area as part of the public network services. The LAN users will be able to converse with ordinary residential subscribers through H0-calls. This will lead to the integration of LANs and public networks.

A new kind of optical local area network (LAN), using a demand assign wavelength division multiple access (DA-WDMA) scheme, has been proposed. The proposed LAN consists of two parts; an ordinary standardized LAN and an overlaid network using wavelength division (WD) channels. The proposed network can provide bit-rate independent communication channels on the ordinary LAN without limiting the capacities for the other channels. It also exhibits upgrade possibilities from present standardized networks. An access controller, which consists of software in addition to the ordinary LAN controller, a digital signal processor (DSP) etc., was developed for DA-WDMA control. The network node operation has been demonstrated using guided-wave acousto-optic (AO) mode converters as a tunable wavelength add-drop multiplexer (ADM).

This paper proposes a high-speed VP bandwidth control scheme for ATM networks that utilizes a distributed control mechanism. First, the characteristics of VPs are compared with those of digital paths in STM networks. A distributed control scheme is adopted for rapid control. The basic elements and the necessary distributed function, the control algorithm, and the message transmission mechanism, are elucidated. The bandwidth alteration time with the proposed algorithm is estimated by considering network element processing and queuing delay. The proposed VP bandwidth control scheme can be applied to both public networks and leased line services. Finally, this paper focuses on its application to leased line services, and discusses the resource reduction effects of the proposed scheme.

In our preceding paper, I-ISMA (Idle Signal Multiple Access for Integrated services), a combination of ISMA and time reservation technique, was proposed to transmit an integrated voice and data traffic in third generation wireless communication networks. There, the channel capacity of I-ISMA was evaluated by the static analysis. To fully estimate performance of contention-based channel access protocols, however, we also need dynamic analysis to evaluate stability, delay, etc. Particularly, in systems concerning real-time voice transmission, delay is one of the most important performance measures. A six-mode model to describe an I-ISMA system is set up. With some assumptions for simplification, the dynamic behavior of the system is approximated by a Markov process so that the EPA (Equilibrium Point Analysis), a fluid approximation method, can be applied to the analysis. Then, numerical and simulation results are obtained for some examples. By means of the same analysis method and under the same conditions, the performance of PRMA is evaluated and compared briefly with that of I-ISMA.

Corresponding to the development of B-ISDN, integrated services for data, voice, etc. are imperatively required for the so called third generation wireless communication networks. In this paper, I-ISMA (Idle Signal Multiple Access for Integrated services) is proposed to transmit integrated voice and data traffic from dispersed terminals to a base station. In the system, data packets and the first packets of talkspurts of conversational speeches are transmitted using ISMA protocol over a shared channel while subsequent packets of talkspurts are sent with time reservation technique. The channel capacity of I-ISMA is evaluated and compared with that of PRMA. The region in which I-ISMA has larger capacity than PRMA is figured out. Generally speaking, I-ISMA has larger capacity than PRMA when the duration for transmitting and detecting an idle signal is not too long and the channel is not too congested by the reserved voice transmissions. When we concern real time voice transmission, delay is one of the most important performance measures. Only is a qualitative discussion on delay performance given here. The quantitative evaluation is obtained by the dynamic analysis in our succeeding paper.

Markov chain models are used to derive the average stationary throughput and delay performance of Idle-Signal Casting Multiple Access (ICMA), with and without Failure Detection (/FD), protocols which are suitable for use in mobile packet radio local area networks, where propagation impairments are prevalent. The models consider the effects of pure Rayleigh fading on channel access and packet transmission. Numerical results, validated by computer simulations, are obtained that enable a quantitative study of the performance of the protocols. It is found that the performance of the ICMA/FD protocol is affected more significantly by fading on the base-to-mobile channel than is the performance of the ICMA protocol. In addition, performance improves with larger packet sizes eventhough such packets are more vulnerable to failure due to fading.

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.

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.