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Conventionally, the problem of capacity-flow assignment in packet switching computer networks has been considered mainly for the purpose of improving the network performance and the effective usage of the resources in the networks. The problem of the fair allocation of the network resources to each user (fairness) has been investigated little. Because one of the most important purposes of the computer network is that a plural number are arranged dispersively in the network, we must make sufficient study of fairness in the problem of the flow assignment of the packet and the problem of the distribution of the line capacity. However, no clear definition of the fair allocation has been proposed to data. In this paper, we adopt the "power" (function) as the performance measure and discuss the problem of the network capacity-flow assignment in which maximized network power ensures effective usage of networks, and equalized power for each user realizes the fairness. The results of this study provide a new guideline to solve the problem of distributing the resources effectively and fairly. The method in this paper can be discussed generally without limitation of the network topology. However, since the concrete numerical calculation is so complicated, we take loop networks for the object of the analysis. [ABSTRACT FROM AUTHOR]

In the video and speech transmission using packets, the media synchronization control is applied, especially in the system for real-time transmission. In the system containing the media synchronization, the allowable time delay is defined, and the packet after this delay limit is discarded. In protocols such as ARQ (automated repeat request), where the packet retransmission is executed for error correction, if the delay or the error rate of the channel is large, the packet delay is greatly increased when an error occurs. The packet is discarded when the packet delay exceeds the allowable delay, which produces degradation of quality. This paper proposes an adaptive protocol relay system considering the allowable delay, in order to cope with this problem. In the adaptive protocol relay system, the protocol relay is executed at the relay node, when a certain condition for each packet is satisfied. By reducing the time spent in the packet retransmission due to the error, the number of packets that do not arrive at the receiving terminal within the allowable delay is reduced. This paper examines the effect of the protocol relay on the packet discard rate, when the proposed method is used. The condition for the adaptive control at the relay node is also discussed. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 88(8): 50–59, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10031 [ABSTRACT FROM AUTHOR]

FRN (flexible radio network) is an ad hoc network product that can realize a wide-area data collection network. This paper discusses a performance improvement method when TCP is applied to the transport layer in FRN. It is shown that even when the terminal is not moving as in FRN, the throughput can be improved in radio link failure due to noise or obstacles by combining ELFN (explicit link failure notification), which is proposed as a method for TCP performance improvement in ad hoc networks, with routing information. The proposed method is evaluated by simulation in this paper. A further objective is to reduce the performance degradation due to packet collision. The delayed ACK and the ACK retransmission with priority in the case of packet collisions are analyzed and the effectiveness of the method is demonstrated. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 87(11): 99–107, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10134 [ABSTRACT FROM AUTHOR]

In cellular communication systems, every call is carried through a base station, and when the communication path to the base station is obstructed by buildings or other obstacles, the communication may not be established. It is expected that ad hoc relaying of such an obstructed call will increase the performance of the network as a whole. This paper discusses how to theoretically analyze the communication traffic performance of a cellular system with ad hoc networking and shows that the network performance can be increased even with relatively small range of relaying capability of terminals. This paper also shows by simulation that multihop relaying does not give significantly greater performance improvement than that by two-hop relaying. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 86(11): 1–10, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10135 [ABSTRACT FROM AUTHOR]

Self-healing has been investigated as a technique for efficient path failure restoration in VP-based networks. This paper discusses self-healing in networks with multiple reliability classes, and evaluates restoration performance of different failure recovery procedures that introduce priority orders in the failure recovery process based on the reliability class for each failed path. The authors present a composite-type restoration procedure in which preplanned-type and flooding-type procedures are combined. This paper also presents a bumping added-type restoration procedure. The authors evaluated the performance of the above proposed techniques assuming various failure modes. The simulation results revealed that the proposed methods show improved performance. In addition, we propose an enhanced technique that introduces a subsidiary priority criterion in the paths belonging to the same reliability class, in order to increase the number of paths to be recovered. © 2001 Scripta Technica, Electron Comm Jpn Pt 1, 84(7): 1–17, 2001 [ABSTRACT FROM AUTHOR]

This paper presents a new construction of the distributively controlled homogeneous self-routing space switch network, which is composed of single-kind 2 x 2 unit modules. The feature of the proposed method is that the order is defined among the destination addresses according to their binary representation by the bitwise comparison. The bionic sort is executed based on the order. In the conventional space switch network composed of a single kind of module, the total number of unit modules is of the order of N2 at the minimum, where N is the number of I/O ports. In the space switch network proposed in this paper, the total number of unit modules is of the order of (N log3N). The total number of unit modules in the proposed method is equal to that of the conventional method for N>16. When N = 1024, for example, the total number of unit modules in the proposed method is approximately one-sixth that of the conventional method. Another point is that the processing speed within the 2 x 2 unit module is independent of N, as in the conventional method, and is small. This is advantageous in the high-speed operation. [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 ubiquitous environment, in which a tremendous number of objects, such as physical devices, multimedia contents, and software components, exist everywhere. In order to provide and utilize various services transparently in such an environment, there must be a service synthesis technique that can dynamically construct services by mutually connecting objects. The realization of service synthesis requires a position-independent naming service which can securely and transparently call from the network the functions (functional objects) that act as the service component. The authors designed an interface-based naming service which can perform transparent calling of functional objects on the basis of interface information on the objects. The interface-based naming service not only can call functional objects efficiently from the network, but also can provide several useful features of service synthesis support: when two functional objects are interconnected, the required supplementary functional object is automatically determined, and when a function cannot be found, that function is realized equivalently by using multiple functional objects. This paper describes the architecture of the interface-based naming service, together with its algorithm. The approach is evaluated by simulation and it is shown that the interface-based naming service is a useful system in a ubiquitous environment with an increased number of objects and an enlarged network scale. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 88(3): 63–75, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10177 [ABSTRACT FROM AUTHOR]

In reliable multicast, the same information is transmitted to specified receiving nodes while maintaining reliability, with the information rate aligned to the receiving node with the lowest throughput by end-to-end control. However, the consideration of the receiving node with the lowest throughput produces the problem that other receiving nodes belonging to the same multicast group are limited to a considerably lower throughput than their own reception limits. This is a problem inherent to multicast communications, where the receiving nodes generally differ in congestion status. This fairness problem for receiving nodes is called intrasession fairness. In reliable multicast, it is impossible to realize intrasession fairness by using only end-to-end control. The purpose of this paper is to realize intrasession fairness by a scheme in which servers independently apply congestion control to the downlink by placing servers in the network and providing network support. The proposed scheme is evaluated by simulation. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 88(2): 61–70, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10125 [ABSTRACT FROM AUTHOR]

Channel efficiency can be improved by transmission that integrates voice and the data. When this idea is applied to multicellular systems, channels must be assigned so that the intercellular interference power is minimized. Furthermore, channels must be provided with priority for voice calls, since real-time transmission is required. As a system that considers these requirements, this paper investigates a method of reducing intercellular interference in a circuit switching system with channel segregation in which data are transmitted with priority, using the silent period of voice calls. It is shown by computer simulation that the proposed method can improve performance characteristics such as the interference rate, blocking rate, forced termination rate, and hand-off rate. [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]

With an increase in demand for traffic and in modes of network services, it is required that the backbone network be able to cope flexibly with changes in demand for traffic with high speed and reliability. ATM ring networks can enhance the path capacity by unified processing of various service modes. In addition, due to the low transmission path cost attained by network topology, economical construction of a transport network is possible. In this paper, it is recognized that the path capacity becomes larger as the path length of the protection VP becomes longer. Then, a VP switching method for an ATM ring network, BPSR/PF (Bidirectional Path-Switched Ring/Path-Failure detection), is proposed in which the path length of the protection VP becomes the shortest. With regard to the items for evaluation, such as the maximum required link capacity, switching time, required number of link VPs, data sustaining number by VP pairs, and path survival rate, the proposed method is compared quantitatively with the conventional ATM ring network VP switching method. It is made clear that the proposed ATM ring network VP switching method BPSR/PF is superior. © 2002 Scripta Technica, Electron Comm Jpn Pt 1, 85(5): 57–70, 2002 [ABSTRACT FROM AUTHOR]

This paper describes a failure-resistant network using diversity coding (DC). The technique uses error correcting codes constructed across logically independent virtual paths in ATM networks. This system requires some restrictions on the arrangement of source/destination node pairs in order to avoid increase in parity path capacity. In this paper, the author proposes a network structure where the active paths are divided into several segment networks. This technique enables us to relax constraints on the DC network formation. The author also presents a mechanism for efficient reduction of spare path capacity by partial sharing of spare paths among segment networks. In addition, the author analyzes the performance of spare path capacity reduction and the performance in terms of failure restoration time. © 2000 Scripta Technica, Electron Comm Jpn Pt 1, 83(12): 1–14, 2000 [ABSTRACT FROM AUTHOR]

An efficient scheme for establishing a multicast route in packet networks is constructing a tree-shaped path in which multicast packets are replicated at specific branching nodes, called “copy nodes,” by the packet copy function and are forwarded to downstream nodes. This scheme leads to the efficient usage of network resources, but needs multicast routing control. To realize rapid multicast routing, it is suitable and realistic to restrict the number of copy nodes in calculating the multicast route. However, even in this approach we need to adequately select a limit on the number of copy nodes which satisfies the conditions of both priority for multicasting and geographical distribution. In this paper, a new copy-node allocation algorithm is proposed. This heuristic algorithm makes some “copy node functions” move on the network map sequentially according to attraction and repulsion determined by several conditions and finally makes them converge on the switching nodes suitable as copy nodes. In addition, the performance of this algorithm is evaluated by a computer simulation. This algorithm is applicable to various other server assignment problems. © 1999 Scripta Technica, Electron Comm Jpn Pt 1, 83(1): 87–96, 2000 [ABSTRACT FROM AUTHOR]

Document caching at proxy servers is widely adopted in today's Internet, since it is one of the most efficient ways to save the required capacity of the Internet access line. Hence, for efficient design of an Internet access network, it is essential to know how the characteristics of Internet traffic are affected by document caching. In this paper, we first investigate the effect of document caching on the World-Wide Web (WWW) traffic characteristics using the log data gathered at the proxy server. Based on this analytical result, we next build an M/G[log-normal]/1/PS queuing model to discuss a design methodology for an Internet access network. © 1999 Scripta Technica, Electron Comm Jpn Pt 1, 82(7): 25–34, 1999 [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]

To achieve efficient and highly reliable network operation requires a mechanism to monitor and control the states of remote a mechanism to monitor and control the states of remote terminals (nodes) automatically by control operation system (OpS). This paper is an attempt to obtain such a mechanism. Operation and maintenance (OAM) cell is defined that contains the connection information for remote monitoring and the control path to be set. By successively transferring OAM cells according to the connection information, remote monitoring and control virtual paths (VPs) can be obtained autonomously. Based on the proposed idea, this paper discusses the structure of the OAM cell, the functions to be provided in the cross-connect system, and transmission algorithms for the OAM cell. The proposed system well utilizes the unique feature of the ATM cell. The cell routing information (VPI) is transferred within itself. [ABSTRACT FROM AUTHOR]

Telecommunications networks are designed and managed based on statistical traffic data including basic traffic obtained at peak traffic hours. Network links are also designed based on the basis traffic. The network is designed so that the quality of services (network blocking) is at a certain level when basic traffic is applied to the network and the network is managed so that quality of services is maintained for the basic traffic. This conventional method is universal and simple because the required data for design and management are very small. On the other hand, recent telecommunication systems have the ability to measure and collect detailed traffic data. This paper proposes a new network design method in which time-series traffic data rather than statistical traffic data is used for network design. Combining the time-series traffic data and the tariff data for each hour, we can design a network based on total revenue. Unlike the conventional method, the proposed method makes it possible to estimate revenues, costs, and damages. Using time-series traffic data obtained for every hour of the year, a method is devised that optimizes the number of circuits (i.e., for optimizing network blocking) and thus maximizes profits. This paper also presents a new methodology for traffic data and a new network structure that effectively uses the proposed method.

This paper discusses the multipath allocation problem in the communication network and proposes a multipath allocation algorithm suited to the distributed environment. The multipath allocation problem is a problem where multiple paths arc to be allocated without a conflict. This problem has been difficult to solve in the distributed environment since relations among various routes must be considered. The algorithm proposed in this paper solves this point by realizing the asynchronous path search. In this paper, the basic characteristics of the proposed algorithm are described through the comparison between the proposed and the traditional algorithms for the case of serial computation under the centralized control. Then the proposed algorithm is applied to several kinds of distributed cooperative multiagent models, and it is demonstrated that the proposed model can easily be applied to the distributed environment. A simulation is executed actually using several workstations, and it is shown that the proposed algorithm is desirable as a solution method for the multipath allocation problem m the distributed environment. [ABSTRACT FROM AUTHOR]

The Virtual Path concept has several beneficial features to construct an economical and efficient Asynchronous Transfer Mode (ATM) network. The Virtual Path capacity control technique can increase network flexibility and improve transmission efficiency through statistical sharing of facilities. This paper presents an adaptive virtual path capacity control method which maximizes transmission efficiency under the condition of limited processing load. Evaluation shows that the proposed method successfully im- proves network transmission efficiency with a slight increase in processing load compared to the fixed virtual path capacity scheme. [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]

With the advent of the information age. congestion of communication networks due to natural and social phenomena occurs more easily, and coping with this overload traffic is necessary. In this paper, a mathematical model for evaluating overload traffic is presented, and methods for using this model to evaluate overload traffic are shown. Since the state of a congested network varies with time, the transient behavior of the traffic needs to be analyzed. However, it is impossible to conduct the analysis by means of existing methods because of the enormous amount of computations required. Therefore, by regarding the traffic in a congested network as deterministic, an approximate solution, by means of deterministic equations, is proposed in this paper. The transient analysis of the throughput and overflow rate of networks with finite resources can be evaluated by means of this approximate solution. In accordance with the proposed model, the methods to compute the call blocking probability, which is used to evaluate the network robustness against temporary overload, and the throughput in circuit-switched networks are shown. It is possible to compute the number of resources (i.e., buffers) which is appropriate for a given blocking probability considering the robustness of the network against congestion. The maximum throughput of the network can also be evaluated with transient analysis of the throughput. [ABSTRACT FROM AUTHOR]

With the increased use of personal computers (PCs) in communication environments have come the expectation of reliable communication protocols. This paper describes the implementation details of two very similar packet communication protocol interfaces on PCs. One is the X.25 interface for packet-switched public data networks, and the other is the X.32 interface for public-switched telephone networks. The implementation of these protocols is performed with consideration of the future introduction of OSI higher-layer protocols. This paper describes also that the addition of the assigning mechanism of level 2 address to the Rec. X.32 enables powerful personal computer communication through public-switched telephone networks. Furthermore, the results of the performance evaluation are described which show the high throughput of more than 90 percent on various speeds of the physical circuit. [ABSTRACT FROM AUTHOR]

The cell-wise control of a communication network, such as dynamic routing, is a very complex problem, although the extent of difficulty depends on the control target. Especially, the optimization of a communication network in a multiservice environment contains the difficulty of the choice of the state variables. the analytical method is limited in handling such a problem. This paper discussed the application of the fuzzy system by extending the range of search for the optimal solution for the communication network control from the analytic single expression to the description by the combination of several rules. The usefulness of the idea is demonstrated by a simulation. Then aiming at the full utilization of the descriptive power of the fuzzy system, the modularization of the control system corresponding to the data abstract level, is proposed. Further, this paper considers the revenue maximization control of the communication network as a example, and the usefulness of the hierarchical fuzzy system is demonstrated through a simulation. By applying the proposed method, it is made possible to cope with a range of problems not previously handled by the analytical method. [ABSTRACT FROM AUTHOR]

The mixed media network including the terrestrial and the satellite communication links is considered, aiming at the realization of a highly reliable information network. This paper proposes a method of physical level switching control in the high- speed multiplexer between the terrestrial and the satellite links. The following three aspects of the physical level switching control method are described. The high-speed digital network management system instructs the link switching to the high-speed multiplexer according to detected faults. The high-speed multiplexer has a function to handle equally the high-speed digital link and the satellite link. The Earth station has the interface function which enables the high-speed multiplexer to switch the master clock. This paper outlines the switching control method at the physical level using the high-speed multiplexer in the transmission network. It discusses also the technical issues as well as their solutions in the implementation, such as the loss of paths for managing information, management of intermittent faults and switching of the master clock. The link switching experiments actually are made between the earth and the satellite in the "pilot study for use of satellite," underwritten by the Mm. Post & Telecomm. The time from the fault detection to the complete link switching in the proposed network management system is within 30 s, indicating the effectiveness of the proposed switching control method. [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]

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]

This paper considers the polling system with the single message buffer at the station. Most of the past studies on the polling system have been concerned primarily with the mean delay and the mean throughput. To analyze the multihop system which appears in the interconnection of information networks, however, the interdeparture process in the system must be analyzed. This is because the output process in each system is related closely to the input process in other systems. From such a viewpoint, this paper analyzes the interdeparture process in the polling system with the single message buffer at the station, and presents the Laplace-Stieltjes transform of the interdeparture time distribution. Using the obtained result, numerical examples are presented, indicating some interesting results concerning the coefficient of variation of the interdeparture time distribution. [ABSTRACT FROM AUTHOR]

The network interconnection technique is one of the keys for expansion of a local area network. This paper gives a performance evaluation of the multiring local area network system which consists of several token ring networks interconnected by using this technique. The result shows that this network system is better than the single-ring network system, which is an ordinary token ring network, at heavy load operation. This paper also gives a performance comparison among three network interconnection topologies, ‘MESH connection,’ ‘RING connection,’ and ‘BUS connection,’ and the optimum regions for these network interconnection topologies are suggested. [ABSTRACT FROM AUTHOR]

The token ring system is one of the typical LAN systems. In LAN, several kinds of traffic with different properties occur in bursts. The analysis has not been made for the case where the calls arrive in batch following the general distribution (GI(X) input). In the actual system, the buffer capacity is finite, and the analysis must consider such a property. This paper considers the asymmetric (i.e., the arrival process and the service time distribution are different for different stations) and single-limitation (i. e., when the token arrives, only the first call is served) GI(X)/G/1 token system (where the suffix [X] indicates the batch arrival). The first half of the paper discusses the case where the buffer capacity of the station is infinite. The diffusion approximation is proposed using the elementary return as the boundary condition, and the quantitative expressions are given for the call distribution and mean waiting time for each station. The second half of the paper discusses the case where the buffer capacity in each station is finite. By combining the diffusion approximation used in the first half and the iteration, the blocking probability and the mean waiting time are derived quantitatively. For each case, simulation was made with the result indicating that the proposed approximation is reasonable. [ABSTRACT FROM AUTHOR]

This paper describes the traffic analysis of TSTST type building block speech paths. The main component of this configuration is the switch module. It consists of a small-sized T-switch and part of an S-switch. TSTST configuration is used to make a large capacity switching system with these switch modules. Conventionally, TST configuration was used mainly for speech paths. However, the number of stages must be increased in order to make a large capacity switching system using switch networks with small channel multiplicity. The traffic handling characteristics of TSTST are almost the same as those of TST with the same size T-switch and the same total number of channels, and this configuration can be applied to a wide range of speech path sizes. Furthermore, flexibility and reliability can be improved by adding alternative links to the fundamental TSTST configuration, and thereby link efficiency can be improved by a few percent compared with the configuration without alternative links. [ABSTRACT FROM AUTHOR]

In a packet switching network it is important to route traffic flow so as to minimize message delay. This problem is formulated as a multicommodity flow problem with a nonlinear objective function and it is solved by the steepest descent method. It is also necessary to control the traffic demand at each node so that the total message delay is below a certain level. This paper presents an algorithm to obtain the region vectors consisting of traffic demand at each node which can be transmitted simultaneously under the given delay time. Despite its importance this region has not been adequately considered. A nonlinear T-cut condition is introduced by which the region is obtained. This paper is a non-linear extension of the authors' recent work. [ABSTRACT FROM AUTHOR]

Deflection routing is considered as a method to avoid packet collisions in an optical packet switching system. Deflection routing is a method of avoiding collision by link substitution, in which the colliding packet is sent to an idle link. Thus, the load on the congested part of the network is distributed and the bandwidth can be utilized effectively. When excessively frequent deflections are performed, however, the load on the whole network is increased, which produces an adverse effect. Another problem is that sequence errors can occur in the same flow due to differences of transmission delay, since packets are sent to different routes. From these points of view, this paper proposes a deflection control method which can remedy the above problems, and applies the proposed method to an optical packet switching system. In the proposed control method, class separation is performed with respect to sequence error, and deflection is permitted to packets of nonpriority class. This improves performance by large-scale grouping. In order to alleviate the increase in load on the whole network, a control algorithm which can deal with changes of the link utilization and the number of traversed hops is devised. The effectiveness of the approach is shown by computer simulation. © 2007 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 90(9): 67– 77, 2007; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.20312 [ABSTRACT FROM AUTHOR]

IEEE802.11 WLAN (Wireless LAN) has been widely used in enterprises and public spaces such as airports. In these large WLAN networks, RRM (Radio Resource Management) is necessary for efficient use of radio resources, and also load balancing among APs (Access Point) has to be taken into account. Since, in IEEE802.11, STA (Station) has the right to select an AP with which it will associate, the AP selection mechanism implemented in STA is important for load balancing. This paper proposes an association mechanism called HRFA (High-Rate First Association) in order to achieve load balancing and efficient use of radio resources. Furthermore, IEEE802.11e was developed by Task Group E of the IEEE802.11 working group and specified a MAC protocol to provide QoS in the MAC layer and to support real-time traffic over WLAN. In order for WLAN to be more widely used, the functionalities of IEEE802.11e must be provided in WLAN devices. Our proposed HRFA can be applied to WLANs compliant with IEEE802.11e. Since the HRFA can be implemented without any modifications in the IEEE802.11 and IEEE802.11e standards, it is useful from the implementation-cost and compatibility point of views. Simulation results show that HRFA can efficiently utilize radio resources and can also achieve load balancing in IEEE802.11e WLAN networks. © 2006 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 90(4): 35–45, 2007; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.20347 [ABSTRACT FROM AUTHOR]

Mobile ad hoc networks attract tremendous attention as one form of future network configuration. Among wireless LAN technologies, the IEEE 802.11MAC standard can be considered as the most prominent technology deployed in ad hoc networks. For realization of mobile ad hoc networks, providing QoS support in ad hoc networks is an important task. Among numerous parameters of QoS, per-flow fairness is an important and problematic issue in ad hoc networks. In the traditional wireline networks, per-flow unfairness occurs mainly due to the link layer. However, in mobile ad hoc networks, in addition to the link layer, the MAC layer has significant impact on per-flow fairness problems. In particular, although IEEE 802.11MAC provides per-node fair bandwidth allocation, in order to achieve per-flow fairness, a method that provides per-flow fair bandwidth allocation is required. In this paper, a new scheme is proposed that enables a fair bandwidth allocation to each flow. In addition to fairness, the proposed scheme provides a positive impact on the network in terms of total performance and medium utility. The performance of the proposed scheme is investigated with an ns-2 network simulator. © 2006 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 89(8): 37–49, 2006; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.20250 [ABSTRACT FROM AUTHOR]

Medium- and small-scale business organizations and schools are sometimes unable to obtain a sufficient number of network operators to achieve an adequate initial network configuration with IP routers, or to perform reconfiguration of the IP routers when the network configuration is dynamically modified, as in the case of addition and removal of IP routers. The purpose of this paper is to reduce the management workload in such an environment. An automatic configuration method for IP routers to deal with dynamic modification of the network is proposed. When the ARCP (automatic router configuration protocol), which is the conventional method for automatic configuration of IP routers, is run using the management server, difficulty arises in the modification of the IP router configuration information when an IP router is removed or three or more IP routers are connected to the same segment. A further problem is that the reduction of the management workload is less than might be expected, since there are many points to be set in the management server before starting the operation. An attempt is made to remedy these problems by proposing a new extended ARCP. The conventional method and the extended ARCP are compared theoretically and by simulation on a network with the assumed environment. The extended ARCP is shown to remedy the above problems and to reduce the number of points to be set in the management server by 30% compared to the conventional method. © 2006 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 89(5): 15–27, 2006; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.20247 [ABSTRACT FROM AUTHOR]

This paper proposes a method of generating network traffic patterns needed in stress testing of Web sites. This method mainly consists of two stages: in the first stage, from a positive root signal, a signal pattern with multifractality and a long-tail distribution is synthesized using wavelets. In the second stage, the short-term distribution characteristics of the pattern generated at the first stage are adjusted by decomposition into low-frequency and high-frequency components by nonlinear filtering and changing the ratios of their two components in order to obtain the appropriate short-term distribution, so as to finally obtain network traffic that also has an appropriate long-range dependence. Simulation results are also presented for validation of this proposed method. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 89(2): 47–56, 2006; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.20223 [ABSTRACT FROM AUTHOR]

In this paper, we propose and discuss a method for achieving expandability and portability through the protocol specifications and implementation of a virtual network. Technologies based on graph abstractions such as link-state routing and broadcast techniques do not fundamentally depend on specific network technologies. Today, in order for these technologies to directly use the identifiers and communication protocols defined by a specific network technology, portability to other network technologies and adaptability to a heterogeneous network environment are degraded. The portability and adaptability to a heterogeneous network environment are improved by designing and implementing these technologies for a virtual network and emulating the virtual network on various networks. In this paper, to construct a virtual network, we explain a method for emulating the virtual datalink ANCIF proposing the same media access control protocol and protocol development using ANCIF. In practice, we applied and implemented ANCIF on various real networks, and measured the protocol overhead resulting from the virtual implementation. The result was extremely small overhead and almost no effect on performance. Finally, we discuss the portability to other network technologies and the adaptability to a heterogeneous network environment which are obtained by applying ANCIF. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 88(8): 32–40, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10204 [ABSTRACT FROM AUTHOR]

The communications environment is currently well organized. Still, it is not easy to handle requirements for urgent communications in a disaster. In a large-scale disaster, it is highly probable that communications functions will be hampered, making access to communication links itself difficult. There is a need for studies directed at assuring robustness and reliability in the emergency communications system. This paper proposes an access protocol for wireless networks in which the centralized control hierarchical network and the horizontal connection ad hoc network are combined. The routing method in which a terminal that cannot directly communicate with the base station (BS) sends urgent data to the BS through neighboring terminals, and an access control protocol for multihopping, are described. In the proposed scheme, the terminal monitors communications in neighboring nodes and dynamically generates a route based on the number of hops to the BS. Evaluations show that the percentage of terminals that can reach the BS by multihopping involving a few hops is significantly improved. It is also shown that the terminal knows the number of hops to the BS in the proposed method, which makes it possible to predict the throughput and the delay time, which is a property suited to urgent communications. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 88(7): 52–64, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.20141 [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]

Recently, more research attention has been focused on LEO (Low Earth Orbit) satellite networks connected to ISL (Inter Satellite Link). If terrestrial and LEO satellite networks are seamlessly interconnected by TCP/IP protocols, a global network that covers the whole earth can be integrated into the Internet. However, handover and rerouting issues in LEO satellite networks cause delay variance, and the delay variance can degrade performance of TCP, originally designed for wired networks, because of unnecessary retransmission. In this paper, we propose a method for avoiding the unnecessary retransmission by using the TTL (Time To Live) field in IP headers. Our simulation results show how our mechanism is effectively avoiding unnecessary retransmissions. [ABSTRACT FROM AUTHOR]

Communication between an IPv6 host and an IPv4 host requires a name server that makes a mapping between an IPv4 address and an IPv6 address, and a translator that performs IP packet conversion. In this paper, we propose an IPv4­IPv6 bidirectional communication system, implement our proposed method, and present the results of performance evaluation. Our system can provide IPv4 hosts with connectivity to more IPv6 hosts in spite of pooling fewer IPv4 addresses. It is realized by adding a source IPv4 address and a port number to an address mapping with cooperation between a name server and a translator. [ABSTRACT FROM AUTHOR]

In mobile ad hoc networks, performance is degraded by link breakage due to node mobility and packet collision due to network congestion. Therefore, it is very attractive to investigate routing protocols which use a routing metric to discover long-lived routes or to disperse network congestion instead of using hopwise shortest routes. In this paper, we examine three route selection schemes in a routing protocol which uses the link expiration time as a routing criterion. Furthermore, we present the Busy Node Avoidance Routing (BNAR) protocol that employs a busy rate representing the nodal load as a new route selection metric. Computer simulation results show the fundamental performance and effectiveness of routes based on the link state and node state. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 87(2): 68–78, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10137 [ABSTRACT FROM AUTHOR]

This paper is concerned with the realization of a ubiquitous computing network designed to improve the reachability of communications and to eliminate complex setup and manipulation related to communications. The new communication system EAPEC (environment-adaptive personal communications) is proposed. The communications context of the user, which changes from time to time, is automatically acquired. Based on that information, the communications terminal and the communications service/medium that are optimal for the receiver to receive the message are automatically selected. Then, according to the selection, the communications service or the communications medium is automatically converted. It is shown in this paper that EAPEC can be realized under the 3GPP (third-generation partnership project) IP network architecture. Placement of five functional modules in the network and the automatic acquisition procedure for communications context are described. As a means of performance evaluation, the call setup delay is evaluated by simulation, and guidelines for network design are obtained. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 87(1): 34–47, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10087 [ABSTRACT FROM AUTHOR]

The author developed an RMI (referred to here as xRMI) that includes a code generation function for object serialization and confirmed its effectiveness. The distinguishing characteristics of xRMI are that it combines runtime code generation for object serialization and migration of generated code while maintaining the ease of use of the Java RMI. This enables the time required for the second and subsequent client-side serializations to be reduced and high-speed deserialization to be done at the server side by migrating the code for deserialization, which is generated at the client side, to the server side. As a result, xRMI runs faster than standard Java RMI, and a significant performance improvement can be expected for network applications. This paper presents the background for the research and describes the basic design and implementation of xRMI, as well as experiments for confirming xRMI performance. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 86(10): 79–88, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.1178 [ABSTRACT FROM AUTHOR]

IP multicast over ATM is considered interesting as a technique to realize multicast communication in ATM network. Proposed methods to realize IP multicast over ATM are the server method and the mesh method. The server method has the problem that a certain delay is required in reconstructing AAL5 SDU in the server. The mesh method has the problem that the point-to-multipoint VC must be set in the receiver group from all senders. This paper proposes the multi VC data server method as follows, as a new method to solve these problems. Multiple point-to-multipoint VC are prepared in the server. When multiple data are transmitted in parallel, different VC are used to reduce greatly the delay produced in the server method. The performances of the multi VC server method, the server method, and the mesh method are compared, and it is shown that the multi VC server method is effective in the sense that the data are transmitted with the shortest data transmission delay. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 86(6): 11–18, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10070 [ABSTRACT FROM AUTHOR]

Mobile IP has gained attention as a technology that can provide mobility to universal users independently of the access network. The introduction of Mobile IP in mobile core networks is currently under investigation. This paper discusses route optimization methods when Mobile IP technology is applied to mobile core networks. More specifically, it discusses means of resolving the “Triangle Routing” problem, which occurs when communications from a correspondent node to a mobile node must be routed through a Home Agent (HA). This paper first describes the conventional route optimization method being investigated by the Internet Engineering Task Force (IETF) and indicates problems with that method. It then proposes a new Mobile IP Border Gateway (MBG) method for solving the problems of the conventional method. The MBG method enables packets that arrive from another network to be directly received at the mobile node by having a Border Gateway router maintain Binding Information consisting of the care-of address and home address that were originally being maintained by the Home Agent for the communicating node. This paper also shows that introducing the MBG method enables the mean route length within the network to be shortened and the traffic volume routed through the Home Agent to be reduced. Finally, it shows methods of reducing the burden of registering Binding Information at the Mobile IP Border Gateway and indicates the applicability of the MBG method to large-scale mobile communications networks. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 86(2): 31–41, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.1151 [ABSTRACT FROM AUTHOR]

With progress in intranet/Internet environments and computer telephony (CT) technology, it is expected that intelligent services using IP network will soon be developed. In such services, high-grade information processing facilities, including terminals, will be actively used, making possible high performance of the user interface and highly sophisticated user-side processing. This paper proposes a new architecture in which an interworking unit (IW-Unit) is used in both the user-side CT system and the net-side IN system, with examples of added-value services. The proposed method is a practical high-grade service for users having CT systems. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 85(11): 106–114, 2002; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.1143 [ABSTRACT FROM AUTHOR]

AntNet was reported to exhibit an undesirable effect called “Routing Lock.” If a link is congested or down, a node can select another link easily with the ability to cancel the effect quickly. As means of canceling the effect, the decay method and the negative feedback method were proposed. However, the decay method is not effective in balancing the cancellation of the effect with the learning speed of network traffic statistics. Also the negative feedback method requires ants to move on at least one congested link so that it takes a long time to cancel the effect. In this paper, we propose an improvement algorithm, BntNetL, which cancels the effect indirectly by giving ants the ability to update routing tables bidirectionally. BntNetL shows better performance than its competitors in simulations. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 85(9): 31–41, 2002; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.1123 [ABSTRACT FROM AUTHOR]

We have already proposed DONA-α (distributed object-oriented network architecture) as a candidate architecture for future telecommunications networks. A node in a conventional network contains both control and switching subsystems. In DONA-α, these subsystems are physically separated, as independent nodes, so the network is composed of a small number of high-performance control nodes that each control numerous low-capacity switching nodes. Since service faults in a large-scale public telecommunications network can have a severe effect on human society, the reliability of conventional networks is improved by designing redundancy into the network architecture. However, this raises the network configuration cost. DONA-α should reduce network configuration costs without drastically decreasing the reliability. This is because each switching node will accommodate fewer subscribers than a conventional subscriber node, so it will be possible to reduce the degree of redundancy. In this paper, we examine the relationship between reliability and the degree of redundancy in the architecture of a DONA-α network. We also evaluate the reliability and cost-effectiveness of a DONA-α network compared with a conventional network. © 2001 Scripta Technica, Electron Comm Jpn Pt 1, 84(11): 1–15, 2001 [ABSTRACT FROM AUTHOR]

This paper considers an ATM switching network in a self-routing system for line switching, and proposes a Batcher Banyan network with a cell copy preparation network. The idea of this switching network is to consider the upper layer line in a communication network with a hierarchical structure, and to note that the traffic of a multicast connection is much smaller than the traffic of a one-to-one connection. The hardware scale is greatly reduced by allowing a small amount of blocking. More precisely, the Batcher Banyan network is used as the basic structure, and a cell copy preparation network with a small amount of hardware and a simple cell transfer algorithm is connected before the Batcher Banyan network. The cells are copied in the cell copy preparation network and the Batcher network. Routing is executed in the Batcher network and the Banyan network. The amount of hardware is greatly reduced compared to the conventional configuration in which the structure is divided into the copying network and the routing network. © 2001 Scripta Technica, Electron Comm Jpn Pt 1, 84(9): 54–63, 2001 [ABSTRACT FROM AUTHOR]

In recent years, the trend to use wireless LANs to realize mobile or ad hoc network is expanding. Previous proposals for access control methods for wireless LANs included such methods as Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA) and polling. However, both methods have problems of transfer efficiency and delay time. Research and development on new access control methods is required to implement high-performance wireless LANs. In this paper, we examine the development of a wireless version of the wired token passing LAN access control method as a new wireless LAN access control method and propose Dynamic Token Passing (DTP) as a wireless token passing LAN access control method that incorporates a token circulation method and a data relay scheme based on a reserved logical ring. The results show that compared to existing methods, such as the IEEE 802.11 protocol, the proposed method exhibits better performance in an environment suffering from transmission path failures caused by blocking objects. A discussion of the optimum configuration for the reserved logical rings shows that if A reserved logical rings are prepared and the optimum reserved logical ring is selected, then a network can continue operation under the condition of A-1 failures. © 2001 Scripta Technica, Electron Comm Jpn Pt 1, 84(9): 16–26, 2001 [ABSTRACT FROM AUTHOR]

In recent years, VPN (Virtual Private Network) has become an important subject not only in the Internet, but also on intranets and extranets. The VPN enables us to communicate safely by using public networks. A number of researchers have studied DVPN (department VPNs). Furthermore, multicasting has been attracting attention as a means to increase the volume of voice data and multimedia data on the Internet. However, a multicast router is needed to deliver data with multicasting on the default network. Although it is possible that the IP multicast function will be added to all new routers in the future, it is not realistic to convert all default routers in use to multicast routers at this moment. Systems which make use of multicast servers without using multicast routers have been proposed. However, many problems arise in the conventional model, such as excess load on the multicast server, and traffic on its domain when data with multicasting are sent or received over the DVPN. In this paper, we propose an effective method of multicast communication over the DVPN in order to establish a secure DVPN for multicast applications such as video conferencing. In the proposed method, the relay function for multicast communication is added to the DVPN terminals themselves, which leads to easy use compared to the conventional method with multicast server. We propose two packet delivery system models, the daisy chain model and the extended daisy chain model. The daisy chain model is made up of a group of DVPN hosts, which have the copy function. In the extended daisy chain model, a branch division function is added to the daisy chain model. We present a network model for numerical calculation to show the efficiency of our proposed models. We investigate the effects of load variance, network traffic, and delay time on the performance of the two proposed models compared to the conventional model, and show that the two proposed models are superior to the conventional model. © 2001 Scripta Technica, Electron Comm Jpn Pt 1, 84(7): 45–56, 2001 [ABSTRACT FROM AUTHOR]

The authors propose two dynamic load balancing methods which give consideration to network delays in a large-scale, wide-area distributed system. In general in load balancing, the number of candidate computers for transfers increases as the number of computers being used increases, and so efforts are made to improve response time. There is, however, a concern that performance will drop due to delay overhead when candidate destinations for transfers include computers with large delays. Therefore, in this research the authors configure a collection of computers with low network delays as a cluster. They then propose a clustered dynamic load balancing method that emphasizes load information within a cluster and delay information when transfers are performed between clusters, and a threshold dynamic load balancing method which restricts process destinations by creating a threshold value with respect to the ratio between the delay required for a process transfer and the average processing request time for a process. The results of a performance evaluation performed using simulations show that each proposed method can greatly improve the average response time compared to methods that do not consider delays. In addition, each method has an environment in which it is thought to be particularly effective given its operational characteristics. © 2001 Scripta Technica, Electron Comm Jpn Pt 1, 84(6): 11–21, 2001 [ABSTRACT FROM AUTHOR]

A network interconnecting facility such as a router and a bridge may convert protocol data units passing through it. For example, ATM networks convey 53-byte fixed size cells. On the other hand, variable-length packets are transmitted on legacy LANs such as 10BASE, FDDI, and so on. Thus, an edge device at the boundary of an ATM network must perform segmentation and reassembly of packets. In this paper, we focus on packet reassembly at an edge device under the condition that interactive and noninteractive data-transfer traffics coexist. Via simulation of a queuing model, we investigate the influences of traffic load, packet length, and service priority on packet loss and delay characteristics. The result shows that one should determine the reassembly-buffer size taking into account the packet size for noninteractive service when the interactive service has high priority. Furthermore, one should determine the priority levels of both services depending on their delay characteristics when the bursty nature of the ATM traffic is reduced by flow control. © 2000 Scripta Technica, Electron Comm Jpn Pt 1, 84(1): 40–48, 2001 [ABSTRACT FROM AUTHOR]

In the super-high-speed network, a large amount of data must be transmitted and processed instantaneously. Thus, a technique which can examine past traffic and predict future situations is needed. This paper considers a traffic prediction technique in which a neural network is used for prediction. Several improvements for the basic neural prediction system are presented. The usefulness of the neural prediction system is demonstrated by comparison of the prediction performance to the linear prediction method, which is considered at present as an effective method of traffic prediction. More precisely, the prediction performance is evaluated in terms of the mean-square error for a video traffic model with short-term time correlation, and measured LAN traffic data for which the long-term time correlation has already been verified. For the former, neural prediction realizes almost the same prediction performance as linear prediction based on the ARMA model. For the latter, neural prediction is shown to give better performance than linear prediction by providing prediction error input that allows accurate recognition of the trend of change. © 2000 Scripta Technica, Electron Comm Jpn Pt 1, 83(11): 33–42, 2000 [ABSTRACT FROM AUTHOR]

This paper considers convergence type multimedia communication, where packet communications are executed concurrently from multiple information sources to a single destination node and discusses the priority assignment method for each link in each communication path. As the first step, the degradation is defined for each medium based on the average delay time in each communication path, and it is pointed out that the average, the maximum, and the variance of the degradation can be used as evaluation measures. Then, the priority assignment that minimizes the average degradation, as well as the priority assignment that minimizes the weighted sum of the maximum degradation and the variance of degradation, are discussed. An algorithm is presented for the former to derive the optimal solution, and another algorithm is presented for the latter to derive the quasi-optimal solution. Designs are tried for some network configurations using the proposed method, and their usefulness is verified. © 2000 Scripta Technica, Electron Comm Jpn Pt 1, 83(6): 88–97, 2000 [ABSTRACT FROM AUTHOR]

The authors have been proposing a satellite point-to-multipoint communication hybrid ARQ protocol in a system in which the forward path is the satellite link and the feedback path is the terrestrial computer network. This paper evaluates the performance of the protocol by a simulation, for the case where the error rates of the receiving stations are not the same or the round-trip delay times are not the same, as well as for the case where the ACK packet may be lost in the feedback path. When the packet is lost and the reception ACK packet for a frame is not returned for a certain period, that frame must be retransmitted (time-out). This paper examines values adequate for the time-out based on the round-trip delay time. Lastly, the case where the state of the forward path is nonstationary is assumed, and a method is considered where the bit error rate is successively estimated in the receiving station. It is shown that by introducing the adaptive multicopying function for the protocol based on the estimated error rate, the average transmission delay can be improved. © 2000 Scripta Technica, Electron Comm Jpn Pt 1, 83(6): 21–31, 2000 [ABSTRACT FROM AUTHOR]

To achieve fair sharing of buffer memory among the output port queues in a shared buffer ATM switch, a maximum allowable queue length (a threshold) for each queue should be set. This paper proposes a new dynamic queue length threshold control scheme in which a simple function is used to calculate the threshold value. Via simulation, we investigate cell loss characteristics when a single class and two classes of cells arrive under nonuniform load conditions. The results show that the threshold should be set to a slightly larger value than the current vacant buffer spaces to reduce the cell loss probability. © 1999 Scripta Technica, Electron Comm Jpn Pt 1, 83(4): 1–11, 2000 [ABSTRACT FROM AUTHOR]

To utilize a shared buffer efficiently, a shared buffer ATM switch with multicast function should not replicate a multicast cell at the arrival time, but should replicate it at the departure time. In such a switch, however, a multicast cell occupies the buffer for a long time, since it must stay there until the copy is sent out to all destinations. In this paper, we propose an ATM switch which can flexibly change the service mode. In the switch, if the buffer becomes congested, it serves the unicast cells prior to the multicast cells, which can reserve more vacant buffer spaces for the arriving cells. To do so, the switch has two dedicated address queues, one for the unicast cells and the other for the multicast cells. We evaluate cell loss characteristics under various load conditions via simulation. The switch is shown to provide a lower cell loss probability than the previously proposed switches. © 1999 Scripta Technica, Electron Comm Jpn Pt 1, 82(10): 1–9, 1999 [ABSTRACT FROM AUTHOR]

A recent tendency in parallel computer design has been to use general-purpose components for system configuration elements such as CPUs, disks, and memories, which used to be specially developed. Although the connection network between the processors has been specially developed, it is now possible to configure a large-scale PC cluster with good performance at low cost by making use of an ATM network as a processor connection network because of the development and cost reduction of ATM network technologies in the communication field. In this paper, a large-scale PC cluster is constructed by connecting 100 personal computers by means of a general-purpose ATM network. Applications to parallel data mining are evaluated and discussed. In particular, an analysis is carried out with a focus on the effect of TCP retransmission with cell discarding of the ATM switch on the performance. The parameter setting of a retransmission mechanism suitable for the parallel processing in the cluster is found. Further, by developing a method for setting the retransmission spacing parameters to random values for each node, it is shown that a further improvement is possible. © 1999 Scripta Technica, Electron Comm Jpn Pt 1, 82(7): 12–24, 1999 [ABSTRACT FROM AUTHOR]

This paper proposes a formulation of a packet routing problem as a resource allocation problem, where we consider a packet as an agent and a link lead to a next node as a resource. Then, we propose a packet routing method for a network using a computational ecology model with a reward mechanism proposed by Hogg and Huberman, which is an approach to multiagent systems. The efficiency of the proposed method is shown by simulation. © 1999 Scripta Technica, Electron Comm Jpn Pt 1, 82(5): 16–23, 1999 [ABSTRACT FROM AUTHOR]

An ad hoc LAN is a LAN which is temporarily formed of arbitrary stations at an arbitrary site. This paper discusses the Medium Access Control (MAC) scheme in the wireless LAN, from the viewpoint of station mobility and the basic aspect of the ad hoc LAN. In order to support the ad hoc LAN, it is desirable to apply the peer-to-peer MAC scheme, which does not require a particular infrastructure. In the peer-to-peer MAC scheme in general, however, performance degradation due to hidden terminals is a problem. Especially when the station is moved or an ad hoc LAN is formed near an existing service area, stable performance is difficult to realize. From such a viewpoint, this paper newly proposes the STMA/DA (single tone multiple access with directional antennas) peer-to-peer MAC scheme. STMA/DA utilizes the busy tone and the reservation mechanism, so that stable communication is secured even if there are hidden terminals. The geographical area occupied by the communication is minimized by using sector antennas, so that the number of simultaneous communications in the system can be increased. The performance of STMA/DA is evaluated by computer simulation. Through a comparison to CSMA and RI-BTMA, the effectiveness of STMA/DA is demonstrated. © 1999 Scripta Technica, Electron Comm Jpn Pt 1, 82(5): 74–84, 1999 [ABSTRACT FROM AUTHOR]

In this paper, we propose a multihop network combined with unidirected ring networks. Our proposed network has large connectivity, small internodal distance, and high efficiency of link assignment. We theoretically analyze the mean internodal distance and the network connectivity of the proposed network. It is shown by numerical calculation and computer simulation that the proposed network achieves smaller mean internodal distance than the conventional connective semirandom network and the same network connective probability as the regular networks. © 1998 Scripta Technica, Electron Comm Jpn Pt 1, 82(4): 14–22, 1999 [ABSTRACT FROM AUTHOR]

In order to manage telecommunication networks efficiently, it is necessary to isolate the essential cause of problems by relating all alarms, using knowledge of the physical connections of network elements or knowledge derived from experience. This is called “alarm correlation.” As the number of managed elements increases, however, the number of alarms becomes enormous; thus, it is crucial to correlate these alarms efficiently. In this paper, we introduce a new concept of “relational sets” and propose a high-speed searching method, the dynamically hierarchical hash (DHH), which uses temporal correlations between the alarms received. Further, we validate the effectiveness of this method through computer simulations. © 1998 Scripta Technica, Electron Comm Jpn Pt 1, 82(4): 71–79, 1999 [ABSTRACT FROM AUTHOR]

The authors previously derived the condition for a non-blocking architecture in a wavelength division multiplexing (WDM) photonic switching network, i.e., the generalized parallel delta network (GPDN) [1]. GPDN is configured as a multi-stage link connection of multiple unit switches, where the hardware complexity can be adjusted by varying the number of stages according to the network size. It can provide a framework for network design of various trunk networks, access networks, and in-station networks. This paper presents an architecture for a wavelength division multiplexing photonic unit switch that is applicable to a wavelength division multiplexing photonic switching network (GPDN), and evaluates the cost. The proposed wavelength division multiplexing photonic unit switch is constructed by assuming the situation in which an optical signal is multiplexed and handled without being converted to an electrical signal and employing the wavelength division multiplexing technique. The latter is considered promising for future networks from the viewpoint of reducing hardware complexity and economizing broadband communication. The input call to the carrier is wavelength converted to the carrier corresponding to the required outlet. Then, an optical filter is used to realize spatial switching. By combining the photonic unit switches proposed here, a large-scale switching network can be constructed. © 1998 Scripta Technica. Electron Comm Jpn Pt 1, 81(8): 59–67, 1998 [ABSTRACT FROM AUTHOR]

In multiple-access schemes for digital satellite communication, a collective arrangement is needed in which all the uncoordinated users may share a single satellite channel. The slotted ALOHA scheme is basically a random access scheme where the tint axis is slotted and each earth station transmits packets at random. In this paper, the stability issues of the slotted ALOHA scheme and its useful limits tie discussed first. This gives the useful range of a basic type of random access scheme. Next, the instance conflict scheme is explained where multiple access according to the demands of various users is possible. This scheme can satisfy the basic requirements of the satellite channel according to the demand of various users. [ABSTRACT FROM AUTHOR]

In the clock synchronization of the closed area network (such as local area network (LAN) and metropolitan area network (MAN)) to the wide area network, it is advantageous to use the hybrid synchronization from the viewpoint of reliability, where the master-slave synchronization is applied at several points from the wide area network, and the mutual synchronization is applied within the closed area network. This paper considers the case where the hybrid synchronization is applied to the ring-type area network and derives the condition for the existence of the stable phase state. When the synchronization circuit is a two-input complete integrating second-order PLL with the sinusoidal or equilateral triangular phase comparator characteristic, the sufficient condition for the existence of the stable phase state of the whole network is that the ratio of the gain for the mutual synchronization to the gain for the master-slave synchronization is less than 1 in at least one PLL. When the synchronization circuit is the incomplete integrating PLL, the sufficient condition is that the above gain ratio is less than a certain limit in at least one PLL, and the ratio is above the inverse of that limit in other PLL, where the limit is less than 1 and is determined by the difference between the input frequency from the wide area network and the center frequency of PLL as well as the magnitude of the gain. The above properties are shown theoretically and are verified also by computer simulation. [ABSTRACT FROM AUTHOR]

In realization of high-speed and broadband networks of the next century, expectations are toward optical switching and a linear lightwave network (LLN) is proposed as one of the possibilities. Using LLNs, this paper proposes a multicast routing system in which a tree structure is formed with no loop inside the network. In this way, the waste of bandwidth caused by transmission of signals to unintended destinations is avoided. The scheme in which the tree is settled dynamically in case of generation of a call is proposed and its function is evaluated through simulations. As the results, the possibility of the efficient use of the unused links and realization of call setup control of equitable terminals are shown. Further, because of inherent LLN characteristics, the generated call blockings are small and superior performance is achieved in comparison to the previously proposed methods. [ABSTRACT FROM AUTHOR]

Message sequence charts have recently been the focus of considerable attention as formal specification descriptions and become standardized. However, it is difficult to use them for specifying services with variant process structure, such as interrupt services. This paper proposes a method that allows the number of processes in a service to change by selecting a suitable message sequence chart and switching the operation of each process over from the previous message sequence chart according to a specific procedure. Using this method, conventional synthesizing and verification techniques that deal with services with constant process structure can be applied to services with variant process structure. [ABSTRACT FROM AUTHOR]

Optical networks that support high-speed, low-cost communication services are required. An optical path is proposed that utilizes wavelength multiplexing technology at the path layer of transport network layers. Optical path achemes, wavelength path (WP)/virtual wavelength path (VWP) that can realize optical path networks are also proposed. Although wavelengths are a resource in optical networks, the number of wavelengths required in construct an optical path network has not been evaluated. This paper describes a method of network protection using the optical path technique that enhances network reliability and proposes a method for establishing a restoration path that restores failed active paths in both WP and VWP schemes. Proposed herein is an optical path accommodation design to establish WPs or VWPs in a physical network considering the proposed restoration method. Furthermore, optical path networks are constructed by establishing WPs or VWPs on a physical network that has a regular structure and the number of wavelengths required in the network are evaluated. This evaluation makes it clear that the VWP scheme requires fewer wavelengths than the WP scheme. [ABSTRACT FROM AUTHOR]

This paper describes a hitless path switching technique that utilizes fully the benefits of an ATM. First, hitless path protection switching in ATM networks is discussed. It is shown that for the path configuration of working and preestablished protection paths, the best switching method switches on the transmitting side. Then the proposed switching algorithm is described, along with several examples of its application to mesh and ring networks. A cell interval compression method suitable for hitless path switching is proposed that empties and disconnects the transmission delay difference absorption buffer. Finally, an approximate formula is proposed that directly determines the multinode queuing delay. This delay is needed to calculate the capacity of the delay difference absorption buffer. The accuracy of the formula is confirmed by simulations. [ABSTRACT FROM AUTHOR]

This paper considers the application of the modulation level-controlled variable capacity radio channel to asynchronous transfer mode (ATM) networks and analyzes the cell loss characteristic. A system is proposed in which the transmission capacity of the overall system can be enlarged by controlling the number of modulation levels. This helps to enhance the absorption effect for the traffic variation in ATM networks. Therefore, the proposed system can reduce the cell loss probability. As a result of quantitative analysis of the system. it is shown that the cell loss improvement is by the factor of several hundreds to several thousands compared to the virtual path fixed-capacity transmission system and several to several thousands compared to the virtual path capacity control system proposed up to present. [ABSTRACT FROM AUTHOR]

This paper analyzes the performance of a number of MAC protocols m high-speed ring-type networks and determines the suitable region for each protocol. The authors previously have evaluated the slotted rings, register-insertion rings and register-insertion-type slotted rings, and begin with the analysis of the token ring in this report. To compare the results of the various analyses, the models of each MAC protocol are constructed under the same set of conditions and evaluated with the same method (EPA). Since the nodes of each model are assumed to have buffers to finite capacity, the discussion of buffer overflow is made possible. The results obtained from analysis matched closely with simulation results verifying the accuracy of the analysis. The MAC protocols also are analyzed under high-speed network conditions, and a comparison of the results reveals the most suitable protocols for various environments keeping m mind the issue of buffer overflow. [ABSTRACT FROM AUTHOR]

With the widespread use of high-speed broadband communication, nonblocking switching networks in video distribution and other applications for the broadcast distribution connection will be required. Numerous studies have been made of performance analysis for switching networks for one-to-one switching connections, but it is anticipated that a future study is needed for one-to-many switching connections. This paper presents a construction for the copy distribution stage, which copies the cell for the broadcast distribution. The copy distribution stage is composed of a multistage connection network. The basic principle of operation is that the cells are arranged appropriately and are copied by the differential address driving. The proposed network is suited to high-speed operation since nonblocking self-routing is realized and the control logic is simple. The proposed network assumes that it precedes the network for one-to-one switching connection. Then one-to-many broadcast distribution connection is realized. [ABSTRACT FROM AUTHOR]

Packet exchange methods have come to play a central role as exchange methods in various networks. Especially, the random access method has been attracting attention since networks cm be constructed comparatively easily by this method without having to establish special control offices. However, a collision of packets is inevitable with the random access method, which becomes a major factor in decreasing network efficiency. In past access methods, collided packets, in general, are retransmitted. In this paper, a method of considering a collision of packets as collision error, correcting this error, and demultiplexing (isolating) into original packets is proposed. In this method, demultiplexing (isolating) collided packets as well as detecting and correcting the transmission path error occurring simultaneously is possible. In addition, the method has a characteristic of superior compatibility with conventional systems. In addition, it has been clarified that the performance is improved greatly when the proposed method is applied to a random access network, by taking the ALOHA system as an example. [ABSTRACT FROM AUTHOR]

When a long message such as file is to be transmitted in a packet-switching network, the source node segments the message into a number of packets and the packets are transmitted to the network, Those packets are transferred to the destination node through several intermediate nodes and then reassembled into the message. This paper models the packet-switching network into a tandem queueing network. By analyzing the model, the performance of message transfer in the packet-switching network is evaluated. The discrete-time system is considered in the analysis. An algorithm is presented which strictly derives the distribution of time needed in the transfer of all packets composing the message from the source node to the destination node (message transfer time). In the numerical example, it is assumed that the service time at each node is composed of the packet processing time and the packet transmission time. The effect of the packet size on the mean and variance of the message transfer delay is analyzed. The case where the message size and the number of intermediate nodes are increased also is considered. [ABSTRACT FROM AUTHOR]

The NUCLEI (Nagasaki University codec using large interframe prediction errors as important parts) method is proposed in this paper as a new interframe coding method for the asynchronous transfer model (ATM). Characteristics of this method are as follows: 1) interframe predictions are done using a full band, considering the high coding efficiency during the normal state; 2) interframe predictions are done even after toll loss, considering the high coding efficiency after cell loss; and 3) utilizing the fact that the small moving areas are less important than the large moving areas in dynamic images (motion pictures), large moving areas or blocks with large interframe differences are allocated to priority (higher priority) cells as priority blocks and comparatively small blocks are allocated to nonpriority (lower priority) cells as nonpriority blocks. These characteristics are realized by assigning frame memories for storing priority block signals in addition to frame memories storing all images. It is shown by computer simulations that good interframe coding characteristics can be realized in both during the normal state and after a cell loss. [ABSTRACT FROM AUTHOR]

This paper evaluates the performance of packet voice multiplexer with selective packet-discarding (SPD) control. The control scheme is based on the embedded ADPCM coding technology wherein the encoded information is grouped into two separate parts: a more significant part (MSP); and a less significant part (LSP). When the buffer length exceeds a certain threshold, the LSPs of the incoming packets will be discarded to relieve the congestion. By assuming that the voice packets (cells) arrive at the multiplexer in phase-type Markov renewal process (PH-MRP) and that they are all of the fixed length, we model the system under consideration as PH-MRPα)/-Ek/1/m(m1) with sufficiently large k. Using the matrix-analytic (MA) method, recursion formulas and effective computation algorithms for the performance measures of interest are derived. Numerical examples are illustrated. [ABSTRACT FROM AUTHOR]

This paper proposes a control method for the retransmission rate in the continuous-time 1-persistent CSMA/CD system with a finite number of stations so that the throughput is maximized. It is shown first that there exists an optimal load that maximizes the throughput. Assuming that the system always operates for the optimal load, the maximum throughput that can be achieved in the 1-persistent CSMA/CD system is derived. Then an adaptive CSMA/CD system is proposed in which each station monitors the number of collisions and other parameters to estimate the present load and determines the retransmission rate so that the optimum load is realized. As a result of simulation, it is shown that the adaptive CSMA/CD system exhibits a considerably better performance than the Ether-type CSMA/CD system employing the binary on the 1-persistent scheme. It is shown that the proposed system can realize the throughput which is almost close to the forementioned maximum in the highly loaded condition. [ABSTRACT FROM AUTHOR]

Communication protocols in hierarchical network architecture have become increasingly complicated. At the same time, the costs of implementing communication software have been increasing. Thus, for the widespread use of these protocols, it is important to reduce these costs as much as possible. This paper adopts Estelle [6], standardized by the International Standardization Organization (ISO) as a Formal Description technique (FDT), to specify standard communication protocols. Moreover, since Estelle is based on an extended finite state machine, it is easy to translate any of its specifications into an efficient program. For programs which build and analyze Protocol Data Units (or PDU), a capability of specifying PDU formats into an existing FDT has been integrated. In this paper, a protocol specification method which integrated ASN.1 (Abstract Syntax Notation One)[7] into Estelle has been developed. A translator which translates a specification in the proposed method into Ada (a concurrent programming language) has also been developed. [ABSTRACT FROM AUTHOR]

The wide zone AVI system is an automatic vehicle identification system that can cope with more than one vehicle in the detection zone. Such a system includes a possibility of identification failure by the collision of ID packets, in addition to the ordinary transmission failure. A method to remedy this is to transmit the ID packet several times. This paper considers two methods of transmitting the ID packet from each vehicle. In the first, the packet transmission interval follows an exponential distribution, and in the second, a specified number of transmissions are made at random points which are distributed uniformly during the zone pass period. Using approximate analyses of the Identification failure probability, the effects of the design parameters on the system performance are examined for each method. As a result, it is shown that the constant number transmission method has a better performance than the exponential transmission interval method, in terms of the probability of identification success, the number of ID packet transmissions, and other aspects. [ABSTRACT FROM AUTHOR]

As a promising means of realizing broadband ISDN, the ATM (asynchronous transfer mode), which is based on fast packet switching, is considered interesting. In video communication by ATM, the constancy and improvement of picture quality by variable rate coding is expected, while the quality deterioration by cell loss is a serious problem. This paper proposes a video coding method Introducing hierarchical coding to cope with the cell loss problem. The proposed hierarchical coding extracts the low- frequency components of the image by iterating the hierarchical processing based on the spatial filter. Then it is possible to sup- press the effect of the quality deterioration caused by cell loss, as long as the low-frequency components are retained. First, the configuration of the proposed system is discussed. Simulations are executed to observe the effect of the cell loss and the coding characteristics, demonstrating the effectiveness of the proposed method in coping with the cell loss. Then discussions are made on the optimal number of hierarchical processings to cope with the cell loss and the protection level for hierarchical Information. Furthermore, it is shown that a new type of video communication can be realized by introducing hierarchical coding. [ABSTRACT FROM AUTHOR]

Mobile telecommunication systems by satellite meet the requirement of antenna radiation pattern of wide beamwidth. The wide beamwidth of the circular microstrip disk antenna is obtained by high dielectric constant substrate. However, the dielectric losses and difficulty in manufacturing are problems for high dielectric constant materials. Wide beamwidth is obtained also using the antenna with an electrically small magnetic current loop. This paper presents the ring cavity antenna without dielectric materials. It also has electrically small magnetic current loop. Antenna characteristics calculated by the cavity model showed that the ring cavity antenna has wide beamwidth. Theoretical values are verified also by the measured radiation pattern and the input impedance. [ABSTRACT FROM AUTHOR]

In multimedia communication, the quality requirement from each medium to the net- work is not a definitive measure but usually contains a fuzziness. This paper represents, by a fuzzy set, the propagation delay of the network that can be tolerated by each medium and discusses the optimal burst multiplexing method for the required quality. First, various kinds of fuzzy quality requirements are examined, and it is shown that the quasi-optimal quality measure is obtained in common with the control at the network node. Then based on the quasi-optimal quality measure, two control methods at the network node are proposed: the variable priority and fuzzy priority methods. The performances of these methods are examined by numerical analysis and computer simulation, and the results are compared with those of the traditional control methods. Finally, it is shown that the variable priority method is better when the packet delay at each node can be measured, and the fuzzy priority method is better when it cannot be measured. [ABSTRACT FROM AUTHOR]

The advent of VSAT (very small aperture terminal) has made it possible to consider a new utilization of satellite communication. This paper proposes a new access protocol v-ALOHA, which is matched to the characteristics of the packet switching network based on VSAT. p-ALOHA is used as a means to cope with the requirement for simplification. To complement its disadvantage of low through- put, a new control scheme is combined based on the new utilization. The performance is improved by the capture effect. The basic idea of the control scheme is to vary the power with time while transmit- ting packets. The difference in the transmission-start time is converted into the power difference and produces the capture effect, thereby improving the performance. Thus, an access protocol is realized which improves the low throughput of p-ALOHA while preserving its simplicity. This paper describes the basic idea of this v-ALOHA and the protocol structure, indicating qualitatively its features. Then the throughput performance is formulated through a quantitative analysis. It is shown from the numerical results for the fixed-length packet that the performance of the proposed scheme is better than other capture schemes, approaching sufficiently the performance of s-ALOHA. For the case of the variable-length packet, there is no slot loss as in s-ALOHA, and the proposed scheme is shown, using an example, to be better sometimes than s-ALOHA. [ABSTRACT FROM AUTHOR]

This paper considers the logical link control (LLC) sublayer in the CSNAICD local area network, and compares analytically the performance of connection-oriented (CO) and connectionless (CL) protocols. As the medium access protocol, slotted nonpersistent CSMA/CD with a priority assigned to some types of frames is employed. In comparison, the overhead for connection establishment/ release in the CO-LLC protocol, and the receiver-buffer overflow in the CL-LLC protocol are especially considered, and the two kinds of protocols are compared. In the CO-LLC protocol, a connection is established corresponding to a message. Employing the technique of equilibrium point analysis, the throughput and the average response time of the two kinds of MAC protocols are derived. To compare the performance of the two kinds of protocols, the receiver-buffer size is kept constant, and the average number of data frames in a message is varied. As a result, it is shown that the CL-LLC protocol is better than the CO-LLC protocol when the average number of data frames is sufficiently less than the receiver-buffer size, but the order is reversed when the average number of data frames is increased. [ABSTRACT FROM AUTHOR]

In multislot circuit-switched networks, it is necessary to apply a GOS control method to trunk groups to improve network utilization and to protect services against overload. As a GOS control method for multislot networks, this paper considers a trunk reservation method, split high-usage group method, split final-group method and virtual circuit method, which have also been studied for telephone networks. They are compared then by evaluating their blocking performance and peakedness of overflow and carried traffic involved. Based on numerical examples, it is shown that although the trunk reservation method is superior with respect to trunk utilization, there are some cases where the other methods are advantageous with respect to the ability to withstand overload and peakedness of traffic. [ABSTRACT FROM AUTHOR]

The performances (throughput and transmission delay) of the Co-back-N ARQ scheme are affected greatly by the bit error rate of the link and the round-trip propagation delay. This paper discusses the transmission delay of the tandem Co-back-N scheme, in which the round-trip propagation delay can substantially be reduced. First, the generating function for the transmission delay distribution is derived, and then the result is applied to derive the average transmission delay. As a result, it is shown clearly that the performance of the tandem Co-back-N ARQ scheme is better than that of the standard Co-back-N ARQ scheme. [ABSTRACT FROM AUTHOR]

In the information communication system it is necessary to establish the protocol conversion, network connection and media conversion techniques to realize communications among crputers and terminals with different communication protocols. This paper considers the hierarchical protocol defined based on the OSI basic reference model, and discusses the protocol conversion method together with its realization. A versatile method is introduced in the proposed hierarchical protocol conversion system, in which a protocol conversion can be made from PAD function for various terminals as well as from the nonprocedural level to the personal computer communication protocol recommended by the Ministry of Posts and Telecommunications. The protocol conversion equipment developed based on the proposed Idea is demonstrated, and its usefulness is indicated. [ABSTRACT FROM AUTHOR]

This paper proposes a new protocol validation method to detect such errors as lack and redundancy in given protocol specifications. This method is characterized by the fact that it enumerates a minimum number of global system states and global state transitions that are required for the validation. This leads to a remarkable reduction in both validation tine and required memory size. Furthermore, this proposed method enables the validation of large-scale protocol specifications, which was impossible by the conventional validation methods. [ABSTRACT FROM AUTHOR]

This paper compares the stability of five routing-procedures: nonalternate-routing (NAR), alternate-routing (AR), alternate- routing-trunk-reservation (AR-TR), alternate-routing-external-blocking (AR-EB), alternate-routing-trunk-reservation-and-external-blocking (AR-TR-EB), in circuit-switched network. Discussion is also made on how to change the routing procedure dynamically when network traffic becomes large. First the stability of AR-TR, AR-EB and AR-TR-EB routing-procedures is discussed by applying the cusp catastrophe theory, and the cusp is found in which a sudden change of end-to-end-blocking-probability (EEBF) occurs, Next, the stability of five routing-procedures is compared using the cusp. Finally, two methods are proposed that change the routing- procedure dynamically when network-traffic becomes large, and their effectiveness is shown. [ABSTRACT FROM AUTHOR]

For efficient processing of various types of traffic in LAN's, the introduction of a priority mechanism is considered to be effective. This paper considers nonpersistent CSMA/CD systems with a new priority mechanism, and derives the throughput and the average response time for each priority class by using the imbedded Markov chain method. The priority structure is achieved by assigning higher exponential retransmission rates to higher priority classes. As for the basis of priority assignment, we analyze two mathematical models: one is terminal-based and the other is message- based. Because the models allow feedback of packets, we can apply our results to error- prone CSMA/CD systems, also. Numerical results show that the simple priority mechanism works effectively, and clarify the differences between the terminal-based system and the message-based system. [ABSTRACT FROM AUTHOR]

High-speed packet switching is a promising technology for multimedia communications in such wide-band ISDN as speeches, images, add data. In this type of high-speed packet switching, priority-transmission technology will be effective, since a different packet-transmission quality (e.g., transmission delay and loss-rate) is required a nonpreemptive Σ M1/G/1(Σ N1) system which is the basis of the priority transmission system. In addition, n-priority classes (where n is arbitrary) are treated using models of the Poisson arrival, general service-time distribution, and individual waiting-position type, two priority classes. To obtain packet loss-rate and delay characteristics, this paper analyzes mainly the marginal distribution of each priority class. The results of the analysis include an extension of the Cooper-Gebhart relationship between distributions at an arbitrary moment and after the completion of a service. For the packet loss characteristic and delay-time characteristic and their approximate calculation methods, the reader is referred to the author's work in (6). [ABSTRACT FROM AUTHOR]

Recently, some efforts have been made to construct a multipoint teleconference service, where a number of remote locations are mutually connected through communication-channels. This paper proposes a talker location recognition through sound image localization for easy recognition. More precisely, three proposals are made: the coding for talker location; the code processing at the node connecting the multipoints; and the sound image control with excellent talker localization. It is shown that the proposed coding scheme can simplify the code processing, and can be executed correctly even when node devices are connected in cascade. For the sound image control, subjective assessment experiments were made for the case of a single talker, and the case of more than one talker. The result indicated that the proposed method is useful in the talker location recognition. [ABSTRACT FROM AUTHOR]

When different kinds of local area networks are interconnected, a gateway holds the key to better performance of the interconnected network. Since gateways are liable to overflow easily because of their comparatively long processing time, it is necessary to estimate their appropriate buffer size with regard to the input traffic. This paper considers a local area network working under CSMA/CD protocol. First, we analyze the output process of messages which constitutes a part of arrival process at a gateway, by solving an embedded Markov chain. Second, two stochastic models of the gateway are formulated by queueing theory and loss probability of messages is evaluated in regard to its buffer size. Numerical results show some interesting features of the output and arrival processes. [ABSTRACT FROM AUTHOR]

In a multimedia packet-switching system handling voice, moving images, data, etc., the bursty traffic generated by moving image terminals can have an unfavorable effect on the traffic quality of other media. To avoid this problem, the handling of bursty traffic should be established. In this paper, the packet bursts are modelled by a batch arrival and the effect of burstness on packet overflow characteristics is examined by analyzing batch arrival queueing system with finite waiting rooms. The results obtained are summarized in the following. (1) To achieve the practical range of packet overflow probabilities (e.g., 10–6) in a system handling burst traffic only, a high buffer size indicator k (the ratio of the buffer size to the number of packets in a burst) is needed to utilize the transmission lines efficiently. (2) To provide efficient utilization of the transmission lines and to realize practical packet overflow probabilities (e.g., 10–6) in systems transmitting a mixture of burst and nonburst traffic, the share of burst traffic should be very small. [ABSTRACT FROM AUTHOR]

This paper analyzes a loss system with finite sources. The idle period of a source forms a renewal process and each customer requires several channels simultaneously. The service time of customers which require the same under of channels exhibit a general independent identical distribution. This is a general traffic model for a subscriber concentration stage in a high-speed wide-band communication network. The distributions of the numbers of customers in the system at arbitrary and customer arrival epochs are derived using the supplementary variable techniques, where the backward recurrence-time of call idle period and holding time are chose as supplementary variables. It is shown that these distributions are determined only by the mean service time and the mean idle period of a source, irrespective of their distributions (robustness). Finally, some traffic characteristics, such as loss probability and circuit occupancy, are evaluated numerically for two-traffic models and unbalanced input traffic models. [ABSTRACT FROM AUTHOR]

Most of the performance analyses of CSMA/CD systems have assumed that each user has a buffer space for only one packet. Using the technique of equilibrium point analysis, this paper analyzes the dynamic behavior of a slotted nonpersistent CSMA/CD system with a finite number of users, each of whom has a finite buffer space for more than one packet. Two schemes of packet transmission are studied: (1) once a user captures the channel, he continuously transmits all of the packets which have been stored in his buffer by the instant of the capture (gating type); and (2) he transmits at most one packet at a time (limiting type). First the gating type is analyzed, and its throughput, its average packet delay and its system stability are evaluated. An approximate probability distribution for the number of packets in the buffer space of a user is derived. On the basis of these analytic results, we study how to select both the buffer size and the rescheduling probability of packets which guarantee the system stability, keep the probability of buffer overflow sufficiently small and at the same time minimize the average packet delay. The two schemes are compared and it is shown that the gating scheme out performs the limiting scheme in terms of average packet delay and the average and variance for the number of packets in a buffer. [ABSTRACT FROM AUTHOR]