Search

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]