Summary With the recent advances in machine‐to‐machine communications, huge numbers of devices have become connected and massive amounts of traffic are exchanged. Machine‐to‐machine applications typically generate small packets, which can profoundly affect the network performance. Namely, even if the packet arrival rate at the router is lower than the link bandwidth, bits per second, it can exceed the router forwarding capacity, which indicates the maximum number of forwarded packets per second. This will cause the decrease in the network throughput. Therefore, eliminating the packets per second limitation by chunking small packets will enable machine‐to‐machine cloud services to spread further. This paper proposes new packet‐chunking schemes aimed at meeting both application requirements and improving achievable router throughput. In our schemes, multiple buffers, each of which accommodates packets classified based on their delay requirement, are installed in parallel. Herein, we report on analysis of the theoretically performance of these schemes, which enabled us to derive some important features. We also propose a scheme whereby a single chunking buffer and parallel multiple buffers were arranged in tandem. Through our simulation and numerical results, we determined that these schemes provide excellent performance in reducing the number of outgoing packets from the router while meeting various delay requirements.

This paper presents design and experimental evaluation of a distributed autonomous multi-hop vehicle-to-vehicle (V2V) communication system over TV white space performed in Japan. We propose the two-layer control channel model, which consists of the Zone Aware Control Channel (ZACC) and the Swarm Aware Control Channel (SACC), to establish the multi-hop network. Several vehicles construct a swarm using location information shared through ZACC, and share route and channel information, and available white space information through SACC. To evaluate the system we carried out field experiments with swarm made of three vehicles in a convoy. The vehicles observe channel occupancy via energy detection and agree on the control and the data channels autonomously. For coarse synchronization of quiet periods for sensing we use GPS driven oscillators, and introduce a time margin to accommodate for remaining drift. When a primary user is detected in any of the borrowed channels, the vehicles switch to a vacant channel without disrupting the ongoing multi-hop communication. We present the experimental results in terms of the time to establish control channel, channel switching time, delivery ratio of control message exchange, and throughput. As a result, we showed that our implementation can provide efficient and stable multi-hop V2V communication by using dynamic spectrum access (DSA) techniques.

With the proliferation of new wireless service, scarce wireless resources is expected to become a critical issue. For this reason, cognitive radio mobile ad hoc networks (CogMANET) are being developed as a promising solution to this problem. However, in CogMANET, channel switching is inherently necessary whenever a primary user with a license appears on the channel. Allowing secondary users to choose an available channel from among a wide spectrum range thus enables reliable communication in this context, but communication characteristics such as bottleneck bandwidth and RTT will change with channel switch. In response to this change, TCP has to adaptively update its congestion window (cwnd) to make an efficient use of the available resources. For this purpose, TCP CRAHN was proposed for CogMANET. In this paper, TCP CRAHN is first evaluated in cases where bottleneck bandwidth and RTT drastically change. Based on these results, TCP CoBA is proposed to further improve the throughput of the above use cases. TCP CoBA updates the cwnd based upon the available buffer space in the relay node upon channel switch, as well as other communication characteristics. Through simulations, we show that compared with TCP CRAHN, TCP CoBA improves the throughput by up to 200 percent.

This paper presents design and experimental evaluation of a distributed autonomous multi-hop vehicle-to-vehicle (V2V) communication system over TV white space performed in Japan. We propose the two-layer control channel model, which consists of the Zone Aware Control Channel (ZACC) and the Swarm Aware Control Channel (SACC), to establish the multi-hop network. Several vehicles construct a swarm using location and direction information shared through ZACC, and share route and channel information, and available white space information through SACC. To evaluate the system we carried out field experiments with swarm made of three vehicles in a convoy. As the test case application, the leading vehicle sends real-time speed and sudden break information to the rear vehicle, while the vehicle in between acts as the relay. The vehicles observe channel occupancy via energy detection and agree on the control and the data channels autonomously. For coarse synchronization of quiet periods for sensing we use GPS driven oscillators, and introduce a time margin to accommodate for remaining drift. When a primary user is detected in any of the borrowed channels, the vehicles switch to a vacant channel without disrupting the ongoing multi-hop communication. We present results of the experiments in terms of the time to establish control channel, channel switching time, and throughput.

In a peer-to-peer (P2P) network, where nodes can freely join and leave at any time, the number of participating nodes tends to vary over time. Furthermore, in an unstructured P2P network, where any node can freely select direct linkages with any other node or nodes, this free selection/linkage process determines the network topology. A recent study that examined unstructured real-world P2P networks revealed that the observed topology of such networks will only follow a power-law degree distribution in a cumulative manner when the observation time period is long enough to allow numerous nodes to join and leave the network. Thus, to ensure the accuracy of simulation-based evaluations in unstructured P2P network research, this property, observed in real-world P2P networks, should also be reflected in topology generation for simulation. Accordingly, in this study, we propose a topology generation method in which the nodes participating in the network vary over time, a newly participating node randomly selects other nodes for linkage, and the resulting P2P network topology only follows a power-law degree distribution when the topology information collection time exceeds a certain duration. The proposed method is validated by simulation results.

Vehicular networking has significant potential to enable diverse range of applications, including safety and convenience. As the number of vehicles and applications using the specially designated wireless spectrum grow, one can expect substantial increase in the bandwidth requirements. In this paper and demonstration, we advocate that dynamic spectrum access techniques facilitating the utilization of white spaces for vehicles will be the first step towards solving the expected spectrum shortage. In the demonstration, we will introduce field tests of a multi-hop inter-vehicle communication system with distributed and autonomous TV white space channel selection, performed in Japan from January to February 2012. Furthermore, in addition to the field test videos, we will show an indoor emulation of the entire system used during the field tests in the vehicles.

The present paper proposes a self-adaptive routing method for temporary information sharing in a wireless mesh network without a global connection that is deployed in a geographically limited area. The temporary information sharing indicates that for the case in which information senders and receivers freely appear in the network, information is forwarded from an information sender to a receiver. The self-adaptive routing method determines a unique route by operations among numerical values assigned to each node and adaptively changes the assigned numerical values according to the routing results. A variety of types of operations and numerical values can be used for this routing method. In simulations, the method is evaluated in terms of routing reliability under an appearance pattern of an information sender-receiver pair in which a location of the receiver is fixed. Simulation results reveal that routing reliability depends on the types of operations and numerical values and is related to how frequently a closed route is generated in the routing. The results also suggest that the degree of randomness in routing is important for reliable routing and varies with the types of operations and numerical values.

Future vehicular communications systems are expected to utilize the vacant channels (white spaces) of the spectrum, otherwise allocated for specific designated use. One such candidate of white space comes from the TV broadcast band. In this demonstration, we will first present animated results of a TV spectrum measurement campaign along the entire portion of Interstate I-90 located in the US state of Massachusetts. Next, we will demonstrate a cyber-physical proof-of-concept lab implementation of our previously developed control and data channel assignment schemes for vehicle-to-vehicle communications over (TV) white space. Finally we will show a video of actual vehicle to vehicle communications field tests conducted in Japan using TV white space.

In ubiquitous Wireless LANs (WLANs), Mobile Nodes (MNs) are likely to experience many and frequent handovers between WLANs managed by different organizations/ISPs during TCP communication. Although we proposed a WLAN handover management scheme based on the number of frame retransmissions, the proposed scheme cannot adapt to multirate WLAN used in a more realistic environment. Furthermore, the handover is controlled based on only a predetermined threshold of the number of frame retransmissions, thereby degrading the performance drastically when the inappropriate threshold is employed. Therefore, in this paper, to enhance the practicality of handover scheme in a realistic environment where WLAN supports multiple data rates and automatically changes the rate in response to wireless link condition, we propose a handover management scheme adaptable to multi-rate WLANs. Our scheme exploits two sorts of information to recognize the wireless condition appropriately: (1) data rate used most frequently (DRMF) by each interface and (2) frame retransmission ratio (FRR) on each interface for some duration. The former of two criteria first enables us to estimate an area where we should start handover, and if DRMFs of two interfaces are same in the area, the latter then allows us to compare wireless condition on two interfaces precisely, thereby giving an optimal handover point. The simulation results demonstrate the advantages of the proposed method, especially in the multi-rate WLAN environment.

Abstract In this paper, we introduce a Mean Value Analysis (MVA)‐based methodology for the performance evaluation of transactions executed in a multi‐server distributed system. We first present probabilistic arguments to investigate under which conditions MVA models provide worst‐case response times, without large overshoots. Then, we show how to characterize the distributed server system so as to construct a model to predict response times as well as to estimate system capacity. Finally, we exemplify the methodology usage via transactions implemented in two distributed Linux systems. Copyright © 2010 John Wiley & Sons, Ltd.

In this paper, we focus on the problem of tracking a moving target in a wireless sensor network (WSN) in which the capability of each sensor is relatively limited in order to construct large-scale WSNs at a reasonable price. In our prior study, we proposed a simple multi-point surveillance scheme for a moving target in a WSN and demonstrated that the scheme achieved high tracking probability with low power consumption. However, we have not suggested guidelines for designing the WSN to completely satisfy the wide range of user requirements. Therefore, in this study, we examine the relationship between tracking probability and sensor density through simulations, and then derive an approximate expression representing the relationship. As a result, we show guidelines for sensor density, tracking probability, and the number of monitoring sensors that satisfy a variety of application demands.

In this paper, we introduce a mean value analysis based methodology for performance evaluation of transactions executed in a multi server distributed system. We first present probabilistic arguments to investigate under which conditions MVA models provide worst case response times, without large overshoots. Then, we show how to characterize the distributed server system so as to construct a model to predict response times as well as estimate system capacity. Finally, we exemplify the methodology usage via transactions implemented in two distributed Linux systems.

This paper proposes an evolutionary approach to the network traffic optimization under the constraint of congestion avoidance. The individuals of the evolving population directly represents a set of paths in a network, and corresponding crossover and mutation operators are provided. The optimization is a global one, i.e. it will not optimize the paths independently but also taking link sharing into account. To avoid the situation that the optimization will result in no traffic for some of the senders (which is also an element of the feasible space in congestion avoidance), we use the user fairness concept. A general approach to user fairness is also provided. The fitness of an individual (path set) is computed from the total traffic in the maxmin fairness state. Experiments on certain graph structures were performed. The results were compared with a path selection strategy based on single path evaluation only. The experiments for networks in the dimension 10 to 80 nodes demonstrate that an increase in performance of around 10% can be achieved in many cases, even with rather small population sizes and number of generations.

The use of dynamic spectrum access techniques has a great potential in future inter-vehicle communications, while it must cope with (i) temporal and spatial spectrum utilization changes introduced by the primary and secondary users (environmental changes); and (ii) topology changes due to movement of vehicles (spatial movement). In the present paper, along this line, dynamic per-hop channel switching in multi-hop vehicular ad hoc networking is investigated. After defining a set of simple metric-based dynamic channel selection schemes with/without spatial-awareness, the basic performance (the total communication duration and the amount of transmitted data) are evaluated. These spatially-aware schemes estimate the maximum communication period, and tend to select a channel with a large amount of possible data transmission within the period. The simulation results demonstrate the advantages of the proposed spatial-awareness, especially in the multi-hop and highly congested environments with high-speed mobility.

In this paper, we are studying a generalized version of the Strength Pareto Evolutionary Algorithm 2 (SPEA2). By replacing the algorithmic-internal role of the Pareto-dominance relation with a different, not necessarily transitive relation, the algorithm can become capable to search for the maximum set of the replacing relation. Thus, the SPEA2 algorithm can also become capable of on-line decision making during the evolutionary search. The approach will be exemplified by using the maxmin-fair relation among objective vectors that avoids extremities of a subset of a few objectives only. But without the Pareto-dominance relation, also the way of filling the archive of the SPEA2 algorithm has to be revised. The matter becomes complicated by the fact that the replacing relations might be only transitive in some subpspaces - as it is the case for the maxmin-fair relation. Different methods for maintaining an archive are proposed and experimentally compared. The results of the study suggest, for the case of a larger number of objectives, to use an archiving strategy, which is based on transitive subspaces of the replacing relation.

In this paper, we address the problem of how TCP sessions ramp up their transmission windows in a controlled way. We introduce a conservative slow start scheme that reduces to the regular slow start when session path queues are empty, but slows down the control window increase speed as buffers start to build up. We show via event driven simulation and open source based high speed experimental testbed the effectiveness of our conservative slow start mechanism in reducing packet losses and consequent retransmissions.

In the near future, mobile nodes (MNs) need to automatically select an optimal interface based on communication quality in a multimodal environment, that is, multimodal communication. While the existing handover management schemes avoid the communication termination, they can not satisfy (1) seamless communication and (2) selection of an optimal wireless network based on communication quality, and thus its performance during handover is drastically decreased. In our previous work, we proposed a new handover management scheme that employs (a) cross-layer approach and (b) multiple TCP connections to satisfy (1) and (2), and showed its effectiveness through simulation experiments. However, any other existing studies have not yet proposed a handover management scheme that employs both (a) cross-layer approach by exploiting beacon messages and (b) multiple TCP connections, and demonstrated the effectiveness of them on a real system. Thus, in this paper, we describe the implementation design of our proposed scheme in detail, and then show preliminary results for the fundamental performance of the proposed prototype system in a real wireless environment.

In ubiquitous networks based on Wireless Local Area Networks (WLANs), mobile nodes (MNs) will experience performance degradation at the handover. To achieve seamless and efficient communication, it is necessary for the MNs to satisfy the following three requirements: (I) initiation of han- dover processes based upon quick perception of the change in the wireless link quality, (II) elimination of a communication interruption due to handover processes, (III) selection of an optimal WLAN. In this paper, we propose a unified handover management scheme for TCP communication that can satisfy all of the above three requirements. Our proposed scheme employs both a cross-layer approach for obtaining the number of frame retransmissions to satisfy (I) and (III) and multi-homing to satisfy (II). Through simulation experiments, we demonstrate that our proposed scheme can quickly and reliably perceive the deterioration of the wireless link quality and select the optimal WLAN without communication interruption due to the handover process, thereby maintaining the best TCP performance at the handover. Index Terms— Seamless Handover, Cross-Layer, Multi- Homing, Frame Retransmission, TCP

A huge number of LAN switches are currently in use worldwide, so ways in which they can save energy are of great practical interest. A simple way to save power is to switch the interface to sleep mode when no packets are buffered, and to keep it in active mode while there are packets to be transmitted. This would appear to be the most effective energy saving scheme. However, mode switching gives rise to an in-rush current, which causes electrical damage to devices. This problem arises from over-frequent mode switching, which should be avoided. Thus, the main objective is to develop a way to reduce the number of mode switchings that result in quite short duration sleep modes, as these do not contribute so much to energy saving but lead to damage to the device. To this end, a method is adopted which keeps the interface in active mode for an extra period after all packets are flushed out from the buffer; this is the "extra active period". This scheme protects the device at the expense of energy saving efficiency. In this paper, the scheme is evaluated analytically in terms of its power reduction ratio and the frequency of mode changes, by modifying the M/M/1 and IPP/M/1 queuing models. The numerical results show how the duration of the extra active period degrades the energy saving performance while successfully reducing the number of mode changes. These results will enable LAN switches to be designed with energy saving in mind yet within the constraints of device characteristics.

This paper presents a topology reconstruction method to explore better trade-off points between search and access load balancing performance in unstructured peer-to-peer (P2P) file sharing networks. The proposed topology reconstruction method changes a network topology in a dynamic, autonomous, and decentralized manner. The topology reconstruction is based on local threshold-based rules that use query trails, which stand for information on previous successful search paths. A power-law network is used as the initial network in simulations. The simulation results show that, depending on the setting of the threshold values, compared to the case without topology reconstruction, the proposed method can explore better trade-off points between search and storage access load balancing performance.

As the amount of traffic transferee! on the Internet are increasing, dynamic traffic engineering (TE) becomes important to avoid link congestion. In open shortest path fast(OSPF)-based networks, link costs are statically set according to its long-term utilization for reducing traffic on some congested nodes, hence temporary performance degradation may occur due to short-term traffic fluctuation. For dynamic IE on OSPF-based networks, measurement of the utilization of links / nodes, inferring the set of link costs for improving transmission behavior and setting the cost set to routers are necessary and so-called on-line simulation (OLS) system can operate these functions autonomously and periodically. In this paper, we construct the server prototype in OLS system, and evaluate its scalability and control performance in our testbed network. Experimental results show that the server succeeds in providing low-cost network management and real-time control even if there is the large amount of traffic on the network. Furthermore, the total throughput over the network was greatly improved by the OLS.

A practical method of locating congested segments on the Internet by periodic end-to-end packet delay measurements along multiple paths is presented. This method is a network tomographic approach based on a clustering technique that effectively tackles the correlation among delay variations. Since the Internet is still highly prone to performance deterioration due to transient delays, locating the segments responsible is vital to ensure that Internet service providers can mitigate or prevent such performance deterioration. Our concern is how to detect and locate such segments promptly in a large network. The proposed method was implemented and evaluated through a real-world experiment by actively measuring packet delays from multiple origins to multiple destinations over the Japanese commercial Internet, and was shown to have considerable potential to precisely locate congested segments.

In the near future, wireless local area networks (WLANs) will overlap to provide continuous coverage over a wide area. In such ubiquitous WLANs, a mobile node (MN) freely moves between WLANs with different IP subnets during VoIP communication. In such situations, since an MN experiences several handovers, the communication quality is degraded. In previous studies, in order to solve this problem, we proposed a seamless handover scheme based on frame retransmissions and demonstrated its effectiveness through simulation experiments. However, no existing scheme has demonstrated the effectiveness of cross-layer architecture exploiting the number of frame retransmissions on a real system. In the present paper, therefore, we design and implement a handover scheme based on the number of frame retransmissions. In our implementation, we propose a cross-layer architecture using a shared memory to pass the information, i.e., the number of frame retransmissions, from MAC layer to Transport layer. Finally, we show preliminary results in a real wireless environment and evaluate the performance of the proposed prototype system in a simple topology.

The probability of transmission failure under a given wireless condition and instantaneous transmission rate is introduced as a new scheduling metric to improve the performance of existing scheduling algorithms. Scheduling algorithms for high-speed downlink packet access (HSDPA) in wideband code division multiple access (WCDMA) networks generally deal with the instantaneous transmission rate of each user as a key parameter reflecting the time-varying and locationdependent characteristics of the wireless channel. As a more detailed measure of the wireless condition of each user, the error rate metric improves the effectiveness of sharing resources without sacrificing a good balance between system performance and user fairness. Simulation results demonstrate that a performance gain of approximately 20% can be achieved for several well-known scheduling algorithms when the error rate is taken into account.

Multiple access points (APs) will be much more likely to be available for stations (STAs) due to the spread of wireless LANs. Within this context, we have developed a decentralized AP selection architecture for wireless LANs, and demonstrated that the proposed scheme can achieve an efficient and fair share of wireless access resources. Another issue regarding AP selection is the interference among both APs and STAs because channels partially overlap neighboring channels with the IEEE 802.11b/g standard. Therefore, we first discuss our investigations through experiments into how interference among both APs and STAs affects throughput. Based on these experiments, we propose a decentralized AP selection scheme that avoids interference, and demonstrate that the proposed solution can alleviate the effects of interference and improve throughput compared with the existing AP selection scheme. In addition, we implemented the proposed architecture on actual devices and added a function for avoiding interference. Lvaluations in terms of our scheme's effectiveness and feasibility revealed that the proposed architecture can actually improve the throughput of wireless LAN systems compared with the existing RSSI based scheme

Predicting wireless environment parameters as correctly and timely as possible is of practical importance to optimal resource allocation in wireless systems. For example, as the realtime wireless condition information is not available in reality due to delay, the latest historic wireless environment parameters are used to determine the users' instantaneous transmission rate in most of the existing systems in high-speed downlink packet access (HSDPA) networks. However, the wireless condition variation during this delay (e.g., 6 ms) sometimes results in unsuitable determination of the instantaneous transmission rate and eventually influences the user and system performance. It is shown by simulation results that compared with using the ideal realtime wireless environment information, using the latest available historic information will introduce about 10%-15% throughput performance damage averagely. Therefore, in this paper, in order to mitigate such performance degradation, a simple linear prediction of the current wireless environment parameters is introduced. Simulation results demonstrate that nearly 10% improvement can be achieved under several well-known scheduling algorithms in a HSDPA network when the predictive environment parameter is applied to determine the instantaneous transmission rate

Future pervasive networks will be constructed by various access networks, which will be managed by different operators. Host-based mobility mechanisms are promising for achieving efficient mobile computing in such an environment, because of their ability to handle the multihomed environment. However, the impact of switching network media will significantly degrade performance of communications with large RTT and high moving intensity. Therefore, we propose an adaptive handover scheme on host-based mobility to mitigate the impact of the switching media under such circumstances. This scheme switches the way of handover according to the network conditions. Furthermore, we demonstrate the effectiveness of the proposed scheme through simulation results.

Abstract In parallel and distributed applications, it is very likely that object‐oriented languages, such as Java and Ruby, and large‐scale semistructured data written in XML will be employed. However, because of their inherent dynamic memory management, parallel and distributed applications must sometimes suspend the execution of all tasks running on the processors. This adversely affects their execution on the parallel and distributed platform. In this paper, we propose a new task scheduling method called CP/MM (Critical Path/Memory Management) which can efficiently schedule tasks for applications requiring memory management. The underlying concept is to consider the cost due to memory management when the task scheduling system allocates ready (executable) coarse‐grain tasks, or macro‐tasks, to processors. We have developed three task scheduling modules, including CP/MM, for a task scheduling system which is implemented on a Java RMI (Remote Method Invocation) communication infrastructure. Our experimental results show that CP/MM can successfully prevent high‐priority macro‐tasks from being affected by the garbage collection arising from memory management, so that CP/MM can efficiently schedule distributed programs whose critical paths are relatively long. Copyright © 2005 John Wiley & Sons, Ltd.

Abstract IMT‐2000 has attracted much attention as the Next Generation Mobile Communication System. IMT‐2000 can provide high‐bit‐rate data communication service, so that a great number of packets conveyed by TCP (transmission control protocol) can be transmitted over a wireless link. W‐CDMA, standardized in 3GPP (which standardizes IMT‐2000), allows dynamically allocating transmission rates to flow over a wireless link in response to a changing FER for each flow, which is thought to be essential for a next generation mobile communication system. Therefore, in this paper, we study the characteristics of the dynamic allocation scheme when TCP flows share a wireless link, and, in particular, we focus on the throughput performance of these TCP flows. First, we use simulations to examine the effectiveness of the dynamically allocating down‐link transmission rate for TCP flows in response to changing the frame error rate (FER). Through the simulation results, we will show how it can improve the total throughput performance of TCP flows. Furthermore, we can obtain an effective way to allocate transmission rates to flows with different FERs in order to achieve high total throughput. Finally, we will deal with a case of multiple flows from a fixed host to a mobile host. In actual networks, this often happens. In this case, we will show that the total throughput of TCP flows degrades less than in the single‐flow case, even when the FER is high. Copyright © 2004 John Wiley & Sons, Ltd.

This paper proposes an analytical model for a delayed reservation scheme, which can be exactly analyzed. By applying it to a server-based QoS management network, we can obtain the related blocking probability and the waiting time distribution of requests, and discuss its performance by means of numerical results. The numerical results show that the delayed scheme can improve the blocking probability by about two orders of magnitude compared with the conventional reservation scheme even with a small acceptable waiting time (e.g. 20% of average flow duration). We can conclude that the delayed reservation scheme can lead to significant improvement in a server-based QoS management network performance.

The current Internet has a single class of service, i.e., best-effort service, whereas multimedia information is actually transmitted there. Recently, many researchers have focused on a mechanism to provide some quality of service (QoS) required for applications such as delay-sensitive ones. With such a mechanism, a real-time communication request is sent to a server, such as a policy server, that will try to find a path that meets the request's QoS requirement. If several available paths meet the requirement, the connection will be established on one of them. Otherwise, the request will be rejected. In all likelihood, though, delaying the path reservation to some extent if no available path meets the requirement, rather than immediately rejecting a request, will improve the blocking probability. We assume here that the communication can tolerate some initial setup delay, which we refer to as acceptable delay. Our major objective is to clarify whether or not the delay schemes are effective in improving the blocking probability. Therefore, we examine two schemes for managing buffered requests and two different network topologies, i.e., Mesh and ISP. Through an extensive simulation, we have found that the blocking probability can be improved even when a small acceptable delay and a small buffer size are assumed. In particular, we found that the scheme works well in actual network topologies such as an ISP model.

The logical configuration of a WDM-based network can be changed by reassigning wavelengths to tunable transmitters and/or receivers of nodes. The objective of network reconfiguration is to improve the network performance in terms of achievable maximum throughput. After obtaining an optimal node placement by using a rearrangement algorithm, we must actually reconfigure the network according to some reconfiguration procedure that defines a way logically to move nodes to optimal places. We study reconfiguration procedures for multihop WDM packet networks with non-negligible propagation delay between any pair of nodes. By means of simulation, we examine the impact of the propagation delay and the tuning time of receivers normalized by packet transmission time upon the maximum total delay and the number of lost packets during reconfiguration. Moreover, we introduce some mechanisms for avoiding packet loss during reconfiguration. We then compare the performance of the procedures with these mechanisms.

Most of the QoS-based routing schemes proposed so far focus on improving the performance of individual service classes. In a multi-class network, where high priority QoS traffic coexists with best-effort traffic, routing decision for QoS sessions will have an effect on lower ones. A mechanism that allows dynamic link sharing among multiple classes of traffic is needed. We propose a multi-class routing algorithm based on inter-class sharing of resources among multiple class of traffic. Our algorithm is based on the concept of the "virtual residual bandwidth", which is derived from the real residual bandwidth. The virtual residual bandwidth is greater than the residual bandwidth when the load of lower priority traffic is light and smaller when the load of lower priority traffic is heavy. The idea of our approach is simple since the routing algorithm for individual traffic does not change, the only change being the definition of the link cost. We demonstrate, through extensive simulation, the effectiveness of our approach when the best effort distribution is uneven and when its load is heavy. Also, better performance is noticed when using a topology with a large number of alternative paths.

LSR (label switching router)s in MPLS (multiprotocol label switching) networks map arriving IP flows into some labels on the layer 2 switching fabric and establish LSP (label switching path)s. By using LSPs, LSRs not only transmit IP datagrams fast with a cut-through mechanism, but also solve the traffic engineering issue to optimize the delay of some IP datagram flows. So far, we have analyzed the performance of LSR focusing only on the maximum number of LSPs which can be set on layer 2. We also consider the bandwidth allocated to each LSP and analyze the IP datagram transmission delay and the cut-through rate of LSR. We suppose the label mapping method as the data-driven scheme in the analytical model, so that the physical bandwidth of LSR is shared by both the default LSP for hop-by-hop transmission and the cut-through LSPs. Thus, we investigate the impact of the bandwidth allocation among these LSPs on the performance.

In the Internet, because of huge scale and distributed administration, it is of practical importance to infer network-internal characteristics that cannot be measured directly. We propose a general method of determining characteristics of links from given characteristics of end-to-end paths. Our method can be applied to an arbitrary path-topology. Furthermore we show the general conditions that the link characteristics to be inferred must satisfy. Packet loss and queuing delay time are shown to satisfy them. Case studies which our method can treat are also provided.

We examine the delay performance of packets from constant-bit-rate (CBR) traffic whose delay is affected by non-real-time traffic. The delay performance is analyzed by solving the nD/D/1 queue with vacations. We obtain an exact and closed form solution, hence obviating the need of any approximations or numerical Laplace inversions. We then provide various numerical results for low-bit-rate transmission links, in which packets can experience large delay. From our quantitative evaluation, we conclude that there exists an optimum packet size for a given delay bound. In extremely slow links, such as modem links, transmission control protocol (TCP) packets should be segmented to reduce the CBR delay. We therefore investigate the delay impact of TCP packet sizes as well.

The logical configuration of a WDM network can be changed by reassigning wavelengths to tunable transceivers of nodes. In this paper, we develop reconfiguration algorithms to obtain an optimal node placement of multihop networks with regular topology. The algorithms are based on meta-heuristics such as simulated annealing, threshold acceptance, and multistart local search. We compare the performance of these algorithms in terms of their calculation time required to reach a solution with accuracy of some given degree.

In long haul networks, delay in congestion notification can cause an extended period of congestion which may in turn result in cell loss, and poor utilization of the network. In this regard we investigate various congestion notification strategies and show our results in this paper. Moreover, we identify deficiencies and problems associated with the enhanced proportional rate control algorithm (EPRCA) which is one of the explicit rate control algorithms recommended by the ATM Forum. These deficiencies cause unfairness in bandwidth utilization among contending sources. We suggest modifications in EPRCA and call it the modified enhanced proportional rate control algorithm (EPRCAM). We argue that EPRCAM will lead to better link utilization and fair bandwidth allocation between sources. In our simulation model, EPRCAM results in as high as a 97.5% link utilization without cell loss.

Wavelength division multiplexing (WDM) is a promising technology for high-speed communication. In WDM-based multihop networks, the logical network configuration can be changed by reassignment of the wavelengths to transmitters or receivers of the nodes. We propose some reconfiguration procedures to move the nodes logically according to the result from the reconfiguration algorithms, and evaluate their performance by means of simulation. Through simulations, we examine the impact of the tuning time of tunable receivers on the transient packet transmission delay in the reconfiguration procedures, and then compare the performance of the procedures.

This paper presents a new architecture of optical cross-connect (OXC) system to integrate various kinds of transmission systems like wavelength division multiplex (WDM) transmission systems, ultra short pulse transmission systems etc. This architecture also has the following advantages described in a lot of previous studies: providing a large number of lightwave paths with a small number of required wavelength converters; network failure restoration with desirable features (compactness, small power consumption, and bit rate independency); service integration. In this paper, we evaluate impact of our architecture, especially with regard to the number of required components like wavelength converters (WCNVs) and wavelength multiplexers/de-multiplexers (WMuxs/WDMuxs) to allocate lightwave paths.

In transport-layer protocols such as TCP over ATM networks, a packet is discarded when one or more cells of that packet are lost, and the destination node then requires its source to retransmit the corrupted packet. Therefore, once one of the cells constituting a packet is lost, its subsequent cells of the corrupted packet waste network resources. Thus, discarding those cells will enable us to efficiently utilize network resources, and will improve the packet loss probability. We focus on tail dropping (TD) and early packet discard (EPD) as selective cell discard schemes which enforce the switches to discard some of the arriving cells instead of relaying them. We exactly analyze the packet loss probability in a system applying these schemes. Their advantages and limits are then discussed based on numerical results derived through the analysis.

In transport-layer protocols such as TCP over ATM networks, a packet is discarded when one or more cells are lost in that packet, and the destination node then requires its source to retransmit the corrupted packet. Therefore, once one of the cells constituting a packet is lost, the subsequent cells of the corrupted packet waste network resources. Thus, discarding those cells will enable us to efficiently utilize the network resources and improve the packet loss probability. We focus on tail dropping (TD) and early packet discard (EPD) as selective cell discard schemes which force the switches to discard some of the arriving cells instead of relaying them. We exactly analyze the packet loss probability in a system applying these schemes. Their advantage and limits are then discussed based on numerical results derived through the analysis.

In ATM networks, preventive congestion control is widely recognized for efficiently avoiding congestion, and it is implemented by a conjunction of connection admission control and usage parameter control. However, congestion may still occur because of unpredictable statistical fluctuation of traffic sources even when preventive control is performed in the network. The authors study another kind of congestion control, i.e., reactive congestion control, in which each source changes its cell emitting rate adaptively to the traffic load at the switching node (or at the multiplexer). The intention is that, by incorporating such a congestion control method in ATM networks, more efficient congestion control is established. They develop an analytical model, and carry out an approximate analysis of reactive congestion control algorithm. Numerical results show that the reactive congestion control algorithms are very effective in avoiding congestion and in achieving the statistical gain. Furthermore, the binary congestion control algorithm with push-out mechanism is shown to provide the best performance among the reactive congestion control algorithms treated.>

In almost all conventional transport protocols, ARQs (automatic repeat request) have been used to recover lost packets, in which receivers would request sources to retransmit lost packets. In these retransmission based error recovery schemes, the transmission delay can reach an unacceptable level for traffic with stringent delay constraints in high speed networks because their propagation delay is very large relative to packet transmission times. On the other hand, FEC (forward error correction) adds redundant packets to data ones, all of which are organized into a group called a block, and hence allows the receivers to correct some of the lost packets without retransmissions. This capability is desirable for high speed networks. The authors exactly analyze the distribution of the number of cells lost within a block in ATM networks with correlated cell arrivals, and examine the benefits due to FEC by means of the analysis.>

The authors consider a three-stage switching configuration with no internal buffers, i.e., bufferless switches are used at the first and second stages, and output buffered switches at the third stage. Short-term cell loss probability is analyzed in order to examine the influence of bursty traffic on the performance of the bufferless switches used at the first two stages. A four-stage switching configuration with traffic distributors added at the first stage is proposed. This switch provides more paths between a pair of input and output ports than the three-stage configuration. Some schemes for distributing cells are compared. It is shown that the distributor successfully reduces cell loss probability due to bursty traffic by splitting incoming cells into several switching modules.>

The performance of nonblocking packet switches such as the knockout switch and Batcher banyan switch for high-speed communication networks can be improved as the switching capacity L per output increases; the switching capacity per output refers to the maximum number of packets transferred to an output during a slot. The N*N switch with L=N was shown to attain the best possible performance by M.J. Karol et al. (1987). Here a N*N nonblocking packet switch with input and output buffers is analyzed for an arbitrary number of L such that 1>

The major ATM switch architectures and their improvement techniques are categorized and discussed, focusing on performance issues. The performance measures of interest are the maximum throughput, the delay time, and the cell loss probability. The major assumptions and notations used are summarized. A survey is presented of nonblocking switches and their improvement techniques. The performances of a variety of nonblocking switches are compared.>

An exact analysis is presented for the 2*2 switching element with input port buffers that use the cell bypass queueing discipline. An approximate analysis is developed for the Banyan network composed of 2*2 switching elements with buffers that use the cell bypass queuing discipline. An investigation is made of the effect of this discipline on performance through numerical results obtained by analyses as well as simulation results. The authors also examine the nonuniform input traffic models. The analysis shows that the cell bypass queuing discipline considerably improves the performance of the ATM (asynchronous transfer mode) switches based on Banyan networks with respect to both throughput and cell blocking probability.>

The performance of an asynchronous transfer mode (ATM) network subject to admission control and traffic smoothing is analyzed. Basically, an ATM switch is modeled as a discrete-time single-server queuing system in which a new call joins existing calls. Cell arrivals from a new call are assumed to follow a general distribution. It is also assumed that aggregated arrivals of cells from existing calls form batch arrivals with a general batch size distribution and a geometric distribution of the interarrival times between batches. Both finite- and infinite-buffer cases are considered. An exact analysis yields the waiting time distribution and cell loss probability for a new call and for existing calls. Numerical examples are given to show how the network performance depends on the statistics of a new call (burstiness, time that a call stays in an active or inactive state, etc.) and to demonstrate the effectiveness of admission control and traffic smoothing.>

An analysis is made of the throughput and delay performance of two classes of free-access tree algorithms with minislots. In one class, binary feedback information is available in minislots, and in the other, ternary feedback information is available. It is shown that the highest maximum throughput 0.56714 is achieved in the limiting case where the number of minislots in a (large) slot is infinity and minislot overhead is zero. A lower bound of the average transmission delays in these algorithms is analytically derived. The obtained lower bound is also a lower bound of the average delay of the whole class of the free-access algorithms.>