The Available Bit Rate (ABR) service of Asynchronous Transfer Mode (ATM) networks employs explicit rate notification algorithms to ensure better and fair distribution of available bandwidth among contending sources. The Enhanced Proportional Rate Control Algorithm (EPRCA) is one of the explicit rate control algorithms recommended by the ATM forum. In this paper, we identify deficiencies and problems associated with EPRCA and show that these cause unfairness in bandwidth utilization by the contending sources. We propose modification in EPRCA and call it Modified Enhanced Proportional Rate Control Algorithm (EPRCAM). We will argue and show through simulation results that EPRCAM leads to better link utilization and fair bandwidth allocation among contending sources. In our simulation model, EPRCAM results in as high as 97. 8% link utilization without cell loss.

Two congestion control schemes designed specifically to handle changes in the datalink interface of a mobile host are presented. The future mobile environment is expected to involve multimode connectivity to the Internet and dynamic switching of the connection mode depending on network conditions. The conventional Transmission Control Protocol (TCP), however, is unable to maintain stable and efficient throughput across such interface changes. The two main issues are the handling of the change in host Internet Protocol (IP) address, and the reliability and continuity of TCP flow when the datalink interface changes. Although existing architectures addressing the first issue have already been proposed, the problem of congestion control remains. In this paper, considering a large change in bandwidth when the datalink interface changes, two new schemes to address these issues are proposed. The first scheme, Immediate Expiration of Timeout Timer, detects interface changes and begins retransmission immediately without waiting for a retransmission timeout as in existing architectures. The second scheme, Bandwidth-Aware Slow Start Threshold, detects the interface change and estimates the new bandwidth so as to set an appropriate slow start threshold for retransmission. Through simulations, the proposed schemes are demonstrated to provide marked improvements in performance over existing architectures.

Rate based control is a promising way to achieve an efficient packet transmission especially in high speed packet switching networks where round trip delay is much larger than packet transmission time. Although inappropriate tuning for the parameters, increasing and decreasing factors, of the rate control function causes the performance degradation, most of the previous works so far have not studied the effect of the parameters on the performance. In this paper, we investigate the effect of the rate control parameters on the throughput under the condition that the packet loss probability is kept below a specific value, say 10-6. For this purpose, we build a queueing model and carry out a transient analysis to examine the dynamic behavior of the queue length at an intermediate node in a high speed network suffering from large propagation delay. Numerical examples exploit the optimal value of the parameters when one or two source-destination pairs transmit packets. We also discuss the effect of the propagation delay on the performance. Our model can be applicable to investigate the performance of various kinds of rate-based congestion control when the relation between the congestion measure and the rate control mechanism is given explicitly.

An immense number of LAN switches are currently in use worldwide. Therefore, methods that can reduce the energy consumption of these devices are of great practical interest. A simple way to save power in LAN switches is to switch the interfaces to sleep mode when no packets are buffered and to keep the interfaces in active mode while there are packets to be transmitted. Although this would appear to be the most effective energy saving scheme, mode switching gives rise to in-rush current, which can cause electrical damage to devices. This problem arises from excessive mode switching, which should be avoided. Thus, the main objective is to develop a method by which to reduce the number of mode switchings that result in short-duration sleep modes because these switchings do not contribute greatly to energy efficiency but can damage the device. To this end, a method is adopted whereby the interface is kept in active mode for an "extra" period of time after all packets have been flushed from the buffer. This period is the "extra active period (EAP)" and this scheme protects the device at the expense of energy saving efficiency. In this paper, this scheme is evaluated analytically in terms of its power reduction ratio and 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 reducing the number of mode changes. We analytically show an exact trade-off between the power reduction ratio and the average number of turn-ons in the EAP model with Poisson packet arrival. Furthermore, we extend the scheme to determine the EAP dynamically and adaptively depending on the short-term utilization of the interface and demonstrate the effectiveness of the extended scheme by simulation. The newly developed scheme will enable LAN switches to be designed with energy savings in mind without exceeding the constraints of the device.

Congestion inherently occurs on the Internet due to traffic concentration on certain nodes or links of networks. The traffic concentration is caused by inefficient use of topological information of networks in existing routing protocols, which reduces to inefficient mapping between traffic demands and network resources. Actually, the route with minimum cost, i.e., number of hops, selected as a transmission route by existing routing protocols would pass through specific nodes with common topological characteristics that could contribute to a large improvement in minimizing the cost. However, this would result in traffic concentration on such specific nodes. Therefore, we propose a measure of the distance between two nodes that is suitable for reducing traffic concentration on specific nodes. To consider the topological characteristics of the congestion points of networks, we define node-to-node distance by using a generalized norm, p-norm, of a vector of which elements are degrees of intermediate nodes of the route. Simulation results show that both the maximum Stress Centrality (SC) and the coefficient of variation of the SC are minimized in some network topologies by selecting transmission routes based on the proposed measure of node-to-node distance.

W-CDMA (Wideband-CDMA) is expected to play a significant role in the radio access technology of third-generation mobile telecommunication systems. In second-generation systems, voice traffic from each user has been transmitted mainly via the dedicated transport (radio) channel. In addition, the third-generation systems will efficiently accommodate data traffic based on packet transmission in the shared common transport channel. Therefore, data traffic can be transmitted via one of two types of data channels: i.e., dedicated channels or common channels. However, the channel selecting/switching scheme has not been standardized; thus, system architectures and algorithms of channel-switching schemes in the RNC (Radio Network Controller) are dependent on its vendors, and network operators must determine the parameter settings related to channel selection. In this paper, we will deal with aspects of the architecture in detail, and propose possible algorithms for channel selecting/switching for fundamental reference systems which meet the specifications of the RNC. We will then evaluate our algorithms by means of simulations, and discuss the impact of parameter settings on performance, in terms of packet loss probability and utilization of dedicated channels.

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. In this paper, based on a general framework we proposed previously, we present a feasible method of inferring packet loss rates of individual links from end-to-end measurement of unicast probe packets. Compared with methods using multicast probes, unicast-based inference methods are more flexible and widely applicable, whereas they have a problem with imperfect correlation in concurrent events on paths. Our method can infer link loss rates under this problem, and is applicable to various path-topologies including trees, inverse trees and their combinations. We also show simulation results which indicate potential of our unicast-based method.

In parallel computing systems using the master/worker model for distributed grid computing, as the size of handling data grows, the increase in the data transmission time degrades the performance. For divisible workload applications, therefore, multiple-round scheduling algorithms have been being developed to mitigate the adverse effect of longer data transmission time by dividing the data into chunks to be sent out in multiple rounds, thus overlapping the times required for computation and transmission. However, a standard multiple-round scheduling algorithm, Uniform Multi-Round (UMR), adopts a sequential transmission model where the master communicates with one worker at a time, thus the transmission capacity of the link attached to the master cannot be fully utilized due to the limits of worker-side capacity. In the present study, a Parallel Transferable Uniform Multi-Round algorithm (PTUMR) is proposed. It efficiently utilizes the data transmission capacity of network links by allowing chunks to be transmitted in parallel to workers. This algorithm divides workers into groups in a way that fully uses the link bandwidth of the master under some constraints and considers each group of workers as one virtual worker. In particular, introducing a Grouping Threshold effectively deals with very heterogeneous workers in both data transmission and computation capacities. Then, the master schedules sequential data transmissions to the virtual workers in an optimal way like in UMR. The performance evaluations show that the proposed algorithm achieves significantly shorter turnaround times (i.e., makespan) compared with UMR regardless of heterogeneity of workers, which are close to the theoretical lower limits.

We present a TCP flow level performance evaluation on error rate aware scheduling algorithms in Evolved UTRA and UTRAN networks. With the introduction of the error rate, which is the probability of transmission failure under a given wireless condition and the instantaneous transmission rate, the transmission efficiency can be improved without sacrificing the balance between system performance and user fairness. The performance comparison with and without error rate awareness is carried out dependant on various TCP traffic models, user channel conditions, schedulers with different fairness constraints, and automatic repeat request (ARQ) types. The results indicate that error rate awareness can make the resource allocation more reasonable and effectively improve the system and individual performance, especially for poor channel condition users.

The monitoring of performance in VoIP traffic has become vital because users generally expect VoIP service quality that is as high as that of PSTN services. A lightweight method of processing by extracting VoIP flows from Internet traffics is proposed in this paper. Estimating delay variations and the packet loss ratio using knowledge about specific features and the characteristics of VoIP flows, i.e., the inter-packet gap (IPG) which is constant in VoIP flows, is also proposed. Simulation with actual traffic trace is used to evaluate the method, and this revealed that delay variations (IPG variance) can be accurately estimated by monitoring only a few percentage of all flows. The proposed method can be used as a first-alert tool to monitor large amounts of flows to detect signs of degradation in VoIP flows. The method can be used by ISPs to estimate whether VoIP flow performance is adequate within their networks and at ingress from other ISPs.

Wireless LANs (Local Area Networks) are currently spreading over diverse places such as hotels and airports, as well as offices and homes. Consequently, they provide convenient and important ways to access the Internet. Another type of communication model, Peer-to-Peer (P2P) communication on the Internet, has also attracted much attention, and P2P over wireless LANs will soon be very common. There are concerns about the capability of wireless stations (STAs) to send a large amount of traffic on an uplink. In this paper, we first clarify some issues that arise in this context by examining the feature of the Access Point(AP). Furthermore, we consider the role of the AP and propose, as a solution, ways of enabling both efficient and fair transmission over both the downlink and uplink. We evaluate the proposed schemes through simulations and show that communications over the uplink and downlink can share the wireless LAN access resource effectively. Furthermore, we show that coordinating our solution with the 802.11e Enhanced Distributed Channel Access (EDCA) can meet the different requirements of various types of traffic.

In order to prevent the degradation of TCP performance while traversing two WLANs, we present an implementation design of an inter-domain TCP handover method based on cross-layer and multi-homing. The proposed handover manager (HM) in the transport layer uses two TCP connections previously established via two WLANs (multi-homing) and switches the communication path between the two connections according to the handover trigger and the comparison of new/old APs. The handover trigger and comparison are conducted by assessing the wireless link quality using the frame-retry information obtained from the MAC layer (cross-layer). In a previous study, we proposed a preliminary concept for this method and evaluated its functional effectiveness through simulations. In the present study, we design an implementation considering a real system and then examine the effective performance in a real environment because a real system has several system constraints and suffers from fluctuations in an actual wireless environment. Indeed, depending on the cross-layer design, the implementation often degrades the system performance even if the method exhibits good functional performance. Moreover, the simple assessments of wireless link quality in the previous study indicated unnecessary handovers and inappropriate AP selection in a real environment. Therefore, we herein propose a new architecture that performs cross-layer collaboration between the MAC layer and the transport layer while avoiding degradation of system performance. In addition, we use a new assessment scheme of wireless link quality, i.e., double thresholds of frame retry and comparison of frame retry ratio, in order to prevent handover oscillation caused by fluctuations in the wireless environment. The experimental results demonstrate that the prototype system works well by controlling two TCP connections based on assessments of wireless link quality thereby achieving efficient inter-domain TCP handover in a real WLAN environment.

We propose two types of autonomic and distributed cooperative behaviors of peers for peer-to-peer (P2P) file-sharing networks. Cooperative behaviors of peers are mediated by query trails, and allows the exploration of better trade-off points between file search and storage load balancing performance. Query trails represent previous successful search paths and indicate which peers contributed to previous file searches and were at the same time exposed to the storage load. The first type of cooperative behavior is to determine the locations of replicas of files through the medium of query trails. Placement of replicas of files on strong query trails contributes to improvement of search performance, but a heavy load is generated due to writing files in storage to peers on the strong query trails. Therefore, we attempt to achieve storage load balancing between peers, while avoiding significant degradation of the search performance by creating replicas of files in peers adjacent to peers on strong query trails. The second type of cooperative behavior is to determine whether peers provide requested files through the medium of query trails. Provision of files by peers holding requested files on strong query trails contributes to better search performance, but such provision of files generates a heavy load for reading files from storage to peers on the strong query trails. Therefore, we attempt to achieve storage load balancing while making only small sacrifices in search performance by having peers on strong query trails refuse to provide files. Simulation results show that the first type of cooperative behavior provides equal or improved ability to explore trade-off points between storage load balancing and search performance in a static and nearly homogeneous P2P environment, without the need for fine tuning parameter values, compared to replication methods that require fine tuning of their parameters values. In addition, the combination of the second type and the first type of cooperative behavior yields better storage load balancing performance with little degradation of search performance. Moreover, even in a dynamic and heterogeneous P2P environment, the two types of cooperative behaviors yield good ability to explore trade-off points between storage load balancing and search performance.

In a Peer-to-Peer (P2P) network, in order to improve the search performance and to achieve load balancing, replicas of original data are created and distributed over the Internet. However, the replication methods which have been proposed so far focus only on the improvement of search performance. In this paper, we examine the load on the storage systems, which is due to writing and reading, and propose two replication methods for balancing the load on the storages distributed over P2P networks while limiting the degradation of the search performance within an acceptable level. Furthermore, we investigate the performance of our proposed replication methods through computer simulations, and show their effectiveness in balancing the load.

In this paper we propose a file replication scheme inspired by a thermal diffusion phenomenon for storage load balancing in unstructured peer-to-peer (P2P) file sharing networks. The proposed scheme is designed such that the storage utilization ratios of peers will be uniform, in the same way that the temperature in a field becomes uniform in a thermal diffusion phenomenon. The proposed scheme creates replicas of files in peers probabilistically, where the probability is controlled by using parameters that can be used to find the trade-off between storage load balancing and search performance in unstructured P2P file sharing networks. First, we show through theoretical analysis that the statistical behavior of the storage load balancing controlled by the proposed scheme has an analogy with the thermal diffusion phenomenon. We then show through simulation that the proposed scheme not only has superior performance with respect to balancing the storage load among peers (the primary objective of the present proposal) but also allows the performance trade-off to be widely found. Finally, we qualitatively discuss a guideline for setting the parameter values in order to widely find the performance trade-off from the simulation results.

In QoS networks, routing algorithms for QoS traffic have to provide the transmission path satisfying its QoS requirement while achieving high utilization of network resources. Therefore, server-based QoS routing algorithms would be more effective than distributed routing ones which are very common on the Internet. Furthermore, we believe that rerouting function enhances the advantage of their algorithms in which an already accepted flow with established path is replaced on some other path in order to accept newly arriving transmission request if it can not be accepted without doing so. Thus in this paper, we will propose a rerouting algorithm with the server-based QoS routing and evaluate its performance in terms of the blocking probability by computer simulation. In addition, we will investigate the impact of the amount of traffic with high-priority on the performance in some network topologies. Through some simulation results, we also discuss some issues arising in improving the effectiveness of rerouting.

Recent improvements in the performance of end-computers and networks have made it feasible to construct a grid system over the Internet. A grid environment consists of many computers, each having a set of components and a distinct performance. These computers are shared among many users and managed in a distributed manner. Thus, it is important to focus on a situation in which the computers are used unevenly due to decentralized management by different task schedulers. In this study, which is a preliminary investigation of the performance of task allocation schemes employed in a decentralized environment, the average execution time of a long-lived task is analytically derived using the M/G/1-PS queue. Furthermore, assuming a more realistic condition, we evaluate the performance of some task allocation schemes adopted in the analysis, and clarify which scheme is applicable to a realistic grid environment.

In this paper, we discuss configurations of photonic ATM (Asynchronous Transfer Mode) switches and their advantages in terms of the number of optical switching devices to be implemented on the system, the number of wavelengths, throughput, broadcast function etc. In particular, we focus on photonic ATM switch architectures which can be built in the near future; that is, with presently available optical and electrical devices. For example, we assume the optical devices such as optical gate switches with 40 dB on/off ratio. In this context, we evaluate 17 types of photonic ATM switches; they are 6 types of input buffer type switches, 6 types of output buffer type switches, 4 types of shared buffer switches, and 1 proposed type. From our evaluation, for cell switching, wavelength division switching technologies are desirable compared with space division switching technologies in the sense that the former enables us to build a photonic ATM switch with the less number of optical gate switches. Furthermore, we propose a switch architecture equipped with optical delay line buffers on outputs and electric buffers on inputs. We show that our switch architecture is superior in the number of required optical gate switch elements under the given conditions.

As the Internet grows, various types of traffic, such as voice, video and data, are transmitted. Therefore, the Internet should provide the Quality of Service (QoS) required by each type of traffic as well as end-to-end connectivity, and routing decisions should be based on the utilization of links/routers and/or the application types of traffic. This kind of routing is called Traffic Engineering (TE), and its objective is to improve such performance factors as flow loss probability for users and the utilization of links for networks, simultaneously. Some studies claim that the Multi-Protocol Label Switching (MPLS) technique can easily implement TE. So far, some experimental results show that TE is effective on a MPLS network; however, its performance has not been theoretically and quantitatively analyzed. Thus, in this paper, we will investigate the basic and preliminary performance of MPLS networks with TE by analyzing flow loss probability and Smoothness index of link utilization in the queueing system.

The recent evolution on the network tomography have successfully provided principles and methodologies of inferring network-internal (local) characteristics solely from end-to-end measurements, which should be followed by deployment in practical use. In this paper, two kinds of user-oriented tools for inferring one-way packet losses based on the network tomography are proposed. They can infer one-way packet loss rates on paths or path segments from/to a user-host (a client) to/from a specified target host (an application server or a router) without any measurement on the target, and thus can find the congested area along the path between the client and an application server. One is a stand-alone tool running on the client, and the other is a client-server style tool running on both the client and a proxy measurement server distributed in the Internet. Prototypes of the tools have been developed and evaluated by experiments in the actual Internet environment, which shows that the tools can infer the loss rates within 1% errors in various network conditions.