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.

The layer3 switch enables us to fast transmit IP datagrams using the cut-through technique. The current layer3 router would become bottleneck in terms of delay performance as the amount of traffic injected into high speed networks gets relatively large. Thus, the layer3 switch should be an important element constructing the next generation Internet backbone. In this paper, we analyze the cut-through rate, the datagram waiting time and the mis-ordered rate of a layer3 switch in case of flow-driven connection setup. In the analysis, by using 3-state Markov modulated Bernoulli process (MMBP), we model the arrival process of IP flow and IP datagram from each source. Furthermore, we investigate impacts of the arrival rate and the average datagram length on the performance.

Central switches are expected to operate at the rate of Terabit per second in high speed networks, like the B-ISDN. Photonic switches using lightwave technology based on wavelength division multiplexing (WDM) and frequency division multiplexing (FDM) are promising ones for high speed switching. Such lightwave networks are mainly divided into two groups, according to the number of hops required for packets to arrive at their destinations: single-hop networks such as networks using star coupler and multihop networks such as Manhattan Street Network and ShuffleNet. In this paper we focus our attention on multihop networks and propose a mesh network, referred to as RookNet, for high speed communication. The average transmission delay time and maximum throughput of RookNet is approximately analyzed. It is shown that, as the number of nodes goes to infinity, the maximum throughput aproaches 0.433 and 0.485 when each node is equipped with no internal buffer and internal buffers of infinite capacity for relayed packets, respectively.

We propose an anomaly detection method for finding patterns in network traffic that do not conform to legitimate (i.e., normal) behavior. The proposed method trains a baseline model describing the normal behavior of network traffic without using manually labeled traffic data. The trained baseline model is used as the basis for comparison with the audit network traffic. This anomaly detection works in an unsupervised manner through the use of time-periodic packet sampling, which is used in a manner that differs from its intended purpose – the lossy nature of packet sampling is used to extract normal packets from the unlabeled original traffic data. Evaluation using actual traffic traces showed that the proposed method has false positive and false negative rates in the detection of anomalies regarding TCP SYN packets comparable to those of a conventional method that uses manually labeled traffic data to train the baseline model. Performance variation due to the probabilistic nature of sampled traffic data is mitigated by using ensemble anomaly detection that collectively exploits multiple baseline models in parallel. Alarm sensitivity is adjusted for the intended use by using maximum- and minimum-based anomaly detection that effectively take advantage of the performance variations among the multiple baseline models. Testing using actual traffic traces showed that the proposed anomaly detection method performs as well as one using manually labeled traffic data and better than one using randomly sampled (unlabeled) traffic data.

We have focused on the RIO queueing mechanism in statistical bandwidth allocation service, which uses AF-PHB. We have studied the parameterization of RIO to achieve both high throughput and low delay. We were able to parameterize RIO for that purpose in terms of both minth and maxp used in dropping OUT packets. Furthermore, we have also examined the parameterization regarding EWMA (Exponential Weighted Moving Average), i.e., weight factor wqout, and have shown that dropping OUT packets should depend upon the queue length without much delay unlike in RED. From our simulation results, we could see that our parameterization provided high throughput performance and also limited the queue length in a narrow range more effectively.

LSR (Label Switching Router)s in MPLS (Multiprotocol Label Switching) networks map arriving IP flows into some labels on Layer 2 switching fabric and establish LSP (Label Switching Path)s. By using LSPs, LSRs not only transmit IP datagrams fast by cut-through mechanism, but also solve 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. In this paper, we will 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 will investigate the impact of the bandwidth allocation among these LSPs on the performance.

A basic and inevitable problem in estimating flow duration distribution arises from "censoring" (i.e., cutting off) the observed flow duration because of a finite measurement period. We extended the Kaplan-Meier method, which is used in the survival analysis field, and applied it to recover information on the flow duration distribution that was lost due to censoring. We show that the flow duration distribution from a short period of actual traffic data with censoring that was estimated using a Kaplan-Meier-based method can approximate well the flow duration distribution calculated from a sufficiently long period of actual traffic data.

Differentiated Service (DiffServ) is a technology designed to provide Quality of Service (QoS) in the Internet, and is superior to Integrated Service (IntServ) technology with respect to the simplicity of its architecture and the scalability of networks. Although various simulation studies and estimations over testbeds have investigated the QoS that is offered via the DiffServ framework, almost all of them focused on the characteristics in a single DiffServ domain. However, the Internet is actually composed of a large number of AS domains, and thus packets are very likely to arrive at their destinations after going through many different domains. Therefore, we have analyzed the QoS performance in a model consisting of multiple DiffServ domains, and focused especially on the quality provided by Assured Forwarding Service (AF) to achieve statistical bandwidth allocation with AF-PHB. Our simulation results show some throughput characteristics of flows over multiple Diffserv domains, which clarify how network configurations impact the QoS over multiple DiffServ domains.

Multiple access points (APs) are much more likely to be available for stations (STAs) due to the popularity of wireless LANs. The serious issue of how an appropriate AP is selected from those that are available in a wireless LAN therefore arises. We discuss the development of a decentralized architecture for selecting APs, and examine its fundamental characteristics. The proposed architecture should be introduced without adversely affecting the performance of the existing common architecture that is currently being deployed. Therefore, the deployability of our architecture is examined in this respect. Furthermore, the dynamic behavior of the proposed architecture is studied in addition to static characteristics to evaluate its robustness against various dynamic changes in situation due to AP breakdowns and bursty arrivals of STAs. Simulations revealed that the proposed architecture can attain excellent performance in all the cases treated here.

The performance of a real-time networked application can be drastically affected by delays in packets traversing the network. Some real-time applications impose limits for acceptable network delay, and so a packet which is delayed longer than the limit before arriving at its destination is worthless to the flow to which the packet belongs. Not only that, but the rejected packet is also damaging to the quality of other flows in the network, because it may increase the queuing delay for other packets. Therefore, this paper proposes an adaptive scheme using two mechanisms, in which packets experiencing too great a delay are discarded at intermediate nodes based on the delay limit for the application and the delay experienced by each packet. This earlier discarding of packets is expected to improve the overall delay performance of real-time flows competing for network resources when the network is congested. An extensive simulation is conducted, and the results show that the scheme has great potential in improving the delay performance of real-time traffic in both homogeneous and heterogeneous environments in terms of traffic volume and application delay requirements.

As the Internet role changes from the experimental environment to the social infrastructure, end-to-end quality for data transfer of various types of traffic has been required as well as its connectivity. So we should precisely measure some performance such as packet loss probability and delay at routers on some source-destination path. By using so-called passive measurement technique, we can get the queue length distribution from some routers individually and estimate the end-to-end transmission delay. However, there may be some correlation between queue lengths of two or more routers packets go through in sequence, which would lead to inaccurate estimation of end-to-end delay performance. Thus in this paper, we model two tandem routers as queueing system, and analyze the queue length distributions and their correlation. Through some numerical results, we will investigate the impact of traffic parameters on the degree of correlation and how it affects the estimation of delay performance.

A transport layer mobility management scheme for handling seamless handoffs between appropriate networks is presented. The future mobile environment will be characterized by multimodal connectivity with dynamic switching. Many technologies have been proposed to support host mobility across diverse wireless networks, and operate in various layers of the network architecture. Our major focus is on the transport protocol that recovers packets lost during handoffs and controls transmission speed to achieve efficient communication. Majority of the existing technologies can maintain the connection by updating the information of a single connection around a handoff. Moreover, none of the studies extensively examine the handoff latencies and focus how an appropriate network is selected, during the handoff. In this paper, we first extensively investigate the various handoff latencies and discuss the limited performance of existing technologies based on the single connection. We then propose a new scheme resolving the problems by the transport protocol enabling the adaptive selection of an appropriate interface based on communication condition among all available interfaces. Finally, we demonstrate that the proposed scheme promptly and reliably selects the appropriate interface, and achieves excellent goodput performance by comparing with the existing technologies.

We present the concept of folksonomical peer-to-peer (P2P) file sharing networks that allow participants (peers) to freely assign structured search tags to files. These networks are similar to folksonomies in the present Web from the point of view that users assign search tags to information distributed over a network. As a concrete example, we consider an unstructured P2P network using vectorized Kansei (human sensitivity) information as structured search tags for file search. Vectorized Kansei information as search tags indicates what participants feel about their files and is assigned by the participant to each of their files. A search query also has the same form of search tags and indicates what participants want to feel about files that they will eventually obtain. A method that enables file search using vectorized Kansei information is the Kansei query-forwarding method, which probabilistically propagates a search query to peers that are likely to hold more files having search tags that are similar to the query. The similarity between the search query and the search tags is measured in terms of their dot product. The simulation experiments examine if the Kansei query-forwarding method can provide equal search performance for all peers in a network in which only the Kansei information and the tendency with respect to file collection are different among all of the peers. The simulation results show that the Kansei query forwarding method and a random-walk-based query forwarding method, for comparison, work effectively in different situations and are complementary. Furthermore, the Kansei query forwarding method is shown, through simulations, to be superior to or equal to the random-walk based one in terms of search speed.

With the approval of IEEE 1901 standard for power line communications (PLC) and the recent Internet-enable home appliances like the IPTV having access to a content-on-demand service through the Internet as AcTVila in Japan, there is no doubt that PLC has taken a great step forward to emerge as the preeminent in-home-network technology. However, existing schemes developed so far have not considered the PLC network connected to an unstable Internet environment (i.e. more realistic situation). In this paper, we investigate the communication performance from the end-user's perspective in networks with large and variable round-trip time (RTT) and with the existence of cross-traffic. Then, we address the problem of unfair bandwidth allocation when multiple and different types of flows coexist and propose a TCP rate control considering the difference in terms of end-to-end delay to solve it. We validate our methodology through simulations, and show that it effectively deals with the throughput unfairness problem under critical communication environment, where multiple flows with different RTTs share the PLC and cross-traffic exists on the path of the Internet.

On wide area networks (WANs), UDP has likely been used for real-time applications, such as video and audio. UDP supplies minimized transmission delay by omitting the connection setup process, flow control, and retransmission. Meanwhile, more than 80 percent of the WAN resources are occupied by Transmission Control Protocol (TCP) traffic. As opposed to UDP's simplicity, TCP adopts a unique flow control mechanism with sliding windows. Hence, the quality of service (QoS) of real-time applications using UDP is affected by TCP traffic and its flow control mechanism whenever TCP and UDP share a bottleneck node. In this paper, the characteristics of UDP packet loss are investigated through simulations of WANs conveying UDP and TCP traffic simultaneously. In particular, the effects of TCP flow control on the packet loss of real-time audio are examined to discover how real-time audio should be transmitted with the minimum packet loss, while it is competing with TCP traffic for the bandwidth. The result obtained was that UDP packet loss occurs more often and successively when the congestion windows of TCP connections are synchronized. Especially in this case, the best performance of real-time audio applications can be obtained when they send-small sized packets without reducing their transmission rates.