Survey of switching techniques in high-speed networks and their performance

One of the most promising approaches for high speed networks for integrated service applications is fast packet switching, or ATM (Asynchronous Transfer Mode). ATM can be characterized by very high speed transmission links and simple, hard wired protocols within a network. To match the transmission speed of the network links, and to minimize the overhead due to the processing of network protocols, the switching of cells is done in hardware switching fabrics in ATM networks.A number of designs has been proposed for implementing ATM switches. While many differences exist among the proposals, the vast majority of them is based on self-routing multi-stage interconnection networks. This is because of the desirable features of multi-stage interconnection networks such as self-routing capability and suitability for VLSI implementation.Existing ATM switch architectures can be classified into two major classes: blocking switches, where blockings of cells may occur within a switch when more than one cell contends for the same internal link, and non-blocking switches, where no internal blocking occurs. A large number of techniques has also been proposed to improve the performance of blocking and nonblocking switches. In this paper, we present an extensive survey of the existing proposals for ATM switch architectures, focusing on their performance issues

Inferring Traffic Flow Characteristics from Aggregated-flow Measurement

In the Internet, a statistical perspective of global traffic flows has been considered as an important key to network operations and management. Nonetheless, it is expensive or sometime difficult to measure statistics of each flow directly. Therefore, it is of practical importance to infer unobservable statistical characteristics of individual flows from characteristics of the aggregated-flows, which are easily observed at some links (e.g., router interfaces) in the network. In this paper, we propose a new approach to such inference problems based on finding an inverse function from (observable) probabilities of some states on aggregated-flows to (unobservable) probabilities of some states on flows on a discrete state model, and provide a method inferring arrival rate statistics of individual flows (the OD traffic matrix inference). Our method is applicable to cases not covered by the existing normal-based methods for the OD traffic matrix inference. We also show simulation results on several flow topologies, which indicate potential of our approach

Efficient quality of service‐aware packet chunking scheme for machine‐to‐machine cloud services

With the recent advances in machine-to-machine(M2M) communications, huge numbers of devices have become connected and massive amounts of traffic are exchanged. M2M 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(BPS), it can exceed the router forwarding capacity, which indicates the maximum number of forwarded packets per second(PPS). This will cause the decrease in the network throughput. Therefore, eliminating the PPS limitation by chunking small packets will enable M2M 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.The 2nd IEEE International Workshop on High-Performance Interconnection Networks in the Exascale and Big-Data Era (IEEE HPCA 2016),March 12, 2016, Barcelona, Spai

Real-Time IP Flow Measurement Tool with Scalable Architecture

There is an emerging requirement for real-time flow-based traffic monitoring, which is vital to detecting and/or tracing DoS attacks as well as troubleshooting and traffic engineering in the ISP networks. We propose the architecture for a scalable real-time flow measurement tool in order to allow operators to flexibly define "the targeted flows" on-demand, to obtain various statistics on those flows, and to visualize them in a real-time manner. A traffic distribution device and multiple traffic capture devices processing packets in parallel are included in the architecture, in which the former device copies traffic and distributes it to the latter devices. We evaluate the performance of a proto-type implementation on PC-UNIX in testbed experiments to demonstrate the scalability of our architecture. The evaluation shows that the performance increases in proportion to the number of the capture devices and the maximum performance reaches 80 K pps with six capture devices. Finally we also show applications of our tool, which indicate the advantage of flexible fine-grained flow measurements

Cognitive Radio-Aware Transport Protocol for Mobile Ad Hoc Networks

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

Effective data collection scheme for real-spatial group communication over hybrid infra-ad hoc wireless networks

This paper presents an effective data collection scheme to provide group communications among appropriate members selected by each user’s geographic situation and preference (real-spatial information). When each user directly notifies central servers of user’s information via wireless network infrastructure (Wi-infra), message delivery latency and losses drastically increase due to the network congestion. Therefore, we employ representative nodes (RNs) selected in a distributed manner. The RN first collects the real-spatial information from neighboring nodes via an ad hoc network and then notifies the server via Wi-infra. From simulation experiments, our scheme can drastically reduce both message delivery latency and losses

Measurement Tool of One-Way Packet Loss Rates Based on Network Tomography

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

Inferring Link Loss Rates from Unicast-Based End-to-End Measurement

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

Adaptive Early Packet Discarding Scheme to Improve Network Delay Characteristics of Real-Time Flows

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