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Content available remote Queue-based simulators of P2P networks
EN
A technique is introduced for the construction of simulators of telecommunication networks based on the P2P protocol with a decentralized directory. The context of the application concerns P2P networks used both by sharing software and by telecommunication applications, as in VoIP applications. P2P systems have the advantages of boundless scalability and the break-up insensibility of devices, but suffer from heavy traffic of signals and network delays. Therefore, simulation of signal traffic in P2P networks is an important issue. The technique we have developed enables individuation of a generic P2P network's components and their functional characterization. We have worked at this level of description in order to determine the architecture of every component as a model of queues' networks. We have therefore defined a library of new types of objects for the construction of P2P network simulators with the Qnap2.V9 programming language. In this context, we have defined a procedure automatically generating the simulator of an any P2P network. During its execution, the procedure reads data from a file containing a description of the structure of the considered P2P network and values of the characteristic parameters of its components. Results related to the simulation of three configurations of a P2P network are introduced in Section 5 below. They differ according to the technology used to realize the local interconnection networks. The context of the present paper is a research program developed at the University of Camerino in defining specific computer procedures aimed at construction of simulators of telecommunication and computing networks. This technique also enables simulation of the software elaboration phases on computer networks and evaluation of the impact of specific choices concerning the structure of transmission protocols and operation paradigms of software applications [1, 2].
PL
Przedstawione w artykule stanowisko laboratoryjne umożliwia badanie jakości usług czasu rzeczywistego w sieciach IP. Podstawowym elementem stanowiska jest emulator sieci rozległej LANForge ICE firmy Candela Technologies. Jego wykorzystanie umożliwia badanie wpływu na jakość usług czasu rzeczywistego takich parametrów, jak: przepustowość sieci transmisyjnej, opóźnienie pakietów, fluktuacja opóźnienia pakietów, zmiana kolejności pakietów, a także straty i duplikacja pakietów. Opisane stanowisko może zostać wykorzystane m.in. do oceny przydatności rozwiązań proponowanych mechanizmów zapewnienia jakości usług do określonych zastosowań, oceny i weryfikacji implementacji usługi VoIP oraz oceny wpływu protokołu IPSec na jakość usług czasu rzeczywistego. W artykule zostały opisane miary jakości usług czasu rzeczywistego w sieciach IP, przedstawiono przykładowe scenariusze badań oraz zaprezentowano wyniki jednego z badań.
EN
This paper provides an overview of QoS (quality of services), discusses the need for QoS and presents the QoS toolset. QoS is defined as a measure of performance for a transmission system that reflects its transmission quality and service availability. Service availability is a crucial element of QoS. Before any QoS is implemented successfully, the network infrastructure must be designed to be highly available. The transmission quality is determined by the following factors: loss, delay, and delay variation. There is presented how to examine the impact of these factors on voice performance. This paper presents a testbed for evaluation of quality of delay-sensitive services in IP networks. Major testbed components are: LANForge ICE Network Emulator for creating IP network impairments, InterWatch 95000 Protocol Tester for generating backbone traffic, NetMeeting PC software for conveying calls, Ethereal PC software for network protocol analyzing, and an application for estimating quality of the transmission. The LANForge ICE is able to emulate networks of various latency, jitter, throughput (up to 1 Gbps) and packet degradation (loss, duplication, reordering). In consequence, it allows for testing the impact of these factors on the delay-sensitive services performance. In the paper, an examplary testing scenario for quality of voice traffic evaluation is presented. During the test, focused on addressing the QoS needs of voice traffic, you should keep the following in mind: - loss should be less than 1%, - one-way latency should be less than 150-200 ms, - average jitter should be less than 30 ms, - 21-106 kbps of guaranteed priority bandwidth is required per call (depending on the sampling rate, codec and Layer 2 overhead), - 150 bps (+ Layer 2 overhead) per phone of guaranteed bandwidth is required for Voice Control traffic. It is possible to check if the network conditions meet these requirements. The necessity and importance of assuring delay-sensitive services quality is well understood so, the need of such a testbed is obvious. This is because voice, video and mission critical data have stringent service requirements from the network infrastructure. The testbed allows for assessing practical, in simple way, delay-sensitive services quality under various network conditions. The results can be used as input data to large systems simulation.
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