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引用次数: 9
摘要
万维网(WWW)服务的发展已经纳入了新的分布式多媒体会议应用程序,推动了新一代电子服务的发展,并允许改进的交互性[1]。大多数常见的交互式实时应用程序是容错的,但受到QoS限制;低延迟要求和可靠性是会议成功的累积因素[2],它很容易吸收所有的网络资源,如果不加区分地使用可用资源,用户感知的主观质量就会很差。本文模拟了一种网络基础设施设计,在考虑非实时应用性能的情况下,对实时流量进行分类和优先级排序,以提高实时应用的性能。这个基础设施被设计成一个典型的大学校园网络,这样它就可以在任何校园中实现。本文提出了三种场景,首先在路由器上启用任何QoS机制,其次在视频流量中使用基于类的加权公平排队,使用低延迟队列(CBWFQ-LLQ),最后在语音流量中使用CBWFQ-LLQ。仿真采用OPNET IT Guru软件进行。仿真结果表明,在语音中应用LLQ可以提高整体实时和非实时应用的性能。
QoS adaptation in real time systems based on CBWFQ
The evolution of the World Wide Web (WWW) service has incorporated new distributed multimedia conference applications, powering a new generation of e-services development, and allowing improved interactivity [1]. Most common interactive real-time applications are fault-tolerant but suffer from QoS limitations; low-latency requirements and reliability are cumulative to conference success [2], it may easily absorb all network resources and the subjective quality sensed by users would remain poor if the available resources are used indiscriminately. This paper simulates a network infrastructure design that classifies and prioritizes the real-time traffic in order to improve the performance of the real-time applications, taking into consideration of the performance of non real-time applications. This infrastructure has been designed to present a typical network of a university campus, so that in this way it can be implemented in any campus. In this paper 3 scenarios are suggested first before enabling any QoS mechanism at the routers, second use class based weighted fair queuing using low latency queue (CBWFQ-LLQ) for video traffic and last scenario apply CBWFQ-LLQ for voice. The simulation is conducted using OPNET IT Guru. Simulation results demonstrated that applying LLQ for voice improves the performance of overall real-time and non real-time application.