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Deployment issues of a VoIP conferencing system in a Virtual Conferencing Environment

机译:虚拟会议环境中VoIP会议系统的部署问题

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Real-time services have been supported by and large on circuit-switched networks. Recent trends favour services ported on packet-switched networks. For audio conferencing, we need to consider many issues -- scalability, quality of the conference application, floor control and load on the clients/servers -- to name a few. In this paper, we describe an audio service framework designed to provide a Virtual Conferencing Environment (VCE). The system is designed to accommodate a large number of end users speaking at the same time and spread across the Internet. The framework is based on Conference Servers [14], which facilitate the audio handling, while we exploit the SIP capabilities for signaling purposes. Client selection is based on a recent quantifier called "Loudness Number" that helps mimic a physical face-to-face conference. We deal with deployment issues of the proposed solution both in terms of scalability and interactivity, while explaining the techniques we use to reduce the traffic.We have implemented a Conference Server (CS) application on a campus-wide network at our Institute.
机译:实时服务已得到电路交换网络的广泛支持。最近的趋势倾向于在分组交换网络上移植的服务。对于音频会议,我们需要考虑许多问题,例如可伸缩性,会议应用程序的质量,发言权控制和客户端/服务器上的负载等。在本文中,我们描述了一种音频服务框架,旨在提供虚拟会议环境(VCE)。该系统旨在容纳大量同时讲话并在Internet上传播的最终用户。该框架基于会议服务器[14],它可以简化音频处理,同时我们将SIP功能用于信令目的。客户选择基于最近称为“响度数”的量词,该量词有助于模拟实际的面对面会议。我们在解决可扩展性和交互性方面都解决了所提出解决方案的部署问题,同时解释了减少流量的技术。我们已经在研究所的一个校园范围的网络中实现了会议服务器(CS)应用程序。

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