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

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.