Abstract

Abstract. Session Initiation Protocol (SIP) is the de facto standard for Voice over IP implementations. Although this standard does not define conferencing directly, many drafts and papers suggest extensions and solutions to this essential service. In this paper, we first define the issues that are to be considered when building a conference service, and then we motivate our work on some of the limitations of the existing suggested solutions. This framework facilitates the conference control and media handling of a VoIP conference and aims to create a scalable and distributed conferencing system both in terms of load and control. The conference control as well as media is distributed over the network using SIP Servers and Conference Servers, which are described. The floor control is made dynamic by using a metric called Loudness Number [10] that provides a feel of a physical face-to-face conference.

Real-time services have been supported by and large on circuitswitched 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.