Abstract
Video conferencing applications have significantly changed the way people communicate over the Internet. Web Real-Time Communication (WebRTC), drafted by the World Wide Web Consortium (W3C) and Internet Engineering Task Force (IETF) working groups, has added new functionality to the web browsers, allowing audio/video calls between browsers without the need to install any video telephony applications. The Google Congestion Control (GCC) algorithm has been proposed as WebRTC's congestion control mechanism, but its performance is limited due to using a fixed incoming rate decrease factor, known as alpha (a). In this paper, we propose a dynamic alpha model to reduce the available receiving bandwidth estimate during overuse as indicated by the over-use detector. Experiments using our specific testbed show that our proposed model achieves a 33% higher incoming rate and a 16% lower round-trip time, while keeping a similar packet loss rate and video quality, compared to a fixed alpha model.