A Dynamic Alpha Congestion Controller for WebRTC

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.