Performance analysis of receive-side real-time congestion control for WebRTC

In the forthcoming deployments of WebRTC systems, we speculate that high quality video conferencing will see wide adoption. It is currently being deployed on Google Chrome and Firefox web-browsers, meanwhile desktop and mobile clients are under development. Without a standardized signaling mechanism, service providers can enable various types of topologies; ranging from full-mesh to centralized video conferencing and everything in between. In this paper, we evaluate the performance of various topologies using endpoints implementing WebRTC. We specifically evaluate the performance of the congestion control currently implemented and deployed in these web-browser, Receive-side Real-Time Congestion Control (RRTCC). We use transport impairments like varying throughput, loss and delay, and varying amounts of cross-traffic to measure the performance. Our results show that RRTCC is performant when by itself, but starves when competing with TCP. When competing with self-similar media streams, the late-arriving flow temporarily starves the existing media flows.