Real-Life Implementation and Evaluation of Coupled Congestion Control for WebRTC Media and Data Flows

WebRTC enables users to simultaneously transfer media (over the Real-Time Transport Protocol (RTP)) and data (over the Stream Control Transmission Protocol (SCTP)) between web browsers, multiplexed onto a single UDP port pair. This design choice of using two different transport protocols, each with their own congestion control mechanism, can lead to competition between the flows, resulting in undesirable spikes in queuing delay and packet loss. In this paper, we investigate solutions to the harmful effects WebRTC flows cause on each other by having the different congestion controllers of the flows collaborate. Using implementations in the Chromium browser, we show that our mechanism can combine a set of heterogeneous congestion control mechanisms, fairly allocate the available bandwidth between the flows, and reduce overall delay and losses.