Abstract
WebRTC is the upcoming W3C and IETF standard for peer to peer real time communication in web browsers. It allows users to simultaneously transfer audio and video material via the Real Time Transport Protocol (RTP) as well as arbitrary data via the Stream Control Transmission Protocol (SCTP). These two transport protocols provide Quality of Service (QoS) features which are mutually exclusive in networks without Active Queue Management (AQM): RTP tries to maintain low end-to-end delays, while SCTP tries to maximize its throughput. This contradiction leads to a poor user experience for real-time flows due to excessive queuing delay caused by SCTP’s loss based Congestion Control (CC). We overcome this issue by extending the Flow State Exchange (FSE) approach by Islam et at. to also support flows with loss based CC. By means of simulation, we show that our approach keeps end-to-end delays minimal while also allowing the enforcement of different flow rate priorities.
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