This document describes two methods of congestion control when using real-time communications on the World Wide Web (RTCWEB); one delay-based and one loss-based.
WebRTC is an open-source real-time interactive audio and video communication framework. This paper discusses some of the mechanisms utilized in WebRTC to handle packet losses in the video communication path. Various system details are discussed and an adaptive hybrid NACK/FEC method with temporal layers is presented. Results are shown to quantify how the… (More)
This document proposes an RTP header extension and an RTCP message for use in congestion control algorithms for RTP-based media flows. It adds transport-wide packet sequence numbers and corresponding feedback message so that congestion control can be performed on a transport level at the send-side, while keeping the receiver dumb.
Video conferencing applications require low latency and high bandwidth. Standard TCP is not suitable for video conferencing since its reliability and in order delivery mechanisms induce large latency. Recently the idea of using the delay gradient to infer congestion is appearing again and is gaining momentum. In this paper we present an algorithm that is… (More)
Google congestion control (GCC) has been proposed for the case of delay sensitive traffic (i.e. video-conference) in the WebRTC framework. In this paper we analyze the effect of wireless channel outages on the GCC. We have observed that, when a channel outage ends, there are packets that arrive at the receiver as a burst. This behavior impairs the… (More)