An Analysis of Sender-driven WebRTC Congestion Control Coexisting with QoS Assurance Applied in IEEE 802.11 Wireless LAN

Multimedia communication in the Internet shows evolution from SIP-based to WebRTC architecture which is also related to migration of QoS solutions from RSVP-based to DiffServ-based, supported also in IEEE 802.11. In order to figure out the efficiency of WEbRTC implementation in mobile network, the QoS parameters were used: throughput of media streams, error rate, delay and jitter. During experiments, 4K video streams were transmitted with the use of WebRTC technology via an IEEE 802.11ac network. The results of the experiments were compared with the ones obtained for non-responsive (not congestion controlled) UDP streams. WebRTC transmissions were live transmissions from a web camera, and UDP transmissions were emulated with the use of the iPerf tool. Results of the comparative analysis show that if the total throughput of simultaneously transmitted video streams exceeds the available throughput of the wireless link, the WebRTC is able to scale down the transmitted stream to preserve the real-time character of the stream and significantly reduce the error rate. As a result, the observed bit error rate achieved by the WebRTC was about three orders of magnitude lower than the bit error rate of non-responsive UDP streams. However, this improvement of the quality of service of the WebRTC media streams is at a cost of quality of experience (lower frame rate).