Performance Evaluation of WebRTC-based video conferencing

Performance Evaluation of WebRTC-based video conferencing

Jansen, Bart (Student TU Delft)
Goodwin, Timothy (Columbia University)
Gupta, Varun (Columbia University)
Kuipers, F.A. (TU Delft Embedded Systems)
Zussman, Gil (Columbia University)

2017

WebRTC has quickly become popular as a video conferencing platform, partly due to the fact that many browsers support it. WebRTC utilizes the Google Congestion Control (GCC) algorithm to provide congestion control for realtime communications over UDP. The performance during a WebRTC call may be influenced by several factors, including the underlying WebRTC implementation, the device and network characteristics, and the network topology. In this paper, we perform a thorough performance evaluation of WebRTC both in emulated synthetic network conditions as well as in real wired and wireless networks. Our evaluation shows that WebRTC streams have a slightly higher priority than TCP flows when competing with cross traffic. In general, while in several of the considered scenarios WebRTC performed as expected, we observed important cases where there is room for improvement. These include the wireless domain and the newly added support for the video codecs VP9 and H.264 that does not perform as expected.

WebRTC
Congestion Control
Performance Evaluation

http://resolver.tudelft.nl/uuid:e4eef445-e4bc-4105-ba8c-72a6b8bea036

https://doi.org/10.1145/3199524.3199534

0163-5999

ACM SIGMETRICS Performance Evaluation Review, 45 (3), 56-68

IFIP WG 7.3 Performance 2017, 2017-01-13 → 2017-11-17, Columbia University, New York, United States

Institutional Repository

journal article

© 2017 Bart Jansen, Timothy Goodwin, Varun Gupta, F.A. Kuipers, Gil Zussman