Documents

  • WebRTC

    Accepted author manuscript, 1 MB, PDF-document

DOI

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.
Original languageEnglish
Title of host publication35th International Symposium on Computer Performance, Modeling, Measurements and Evaluation (IFIP Performance 2017)
Pages1-13
Number of pages13
DOIs
StatePublished - 2017
EventIFIP WG 7.3 Performance 2017 - Columbia University, New York, United States
Duration: 13 Jan 201717 Nov 2017
Conference number: 35
http://performance17.cs.columbia.edu/

Conference

ConferenceIFIP WG 7.3 Performance 2017
CountryUnited States
CityNew York
Period13/01/1717/11/17
Internet address

    Research areas

  • WebRTC, Congestion Control, Performance Evaluation

ID: 32870027