Biblio

The 2020 Corona pandemic has shown that on-line real-time multimedia communication is of vital importance when regular face-to-face meetings are not possible. One popular choice for conducting these meetings is the open standard WebRTC which is implemented in every major web browser. Even though this technology has found widespread use, there are still open issues with how different congestion control (CC) algorithms of Media- and DataChannels interact. In 2018 we have shown that the issue of self-inflicted queuing delay can be mitigated by introducing a CC coupling mechanism called FSE-NG. Originally, this solution was only capable of linking DataChannel flows controlled by TCP-style CCs and MediaChannels controlled by NADA CC. Standardization has progressed and along with NADA, IETF has also standardized the RTP CC SCReAM. This work extends the FSE-NG algorithm to also incorporate flows controlled by the latter algorithm. By means of simulation, we show that our approach is capable of drastically reducing end-to-end delays while also increasing RTP throughput and thus enabling WebRTC communication in scenarios where it has not been applicable before.

Communication between two Internet hosts using parallel connections may result in unwanted interference between the connections. In this dissertation, we propose a sender-side solution to address this problem by letting the congestion controllers of the different connections collaborate, correctly taking congestion control logic into account. Real-life experiments and simulations show that our solution works for a wide variety of congestion control mechanisms, provides great flexibility when allocating application traffic to the connections, and results in lower queuing delay and less packet loss.

WebRTC is one of the latest additions to the ever growing repository of Web browser technologies, which push the envelope of native Web application capabilities. WebRTC allows real-time peer-to-peer audio and video chat, that runs purely in the browser. Unlike existing video chat solutions, such as Skype, that operate in a closed identity ecosystem, WebRTC was designed to be highly flexible, especially in the domains of signaling and identity federation. This flexibility, however, opens avenues for identity fraud. In this paper, we explore the technical underpinnings of WebRTC's identity management architecture. Based on this analysis, we identify three novel attacks against endpoint authenticity. To answer the identified threats, we propose and discuss defensive strategies, including security improvements for the WebRTC specifications and mitigation techniques for the identity and service providers.