This master’s thesis explores WebTransport as a viable protocol for real-time media transmission over HTTP/3. The primary goal is to assess its practical applicability for streaming video and audio content and compare its performance to the established WebRTC protocol. Special attention is given to the integration with the WebCodecs API, which enables low-level encoding and decoding directly in the browser environment.
An experimental application was developed to support both unidirectional and bidirectional communication using different transmission strategies: WebTransport via streams, WebTransport via datagrams, and WebRTC with and without WebCodecs. In a controlled local network environment, key performance metrics were measured, including jitter, connection setup time, subjective quality of experience and the number of rendered frames per second.
The results show that WebTransport provides significantly faster connection setup compared to WebRTC. The datagram-based implementation achieved smoother playback with reduced jitter and fewer visual artifacts. While WebRTC remains a more robust option in complex network conditions, WebTransport - with proper implementation - offers greater flexibility and precise control over the transmission pipeline.
|
