Pham, StefanStefanPhamHaarbeck, NiklasNiklasHaarbeckNguyen, MinhMinhNguyenBusse, DominikDominikBusseBassbouss, LouayLouayBassboussArbanowski, StefanStefanArbanowskiSteglich, StephanStephanSteglich2025-05-202025-05-202025-04-28https://publica.fraunhofer.de/handle/publica/48776510.1007/978-981-96-4288-5_23Live video streaming on the Internet has become increasingly popular in recent years. This trend highlights the need for efficient and robust streaming protocols. The advent of HTTP/3, with its core component QUIC, introduces new features that could potentially enhance media delivery in areas such as cloud gaming and real-time rendering applications. WebTransport’s ability to facilitate access of HTTP/3 and QUIC within web browsers makes it highly attractive to cloud gaming providers and game engines, which predominantly use WebRTC to stream cloud-rendered content. Accessible directly within browsers, WebRTC has been a key technology in media streaming due to its universal availability and low-latency features. However, the emerging use of WebTransport promises scalable, lower latency solutions, potentially similar to existing low-latency techniques like WebRTC. This paper analyzes the performance of HTTP/3 and WebTransport in live low latency media streaming context, discusses the possible advantages of adopting these protocols, and compares them with WebRTC. To evaluate the performance of WebTransport and HTTP/3, a comprehensive testbed was developed, enabling a comparative analysis with WebRTC focused on end-to-end latency. Utilizing the widely-used gaming engine Unity for remote rendering, we established a pipeline architecture that employs various streaming protocols. The findings reveal that WebTransport and HTTP/3 are capable of achieving low-latency levels comparable to WebRTC. This underscores the potential of HTTP/3 and WebTransport to perform well low-latency applications and require reduced configuration overhead compared to WebRTC in client-server setups.enLive Low LatencyQUICVideo StreamingWebTransportHTTPconference paper