- From: jianjunz via GitHub <sysbot+gh@w3.org>
- Date: Tue, 21 Jul 2020 03:49:47 +0000
- To: public-webrtc@w3.org
jianjunz has just created a new issue for https://github.com/w3c/webrtc-nv-use-cases: == Low latency streaming with WebTransport + WebRTC == The use case is low latency realtime live streaming from server to browser. The server could be a SFU or a cloud gaming server. Nowadays, we can achieve it by using WebRTC 1.0 or WebTransport + MSE, but both of them are not perfect. WebRTC involves ICE which is not required by server-client architecture. WebTransport + MSE, more specifically, reliable QuicStream + MSE, may have large latency than WebRTC when network is not good. Here are two ideas to reduce the latency - A `PeerConnection` accepts a plugable transport module, so it could be an `RTCDtlsTransport` defined in *webrtc-pc* or a `DatagramTransport` defined in *WebTransport*. In this case, a new transport interface is needed. - Expose RTP module to JavaScript developers to fulfill the gap between *WebTransport* and *WebCodecs*. Then, applications could use `DatagramTransport` + RTP module + WebCodecs to receive and decode media streams. Please view or discuss this issue at https://github.com/w3c/webrtc-nv-use-cases/issues/62 using your GitHub account
Received on Tuesday, 21 July 2020 03:49:49 UTC