WebRTC names a browser protocol suite. JSEP leaves the signaling protocol to the application. SIP over WebSocket is one such protocol.
RFC 8825 is an applicability statement for real-time communication in browsers. It does not itself specify a protocol. It points at the other specifications an implementation follows to be compliant with WebRTC.
RFC 8829, the JavaScript Session Establishment Protocol, says the browser API should control the signaling state while the JavaScript chooses the signaling protocol. The document names the existing SIP call signaling protocol as one option, and a custom protocol as another. The object that has to move is the session description.
RFC 7118 specifies a WebSocket subprotocol so SIP requests and responses can cross a WebSocket connection. It is a transport document. It does not replace INVITE, REGISTER, or the status codes.
RFC 6455 is the WebSocket protocol under that subprotocol: an opening handshake, then message framing over TCP. A browser softphone that uses SIP is combining that transport with ordinary SIP methods, then a media stack that is specified elsewhere.
Room: Signaling, JSEP, and the media plane
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