It's complicated :).
Basically, WebRTC is a combination of a bunch of protocols: ICE, DTLS, SCTP, and RTP. You could theoretically reduce that to ICE + QUIC for p2p use cases and just QUIC for client/server use cases.
For p2p, there is RTCQuicTransport (see https://developers.google.com/web/updates/2019/01/rtcquictra...)
For client/server, there is QuicTranport (see https://web.dev/quictransport/)
Of course, you'll probably want to be able to encode and decode some audio and video as well to make that useful. For that, there is WebCodecs (see https://github.com/WICG/web-codecs or https://www.chromestatus.com/feature/5669293909868544)
For use cases like live streaming and cloud gaming, I did a presentation about the combination of WebTransport + WebCodecs: https://www.w3.org/2018/12/games-workshop/slides/21-webtrans...
And then there is the work happening in the IETF along these same lines: https://datatracker.ietf.org/wg/ript/documents/