WebRTC is a specification defining a baseline of mechanisms needed for realtime multimedia streaming. The WebRTC standard is incorporated into most modern web browsers specifically to support video conferencing, and a lot of web services have emerged to take advantage of these features in modern web browsers.
What you seek is going beyond the baseline.
This is possible to do within the WebRTC framework, by forcing some variables like resolution to use higher values that defaults (because defaults are tuned towards interoperability which is of lessser priority for your usecase).
It is also possible to not use WebRTC, but there is no real benefit in that: WebRTC does not require low quality, only defaults to (relatively) low quality.
For extreme high quality, you my want to tune variables beyond what web browsers consider sane values - e.g. if you want to transfer at resolutions and bitrates only technically viable to push over a local ethernet (where web browsers are designed for internet-wide traffic).
I can recommend to use the backend tool Janus Gateway when you want to go beyond the limits of web browsers. Janus supports the upcoming standard WHIP for connecting broadcast-oriented tools with WebRTC, that you can use to force-feed your extreme data, and learn if your network and client is able to succesfully handle the load. Janus also supports SIP, where you are more likely to find dedicated non-web-browser clients, including video-capable clients.
One quite efficient video-capable dedicated client that I am aware of is BareSIP (with precompiled binaries available for Android).
All tools mentioned here are Free Software (a.k.a. Open Source), which means they are not only free-of-charge, but also free-to-selfhost and free-to-make-a-million-dollar-business-from.