I think overall, WebRTC works reasonably well and is quite reliable. It's technology, though and technology will fail. I had issues in larger videoconferences with people on a super slow and unreliable connections. They'd intermittently drop out, pop back in again etc. And I had issues with people (me) using a Libre version of Firefox and some codecs weren't supported. I also had issues with people having their microphone set to weird sound devices.
Other than that, I generally had a good time with WebRTC. Especially the 1:1 direct peer calls. They're awesome and generally well supported. Peertube etc also work flawlessly here.
I guess 80% of the experience depends on how you implement it. And what code you write to handle edge cases like a poor internet connection. Or people who are bad with computers and can't figure out why their microphone doesn't work.
And if you're looking for a more bleeding edge Web API and data transport channel... I recently learned about WebTransport. It's a W3C draft for the more recent HTTP versions.