* https://github.com/elixir-webrtc (Elixir)
* https://github.com/pion/webrtc (Golang)
* https://github.com/webrtc-rs/webrtc (Rust)
* https://github.com/algesten/str0m (Rust)
* https://github.com/sepfy/libpeer (C/Embedded)
* https://github.com/awslabs/amazon-kinesis-video-streams-webr... (C/Embedded)
* https://github.com/paullouisageneau/libdatachannel (C++)
* https://webrtc.googlesource.com/src/ (C++)
* https://github.com/shinyoshiaki/werift-webrtc (Typescript)
* https://github.com/sipsorcery-org/sipsorcery (C#)
* https://github.com/aiortc/aiortc (Python)
* GStreamer’s webrtcbin (C)
Lots of documentation and demonstrations. And honestly, probably a bit better SEO or maybe searching on my part. For some reason GStreamer's webrtcbin totally flew under my radar.
Also not trying to be snarky/negative about WebRTC. I think it's fantastic at what it does. But as a tinkerer, I want something with more tinkering surface area. Not to detract from the great work on WebRTC and how accessible it's made low-latency streaming.
I've learned a lot about it over the years by following https://webrtchacks.com/, which does just that, i.e. taking apart various open and closed VoIP implementations (WebRTC or otherwise) and seeing how they work.