tl;dr using WebRTC just for realtime client<->server data sucks, but WebTransport[1] is coming soon to serve that exact usecase with an easy API
WebRTC has data channels, which are currently the only way to achieve unreliable and unordered real-time communication (UDP-style) between the browser and other browsers or a server. This is pretty essential for any networked application where latency is critical, like voice and video and fast-paced multiplayer games.
As other commenters have noted, it's a royal pain in the ass to set up WebRTC if all you want is UDP-style communication between a server and browser, since you need to wrangle half a dozen other protocols in the process.
However! A new API, WebTransport[1], is actively being developed that will offer a WebSockets-like (read: super simple to set up) API for UDP-style communication. I am extremely excited about it and its potential for real-time browser-based multiplayer games (which I'm working on).
What do you find hard about setting up a WebRTC server? I hear from users they are able to spin up a Pion DataChannel server in 5 mins (including installing Go)
WebTransport isn’t going to be here soon though, I would be cautious in investing. Stuff like Congestion Control is still a big unknown [0] and we don’t have Datagrams everywhere.
I just starred it, it does seem like it'd alleviate a some of the pain. Another similar (more end to end) project I had come across is: geckos.io. This previous HN thread[1] discusses some of the (at least perceived) difficulty with using WebRTC as a "WebSockets but UDP" solution.
I'm not sure I follow the concern about congestion control. UDP doesn't have it either, presumably since if you're building an application that requires such latency sensitivity, you don't mind rolling your own congestion control algorithm that makes most sense given your application's specific needs, right? Pretty much all FPS games do this, as far as I understand.
WebRTC has data channels, which are currently the only way to achieve unreliable and unordered real-time communication (UDP-style) between the browser and other browsers or a server. This is pretty essential for any networked application where latency is critical, like voice and video and fast-paced multiplayer games.
As other commenters have noted, it's a royal pain in the ass to set up WebRTC if all you want is UDP-style communication between a server and browser, since you need to wrangle half a dozen other protocols in the process.
However! A new API, WebTransport[1], is actively being developed that will offer a WebSockets-like (read: super simple to set up) API for UDP-style communication. I am extremely excited about it and its potential for real-time browser-based multiplayer games (which I'm working on).
https://github.com/w3c/webtransport