WebRTC topic
With WebRTC, you can add real-time communication capabilities to your application that works on top of an open standard. It supports video, voice, and generic data to be sent between peers, allowing developers to build powerful voice- and video-communication solutions. The technology is available on all modern browsers as well as on native clients for all major platforms. The technologies behind WebRTC are implemented as an open web standard and available as regular JavaScript APIs in all major browsers. For native clients, like Android and iOS applications, a library is available that provides the same functionality. The WebRTC project is open source and supported by Apple, Google, Microsoft and Mozilla, amongst others. This page is maintained by the Google WebRTC team.
go-stun
A go implementation of the STUN client (RFC 3489 and RFC 5389)
coplay
Synchronizing video play between two peers.
XSound
XSound gives Web Developers Powerful Audio Features Easily !
blaze
⚡ File sharing progressive web app built using WebTorrent and WebSockets
wooglies
An experimental project for online collaboration (WebXR, Three.js, WebRTC, multiplayer, positional audio)
janus-gateway-android
Implements Janus gateway video room on Android
janus-gateway-ios
Implements Janus gateway video room on iOS
bevy_networking_turbulence
Networking plugin for Bevy engine running on naia-socket and turbulence libraries