tiny-webrtc-gw
Welcome to the tiny-webrtc-gw readme!
tiny-webrtc-gw is a self-contained webRTC video/audio conferencing (many-to-many) server daemon for linux.
The simplest way to roll-your-own (secure) webRTC video broadcast service.
(screenshot here)
Head over to the demo!
Hot features:
- Very low latency 1-many streaming
- HD stream support
- text chat room
- highly scalable (native c/c++ code)
- end-to-end encrypted
- chrome/firefox/opera/safari (iOS) support
- easy compilation (just git checkout --recursive and "make all")
Demo at https://weephone.domain17.net/
Building:
building requires 'go' to compile boringssl (so install those packages)
Make sure you checked out the websocket git submodule by checking out with --recursive or doing git submodule init ws && git submodule update ws
from the base directory just run 'make all'.
You will need to edit at least one line in config.txt so the built-in STUN server knows its own IP address (relative to the clients connecting, if you're using NAT). Go to whatismyipaddress.com and replace the udpserver_addr=x.x.x.x line with your own IP address.