WebRTC
WebRTC
Diagox allows you to handle WebRTC clients that talk SIP and bridge with standard VOIP services without handling this on your side.
It is important that you have Websocket transport for SIP defined and endpoint that has media type of webrtc.
transports:
# Secure WebSocket for WEBRTC Users.
# TLS Configuration is done with envirment configuration. Checkout SERVER_TLS_* env variables
wss:
transport: "wss"
bind: 0.0.0.0
port: 5443
endpoints:
# Example of WEBRTC User Endpoint Configuration
# normally you want to have WS(Websocket) endpoint configured for SIP.
webrtc:
match:
type: "user"
auth:
username: "webrtc"
password: "webrtc"
media:
type: "webrtc" # USE WEBRTC media stack instead standard!