Asterisk and FreeSWITCH

This guide details how to set up Asterisk for WebRTC, enabling browser-based voice and video calls. It covers essential Asterisk configurations for WebSocket, DTLS, and SIP, along with SIP.js setup to create a WebRTC client for making and receiving calls.

This guide provides a detailed setup for enabling WebRTC with FreeSWITCH, allowing for browser-based voice and video calls. It covers FreeSWITCH configuration for WebSocket and SRTP support, along with SIP.js setup for making and receiving WebRTC calls.