Technical guides on connecting Asterisk and FreeSWITCH to Siperb’s WebRTC proxy — configuration, troubleshooting, and integration patterns.
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.
This article explains how to configure Siperb as a softphone for an existing Asterisk extension or endpoint.