LiveSwitch, our latest WebRTC innovation, was released today!
LiveSwitch provides flexible server-based selective forwarding, mixing, broadcasting and recording for multiparty audio/video conferencing - all installable on your own infrastructure or in your own cloud.
LiveSwitch's hybrid approach gives you the ability to establish P2P-, SFU-, or MCU-based media flows, as well as switch between flows as desired.
LiveSwitch also features full SIP integration to support combining WebRTC conferencing with legacy VoIP and PSTN telephony networks, either directly or via virtual PBX platforms.
To find out more about installing the server, and spinning up a demo in your development environment, checkout the Getting Started Guide and, as always, head on over to the Downloads page to pick up the latest!