Add advanced WebRTC capabilities for your SIP server
V.2.0.1 is available
The Mizu WebRTC-SIP gateway performs full conversion between the WebRTC and SIP protocols.
The gateway is an all-in-one self-hosted software solution to convert VoIP from browsers (HTML5 WebRTC using websockets and DTLS secure media) to standard SIP protocol (plain SIP and RTP) which can be processed by common VoIP servers such as Asterisk, OpenSIPS and others. The gateway allows web browsers to interact (make and receive voice calls, video calls, chat, presence and others) with any SIP network with complete protocol conversion from WebRTC to SIP and back, including both the signaling, the ICE and the media streams, without the need to download or install any browser plugin, as WebRTC is already built in most major browsers plugin-free. The gateway includes all the necessary modules for a trouble-free WebRTC protocol conversion, such as DTLS/SRTP transcoder, built-in STUN and TURN servers and extra features to maximize call quality and success ratio, thus boosting your users' VoIP experience. Using the MRTC software you can turn any web page into a VoIP telephone or add click-to-call functionality for any website.
The WebRTC gateway (MRTC) process can be run on your existing softswitch (if that is running on legacy x86 OS) or on a separate box near your softswitch. It can be hosted also in the cloud or as a virtual machine. You will just have to set its basic networking configuration (IP, domain) and specify the SIP address of your existing server and you are ready to accept WebRTC traffic from browsers. If necessary, you can easily turn on extra features or modify its settings such as routing to multiple upper servers, DTMF method, chat/call recording and routing rules.
The Mizu WebRTC to SIP gateway can be installed and configured within minutes even by novices with less or no knowledge about SIP or WebRTC, as the gateway will self-optimize itself automatically for your network and environment, so you can start accepting WebRTC traffic to your SIP server instantly.
The MRTC gateway is intended to provide the most effective and reliable solution for WebRTC to SIP protocol conversion.
The MRTC (Mizutech WebRTC to SIP gateway) is an “all-in-one” solution for WebRTC / SIP protocol conversion with all the necessary modules built-in and with great care for the details such as various connectivity options for all network conditions, providing a reliable service for your users.