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WebRTC SIP gateway 1.0

Description of WebRTC SIP gateway 1.0:
WebRTC SIP is a gateway to convert WebRTC calls from browsers to SIP and inverse turning your browser into a phone with audio, video and SMS capabilities. The gateway allows your web browser to make and receive calls from/to any SIP-legacy network or PSTN for example call to X-Lite or any mobile/landline phone. The supported protocols includes WebRTC, SIP, UDP/RTP/RTCP, WebSocket, ICE, STUN, TURN, DTLS and SRTP. Both the signaling and media conversions are supported. Compatible with any WebRTC implementation such as SIPML5, JSSIP and SIP.JS implementing RFC7118. Beside WebRTC to SIP and SIP to WebRTC, WebRTC to WebRTC and SIP to SIP calls are also supported.

Supported OS: WinVista x64, WinVista, WinServer, Windows Vista Ultimate x64, Windows Vista Ultimate, Windows Vista Starter, Windows Vista Home Premium x64, Windows Vista Home Premium, Windows Vista Home Basic x64, Windows Vista Home Basic, Windows Vista Enterprise x64, Windows Vista Enterprise, Windows Vista Business x64, Windows Vista Business, Windows Server, Windows 2003

System Req: 2 cpu core, 4 GB RAM, 60 GB disk space

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