WebRTC SIP gateway v.1.0

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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.

Gateway to convert WebRTC to SIP. WebRTC SIP is a gateway to convert WebRTC calls from browsers to SIP and inverse turning your browser into a regular softphone.

webrtc sip, webrtc to sip, webrtc sip convert, webrtc

 
  • WebRTC SIP gateway
  • 1.0
  • 25 Feb 16
  • Mizutech S.R.L
  • WinVista, WinVista x64, Windows2003, Windows Vista, Windows Server 2012
  • Shareware
  • 25.03 Mb
  • 528
  • $1900.00
 
 
Latest Versions History
Version Date Released Release Notes
1.0 25.02.2016 WebRTC compatibility fixes, DTLS and SRTP enhancements, RTP routing speedup
 
 

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