WebRTC is an open W3C and IETF standard, that enables multimedia real-time communication with today's browsers. Think of internet telephony without the need for dedicated client software or plugins:
- Real-time audio/video communication browser to browser.
- Real-time audio/video communication browser to telephone network.
- Text messaging in web applications.
- Cross platform support by major vendors of browsers, smart phones, tablets, and phablets.
ISACO offers and enables applications with WebRTC that bridge between today's most important communication media, effortlessly linking the web browser with the telephone. Existing telephone applications can be easily embedded into existing web sites or web applications. The ISACO WebRTC Gateway solves inherent interoperability problems.
- Wideband audio codec support: G.722, G.711, and OPUS.
- Transcoding between different codecs.
- More codecs can be added easily.
- RTP/RTCP multiplexing/demultiplexing.
- Support for encrypted RTP transmissions (DTLS and SDES).
- Standard-compliant SIP signalling via WebSocket.
- ICE (NAT traversal).
- High availability and redundancy.