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| * art = Applications and Real-Time Area | * art = Applications and Real-Time Area | ||
| * [[https://tools.ietf.org/wg/rtcweb/|rtcweb]] Real-Time Communication in WEB-browsers | * [[https://tools.ietf.org/wg/rtcweb/|rtcweb]] Real-Time Communication in WEB-browsers | ||
| - | * linked to WebRTC: | + | * linked to WebRTC |
| + | |||
| + | === WebRTC (W3C) === | ||
| * W3C workgroup [[http://www.w3.org/2011/04/webrtc/| WebRTC]] | * W3C workgroup [[http://www.w3.org/2011/04/webrtc/| WebRTC]] | ||
| * Wikipedia on [[https://en.wikipedia.org/wiki/WebRTC|WebRTC]] = Application API | * Wikipedia on [[https://en.wikipedia.org/wiki/WebRTC|WebRTC]] = Application API | ||
| * http://www.webrtc.org/ | * http://www.webrtc.org/ | ||
| + | * WebRTC is an API defined in [[wp>JavaScript]] and [[wp>JSON]] \\ and has 3 parts: | ||
| + | - MediaStream and getUserMedia (access to camera and microphone) | ||
| + | - RTCPeerConnection (audio/video streams) | ||
| + | - RTCDataChannel (for gaming, RDP, etc.) | ||
| * http://www.html5rocks.com/en/tutorials/webrtc/basics/ \\ <code> | * http://www.html5rocks.com/en/tutorials/webrtc/basics/ \\ <code> | ||
| - | Google bought GIPS, a company which had developed many components required for RTC, such as codecs and echo cancellation techniques. Google open sourced the technologies developed by GIPS and engaged with relevant standards bodies at the IETF and W3C to ensure industry consensus. In May 2011, Ericsson built the first implementation of WebRTC. | + | Google bought GIPS, a company which had developed many components required |
| + | for RTC, such as codecs and echo cancellation techniques. Google open | ||
| + | sourced the technologies developed by GIPS and engaged with relevant | ||
| + | standards bodies at the IETF and W3C to ensure industry consensus. | ||
| + | In May 2011, Ericsson built the first implementation of WebRTC. | ||
| </code> | </code> | ||
| + | * above says: "Suffice to say that the STUN protocol and its extension TURN are used by the ICE framework to enable RTCPeerConnection to cope with NAT traversal and other network vagaries." | ||
| + | * [[https://en.wikipedia.org/wiki/STUN|STUN]] | ||
| + | * [[https://en.wikipedia.org/wiki/Traversal_Using_Relays_around_NAT|TURN]] | ||
| + | * [[https://en.wikipedia.org/wiki/Interactive_Connectivity_Establishment|ICE]] | ||
| + | * [[https://en.wikipedia.org/wiki/Network_address_translation#Full-cone_NAT|Full-cone NAT]] | ||
| + | * [[https://www.youtube.com/watch?v=p2HzZkd2A40&t=21m12s|Real-time communication with WebRTC: Google I/O 2013]] | ||
| + | * http://io13webrtc.appspot.com/#1 | ||
| * http://knowledge.santanu.net/what-is-webrtc-current-scenario-and-why-we-should-follow/ | * http://knowledge.santanu.net/what-is-webrtc-current-scenario-and-why-we-should-follow/ | ||
| * http://www.frafos.com/wp-content/uploads/2014/11/FRAFOS_WebRTC_Deployment.pdf | * http://www.frafos.com/wp-content/uploads/2014/11/FRAFOS_WebRTC_Deployment.pdf | ||
| Line 127: | Line 145: | ||
| === NSIS === | === NSIS === | ||
| + | * Next Steps in Signaling (NSIS) is an open signaling model and in some sense an extension to RSVP | ||
| * https://tools.ietf.org/wg/nsis/ | * https://tools.ietf.org/wg/nsis/ | ||
| * [[https://tools.ietf.org/html/rfc3726|RFC3726]] Requirements for Signaling Protocols | * [[https://tools.ietf.org/html/rfc3726|RFC3726]] Requirements for Signaling Protocols | ||
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