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links:ietf
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links:ietf [2015/11/29 20:14] jdg [Art Area] |
links:ietf [2019/03/06 07:33] (current) |
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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 | ||
| - | * en.wikipedia.org: [[https://en.wikipedia.org/wiki/WebRTC|WebRTC]] = Application API | + | * linked to WebRTC |
| + | |||
| + | === WebRTC (W3C) === | ||
| + | |||
| + | * W3C workgroup [[http://www.w3.org/2011/04/webrtc/| WebRTC]] | ||
| + | * Wikipedia on [[https://en.wikipedia.org/wiki/WebRTC|WebRTC]] = Application API | ||
| + | * 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> | ||
| + | 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> | ||
| + | * 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://www.frafos.com/wp-content/uploads/2014/11/FRAFOS_WebRTC_Deployment.pdf | ||
| ==== Int Area ==== | ==== Int Area ==== | ||
| Line 119: | 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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