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| See [http://webrtc.org/ WebRTC.org] for general information and pointers.
| | Please see [[Platform/Features/VideoConferencing|Video Conferencing.]] |
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| See our [https://github.com/mozilla/rainbow/wiki/RTC_API_Proposal API Proposal]
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| Roc is working on the [http://hg.mozilla.org/users/rocallahan_mozilla.com/specs/raw-file/tip/StreamProcessing/StreamProcessing.html MediaStream Processing API] proposal.
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| == UI Issues ==
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| See [[Webcam_Sharing]] for Feature UX and security around enabling cameras and mics.
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| == Security ==
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| Security will be a tough problem due to conflicting use-case requirements. See [http://www.ietf.org/proceedings/81/slides/rtcweb-10.pdf IETF Security Considerations] for an overview of some of the issues.
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| Main topics:
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| * camera/mic enabling (and disabling)
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| * avoidance of accidental/tricked/malicious calling
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| * security of other information stored in the browser in the face of a rough rtcweb app or compromised server.
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| == Rough Todo ==
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| Items we're thinking about
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| * Codec options: base of G.711 (free subset), Opus, VP8.
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| ** Do we want to support any others? G.722, iLBC, iSAC?<br>All of those are free with no patent issues.
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| * Consider interaction with BrowserId, chat
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| * User-agent vs web content as the bridge initiator?
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| * How to redirect calls among devices?
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| ** Fork (offer connect to person at desktop, laptop, mobile at once)
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| ** Forward (device-based, provider-based) of incoming connections
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| ** Transfer (move connection to another device or person)
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| * Divide and integrate Google's webrtc code drop
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| ** Network
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| ** Capture Devices
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| ** Codecs/processing
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