WebRTC: Difference between revisions

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

Latest revision as of 01:18, 25 April 2012

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