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    API: WebRTC documentation status

    This page tracks ongoing documentation work in the MDN WebRTC section.

    Feel free to contribute! We love help!

    Summary

    Pages Needs* tags Missing tags Editorial reviews Technical reviews Outdated pages Missing pages Dev-doc-needed bugs Documentation requests
    109 5 (5%) 0 (0%) 0 (0%) 12 (12%) 46 (43%) 18 (17%) 20 (19%) 33 (31%)

    See also localization status of this section.

    Needs* tags

    Found 5 pages. Learn more about how to deal with meta-tags.

    Technical reviews

    Found 12 pages. Learn more about how to do a technical review.

    Outdated pages

    Found 46 pages. These pages haven't been updated in over a year. Outdated pages can have problems with both content and format. Look at these pages and consider: Is this page talking about the Web of today? Does it look consistent with newer pages in this topic area? If not, make any needed changes.

    Missing pages

    Found 18 pages. These pages are not yet written. Please help to create them.

    Dev-doc-needed bugs

    Found 20 bugs. Learn more about how to resolve a dev-doc-needed bug.

    Bug Summary Release
    1213056 Change RTCStatsReport to be maplike. mozilla48
    1232082 Add RTCRtpReceiver and fire ontrack for remote tracks mozilla46
    1175609 onnegotiationneeded fires synchronously mozilla45
    1155923 Unprefix WebRTC mozilla44
    1189060 Add hooks to allow an extension to hook into createOffer() and createAnswer() mozilla43
    1184426 Unprefix RTCIceCandidatePairStats.priority (formerly mozPriority) mozilla42
    1152538 Enable WebRTC identity by default mozilla40
    975144 Support electrolysis for WebRTC identity mozilla38
    1091898 Update WebRTC with promises mozilla37
    1032839 Implement Track replace for RTPSender mozilla34
    1032835 Implement basic RTPSender and RTPReceiver with RTCPeerConnection mozilla34
    1036653 Implement application sharing for getUserMedia mozilla34
    1038926 Add support for window sharing for getUserMedia mozilla33
    942367 Implement peerIdentity constraint mozilla32
    1014304 Remove onconnection and onclosedconnection from PC mozilla32
    859565 Remove readyState legacy attribute in future release mozilla32
    796463 Pref on WebRTC by default mozilla21
    752352 Define and implement chrome extensions to getUserMedia and MediaStreams for privileged callers mozilla18
    691234 Write MediaEngine backend for Desktop based on WebRTC code mozilla16
    752353 Implement DOM bindings for getUserMedia mozilla15

    Browse as bug list.

    Documentation requests

    Found 33 bugs. Documentation request bugs can contain various kinds work related to MDN pages. Read through the bug and ask questions in the bug if in doubt.

    Bug Summary
    1050930 [META] Complete WebRTC documentation
    1117356 Update WebRTC basics article
    1119285 webrtc video chat examples and documentation are old and broken
    1243487 Document potential causes of WebRTC ICE renegotiation
    1257984 Document RTCIceTransport interface
    1257985 Update and finish RTCPeerConnection documentation
    1257987 Clean up RTCSessionDescription documentation
    1257988 Clean up and verify documentation of RTCPeerConnectionIceEvent
    1257990 Update and complete RTCRtpSender documentation
    1257992 Update and complete RTCRtpReceiver documentation
    1257994 Update and complete RTCTrackEvent documentation
    1257999 Document RTCCertificate
    1258002 Document RTCDataChannel
    1258004 Document RTCDTMFSender
    1258007 Document RTCDTMFToneChangeEvent
    1258393 Document RTCStatsReport
    1258394 Document RTCIdentityProviderRegistrar
    1258395 Document RTCIdentityProvider
    1258396 Document RTCIdentityAssertion
    1258400 Update documentation: RTCIdentityEvent has been removed from the spec
    1258401 Update documentation: RTCIdentityErrorEvent has been removed
    1258405 Ensure that RTCConfiguration document is complete and up-to-date
    1258408 Document RTCPeerConnectionIceErrorEvent
    1258409 Document RTCRtpTransceiver
    1258411 Document RTCDtlsTransport
    1258412 Document RTCDataChannelEvent
    1258413 Document RTCStatsReport
    1258415 Document RTCStats
    1258418 Complete and verify currentness of MediaStreamTrack documentation
    1258419 Document how to use Web-based identity providers with WebRTC
    1258423 Document how to use DTMF with WebRTC
    1258424 Document how to use WebRTC statistics to debug and measure quality of WebRTC connections
    1258437 Document privacy and security considerations for WebRTC apps

    Browse as bug list.

    Localizations

    Please help us to localize this documentation into different languages. Read more about how to translate.

    Language Pages Translated Translations up to date
    fr 85 8 (9%) 1 (12%)
    ko 95 0 (0%) 0 (NaN%)

    Team

    The documentation work will be handled primarily by:

    Sheppy
    Sample code and guide/tutorial content.
    Romain Gauthier (tOkeshu on IRC)

    has volunteered to help with samples and tutorials. He's got a few already on github, including: Signaling server. HiBuddy (bidirectional video chat)

    Goals

    • Encourage the use of a critical new Web technology.
    • Enable Firefox OS developers to easily learn how to write audio/video apps.
    • Enable Firefox OS developers to perform peer-to-peer data transactions.
    • Enable Firefox OS developers to write code that sends DTMF (touch-tone phone dialing) signals.

    User stories

    A user story is a description of a specific user's needs that should be met by the documentation once it's finished.

    • I want to write a video capture application.
    • I want to write a video/audio chat application.
    • I want to do easy peer-to-peer data transactions.
    • My app needs to send DTMF tones. Note: This is very low priority; Gecko doesn't support it yet and implementing it isn't a priority, and while Chrome supports it, generally devs aren't interested in using it.

    Documents needed

    Landing pages

    User guides

    The following user guide content will be needed.

    • High level guides
      • Introduction to WebRTC (what it is and what it can do)
      • Overview of the APIs
      • Architecture overview
        • Protocols (ICE, STUN, etc)
        • Two-way call architecture
        • Multi-party call architecture
      • The life cycle of a WebRTC connection
      • Security
      • Tools and frameworks
      • Support and compatibility details
        • Browser compatibility details
        • Supported codecs/formats
      • FAQ
    • Low level guides
      • Working with media streams
        • Introduction to the MediaStream API
        • Things you can do with media streams
        • Creating a stream
      • Peer-to-peer connections
        • Introduction
          • What it is
          • Things you can do once connected
            • Streaming media
            • Data transfer
          • What's done for you by RTCPeerConnection
        • Signaling servers
          • What signaling is
          • How it works
            • STUN
            • TURN
            • Which to use when and why
          • Setting up a signaling server
            • Available services to do it for you
            • Setting up your own
              • Available libraries and tools
              • Example: Using <some package yet to be chosen>
        • Establishing a connection
          • Session description object
          • Exchanging session descriptions
          • Opening the connection
      • Data channels
        • Introduction
          • What it is
          • Architecture
            • Use of RTCPeerConnection
            • API modeled on WebSocket
          • Advantages of RTCDataChannel
        • Using data channels
          • Creating the channel
          • Sending and receiving data
      • Using DTMF (need more info on what to do here)
      • Example:
      • Video conferencing
        • Calling another user
        • Receiving a connection

    Code samples

    The following things need to be demonstrated in sample code. This may be accomplished by either multiple small examples or one large example project which can be presented in pieces as needed to demonstrate specific concepts.

    • Video
      • One way video session
      • Audio-only sessions?
      • Starting a two-way video session
      • Multi-user sessions
      • Handling video session errors
      • Adding/removing users from a session
      • Screen sharing
    • Data
      • Initiating a data transfer session
      • Unidirectional/bidirectional?
      • Multicasting? (I don't know yet if this is supported)
    • Chat
      • A larger example of a chat client/server system using a combination of WebRTC and WebSockets
        • Text chat like IRC
        • Option to initiate video conference among users
        • Option to transmit files among users
    • Contact API
      • Using contacts to start a WebRTC session

    References

    The links below contain useful information that may help when producing WebRTC documentation.

    Notes

    Tagging standard

    • Grouping tag is: "WebRTC"

    Other tasks to do

    • TBD

    Tutorial to-do list

    • TBD

    Notes for translators

    Any notes here? Where do we expect updates, so that localizers don't translate and update too often?

    Specifications

    The interfaces defined in the following specifications are tracked by this status page:

    Specification Status Comment
    WebRTC 1.0: Real-time Communication Between Browser Working Draft  
    Media Capture and Streams Editor's Draft  
    MediaStream Recording Working Draft  

    Help the 'API/WebRTC' documentation project…
    Topic driver : Eric Shepherd (IRC nickname: sheppy)
    Look at the current status of the 'API/WebRTC' documentation.
    Don't hesitate to contact us on #mdn or on the dev-mdc mailing-list:

    Document Tags and Contributors

    Tags: 
     Contributors to this page: Sheppy, fscholz, teoli
     Last updated by: Sheppy,