Missing for real services

From Web Real-Time Communications Working Group Wiki

Background

The question "what is missing to build 'real' services" sent on the list December 22nd 2013 [1]. The reason for the question was that at the teleconf the WG held on December 19th 2013 there were mixed views - some said that the current API and functionality is sufficient to build services on top of, whilst others said that it is only sufficient for Demos.

Justin Uberti summarized the responses to the question [2] - this page is a copy of that summary.

Note that in addition to the below, there was also responses saying "what is there now is sufficient for deploying services - in fact we already do that".

Result

Spec (in progress)

  • Error notifications need to be improved
  • More details needed on when callbacks are fired
  • ICE candidate pool missing (PC ctor)
  • Lower image resolution without stopping the stream (RTCRtpSender or MST.applyConstraints)
  • API for capping bandwidth/controlling priorities (RTCRtpSender)
  • Ability to request multiple remote streams in an offer (createOffer)
  • More debugging of candidate pair states (getStats)
  • Determine type of candidate (getStats)
  • Voice/video quality stats (getStats)
  • Remote certificate information (transport.certificates)
  • Recording of streams (MediaStreamRecorder)
  • List all the DCs on a PC (TBD if we need this or not)

Spec (v2)

  • Too attached for SDP, O/A
  • TURN auth failure does not cause an error
    • Mails Feb 6/7th vote for moving this up to v1
  • Better control of video mute behavior
  • Screen sharing without extensions (maybe)

Spec (future)

  • Access PeerConnection from Web Workers
  • Keep PeerConnection across reload/navigation

Implementations

  • Stable multi-stream support
  • NAT/FW traversal, connection stability issues (Q1)
  • AEC performance issues (Q1)
  • BWE and handling of low-bandwidth situations (video squashes audio) (Q1)
  • Not all ICE states implemented/ICE never goes to failed (Q1) (Chrome: https://code.google.com/p/webrtc/issues/detail?id=1414)
  • Processing of received MediaStreamTracks in Web Audio

Services

  • Missing server-oriented version of WebRTC
  • Multiparty, recording, broadcast
  • STUN/TURN setup still too hard

Nontechnical

  • WebRTC support in other browsers (IE, Safari)

References

[1] http://lists.w3.org/Archives/Public/public-webrtc/2013Dec/0105.html [2] http://lists.w3.org/Archives/Public/public-webrtc/2014Jan/0267.html

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