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Re: Summary (updated) of "What is missing for building real services" thread

From: Stefan Håkansson LK <stefan.lk.hakansson@ericsson.com>
Date: Wed, 29 Jan 2014 08:08:49 +0000
To: Justin Uberti <juberti@google.com>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Message-ID: <1447FA0C20ED5147A1AA0EF02890A64B1CF44F66@ESESSMB209.ericsson.se>

many, many thanks for summarizing - there were many responses.

I think we should keep this in the W3C space, so I copied your document to a W3C WebRTC wiki page: https://www.w3.org/2011/04/webrtc/wiki/Missing_for_real_services

It is equally live - anyone with a W3C account can modify.


On 2014-01-29 03:37, Justin Uberti wrote:
Live document: https://docs.google.com/document/d/1EBOnUXjIlEmYO0fRAtbW-woEcPKRuwmIIxVDhyPvaic/edit

On Tue, Jan 28, 2014 at 6:25 PM, Justin Uberti <juberti@google.com<mailto:juberti@google.com>> wrote:
Went back through this 100+ message thread to capture new input from Cullen and others. I've updated the summary to include points not listed in the original summary. Overall, no surprises.

For developers - if you have input on the which of the v1 things (i.e. from spec-in-progress or implementation categories below) you want to see prioritized - I'd be interested in getting your feedback.

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

  *   Better control of video mute behavior

  *   Screen sharing without extensions (maybe)

Spec (future)

  *   Access PeerConnection from Web Workers

  *   Keep PeerConnection across reload/navigation


  *   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


  *   Missing server-oriented version of WebRTC

     *   Multiparty, recording, broadcast

  *   STUN/TURN setup still too hard


  *   WebRTC support in other browsers (IE, Safari)
Received on Wednesday, 29 January 2014 08:09:14 UTC

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