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

From: Justin Uberti <juberti@google.com>
Date: Tue, 28 Jan 2014 18:35:24 -0800
Message-ID: <CAOJ7v-0DM0LeFqSsot=8WF3eceidcttB_t0==rtfBA=use=BmA@mail.gmail.com>
To: "public-webrtc@w3.org" <public-webrtc@w3.org>
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> 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
>
>
> 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)
>
>
>
Received on Wednesday, 29 January 2014 02:36:13 UTC

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