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

From: Rob Manson <robman@mob-labs.com>
Date: Thu, 16 Jan 2014 10:53:02 +1100
Message-ID: <52D71F5E.5050604@mob-labs.com>
To: public-webrtc@w3.org
Hey Justin,

thanks for the summary...makes it much easier to make sure I'm not 
re-raising something I missed in the "long" threads 8)

I'd also like to add that "there's no way to add programmatically 
generated streams to peer connections". I'll assume you'll probably want 
to put that in "Spec (v2)".

At the moment the only streams that can be used are the ones generated 
by gUM. It would also be useful to be able to add post-processed streams 
(or probably more accurately MediaStreamTracks e.g. use face tracking to 
mask a persons face, or use object detection to highlight a specific 
object, etc.). At the moment the only way to send this data to the 
remote client is via another channel (e.g. DC or WS)...and sync'ing is 
definitely an issue there.

For example we would definitely like to be able to share the Augmented 
Web video streams that we can now create.
See one example here http://youtu.be/OJHgBSRJNJY

This also goes for Web Audio API generated audio tracks too.

NOTE: The generation of these streams/tracks is obviously outside the 
webrtc spec scope.

roBman


On 16/01/14 10:18 AM, Justin Uberti wrote:
> Thanks to everyone who posted about what is missing in WebRTC. I 
> attempted to collate the results below, sorted into either "spec" or 
> "implementation" categories.
>
> Basically, I think the key things that are causing trouble are being 
> actively worked on both in this WG and in implementations; we are on 
> track to resolve these problems, hopefully in the next few months.
>
> Full list below:
>
> Spec (in progress)
>
>  *
>
>     Bad error notifications
>
>  *
>
>     Lower image resolution without stopping the stream (RTCRtpSender
>     or MST.applyConstraints)
>
>  *
>
>     API for capping bandwidth (RTCRtpSender)
>
>  *
>
>     Recording of streams (MediaStreamRecorder)
>
>  *
>
>     More debugging of candidate pair states (getStats)
>
>  *
>
>     Determine type of candidate (getStats)
>
>  *
>
>     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 (working on this, some spec dependencies)
>
>  *
>
>     NAT/FW traversal, connection stability issues (Chrome working on
>     this in Q1)
>
>  *
>
>     AEC performance issues (Chrome working on this in Q1)
>
>  *
>
>     BWE and handling of low-bandwidth situations (Chrome working on
>     this in Q1)
>
>  *
>
>     Not all ICE states implemented/ICE never goes to failed (Chrome
>     working on this in Q1)
>     (Chrome: https://code.google.com/p/webrtc/issues/detail?id=1414)
>
>
> Nontechnical
>
>   * WebRTC support in other browsers (IE, Safari)
>

-- 
Rob

Checkout my new book - Getting started with WebRTC
http://www.packtpub.com/getting-started-with-webrtc/book

CEO & co-founder
http://mob-labs.com

Chair of the W3C Augmented Web Community Group
http://www.w3.org/community/ar

Invited Expert with the ISO, Khronos Group & W3C
Received on Wednesday, 15 January 2014 23:53:29 UTC

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