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

From: cowwoc <cowwoc@bbs.darktech.org>
Date: Thu, 16 Jan 2014 23:55:59 -0500
Message-ID: <52D8B7DF.5040407@bbs.darktech.org>
To: public-webrtc@w3.org
Hi Justin,

This isn't strictly tied to the spec, but I think it makes a lot of 
sense to release a Native API at the same time as v1 that implements the 
same functionality as the Javascript API.

Browser peers will use the Javascript API while non-browser peers 
(mobile native apps, servers and gateways) will use the Native API. 
Currently everyone is rolling out their own version and there is a huge 
amount of wasted effort reinventing the wheel.

So to reiterate: the purpose of this API isn't code portability, but 
rather it is meant to be a reference/sample implementation of the 
functionality. Right now some of this functionality resides in Chrome 
(sitting above the Native API). Essentially I'm just asking you to move 
as much WebRTC-specific functionality out of Chrome and into the Native 
API such that (as much as possible) the Javascript API is a thin layer 
on top of native code.

Gili

On 15/01/2014 6:18 PM, 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)
>
Received on Friday, 17 January 2014 04:56:53 UTC

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