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