W3C home > Mailing lists > Public > public-webrtc@w3.org > January 2014

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

From: Justin Uberti <juberti@google.com>
Date: Tue, 28 Jan 2014 18:25:02 -0800
Message-ID: <CAOJ7v-27+Arf==Aq_JKdNf85QV8VvPDz6+tmM2sZP-v+2yc0+w@mail.gmail.com>
To: "public-webrtc@w3.org" <public-webrtc@w3.org>
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 02:25:49 UTC

This archive was generated by hypermail 2.3.1 : Monday, 23 October 2017 15:19:38 UTC