W3C ORTC CG – Editors Draft Update – May 4
http://ortc.org/wp-content/uploads/2016/05/ortc.html#change-log*
B.1 Changes since 01 March 2016
- Added the
gather()
method, as noted in: Issue 165 - Removed “public” from
, as noted in: Issue 224RTCIceGatherPolicy
- Removed the minQuality attribute, as noted in: Issue 351
- Made
send()
andreceive()
asynchronous, as noted in: Issue 399, Issue 463, Issue 468 and Issue 469 - Provided additional information on ICE candidate errors, as noted in: Issue 402
- Added state attribute to
, as noted in: Issue 403RTCSctpTransport
- Provided an example of RTX/RED/FEC configuration, as noted in: Issue 404
- Clarified
payloadType
uniqueness, as noted in: Issue 405 - Updated the list of header extensions, as noted in: Issue 409
- Added “goog-remb” to the list of feedback mechanisms, as noted in: Issue 410
- Added kind argument to the
constructor, as noted in: Issue 411RTCRtpReceiver
- Clarified
send()
restrictions on kind, as noted in: Issue 414 - Added
getAlgorithm()
method, as noted in: Issue 427 - Changed
protocol and label to USVString, as noted in: Issue 429RTCDataChannel
- Clarified nullable attributes and methods returning empty lists, as noted in: Issue 433
- Clarified support for the “direction” parameter, as noted in: Issue 442
- Clarified the apt capability of the “red” codec, as noted in: Issue 444
- Clarified usage of
attributes, as noted in: Issue 445RTCRtpEncodingParameters
- Clarified firing of
onssrcconflict
event, as noted in: Issue 448 - Clarified that CNAME is only set on an
, as noted in: Issue 450RTCRtpSender
- Updated references, as noted in: Issue 457
- Described behavior of
send()
andreceive()
with unset
, as noted in: Issue 461RTCRtpEncodingParameters
- Corrected dictionary initialization in the examples, noted in: Issue 464 and Issue 465
- Corrected use of enums in the examples, noted in: Issue 466
- Clarified handling of identity constraints, as noted in: Issue 467 and Issue 468
- Clarified use of
RTCRtpEncodingParameters
, as noted in: Issue 470 - Changed hostCandidate type, as noted in: Issue 474
- Renamed state change event handlers to onstatechange, as noted in: Issue 475
- Updated description of
closed state, as noted in: Issue 476RTCIceGatherer
- Updated description of
object, as noted in: Issue 477RTCIceTransport
- Updated description of relatedPort, as noted in: Issue 484
- Updated description of
, as noted in: Issue 485RTCIceParameters
- Clarified exceptions in
construction, as noted in: Issue 492RTCDataChannel
- Provided a reference to
error.message
, as noted in: Issue 495 - Clarified
description, as noted in: Issue 496RTCRtpReceiver
- Clarified default for clockRate attribute, as noted in: Issue 500
- Removed use of “null if unset”, as noted in: Issue 503
- Updated
constructor, as noted in: Issue 504RTCSctpTransport
- Clarified behavior of
getCapabilities()
, as noted in: Issue 509 - Addressed issues with
, as noted in: Issue 519RTCDataChannelParameters
W3C ORTC CG Meeting 10 underway
ORTC, WebRTC, H.264, VP8, RID, RtpEncoding, Simulcast and much more. Google, Microsoft and Hookflash leading the discussion, join us!
http://ortc.org/2015/11/04/w3c-ortc-cg-meeting-10-november-20-2015/
Bernard Aboba joins W3C WebRTC WG as Editor
W3C WebRTC working group chairs [Harald Alvestrand (Google), Stefan Håkansson (Ericsson), Erik Lagerway (Hookflash)], made a decision recently to add a new editor to the working group, as Peter St. Andre (&yet) has resigned as editor.
Bernard Aboba (Microsoft) has now been appointed as editor.
Bernard’s attention to detail and advocacy for transparency, fairness and community has been refreshing. It has been my pleasure (as chair of the W3C ORTC CG) to work with Bernard whom also is an author in the W3C ORTC CG alongside Justin Uberti and Robin Raymond (editor). I look forward to working more with him in the WG.
Congrats Bernard!
/Erik
New WebRTC WG Charter
The new charter for the WebRTC Working Group has been approved. Current members will need to re-join, from the WebRTC WG mail list…
Hi all,
Great news, the new W3C WebRTC Working Group charter [1] has been officially approved by the W3C Director [2].
The revised charter adds a deliverable for the next version of WebRTC, has an updated list of deliverables based on the work started under the previous charter, clarifies its decision policy, and extends the group
until March 2018.The charter of this Working Group includes a new deliverable that require W3C Patent Policy licensing commitments from all Participants.
Consequently, all Participants must join or re-join the group, which involves agreeing to participate under the terms of the revised charter and the W3C Patent Policy. Current Participants may continue to attend meetings (teleconferences and face-to-face meetings) for 45 days after this announcement, even if they have not yet re-joined the group. After 45 days (ie. September 10, 2015), ongoing participation (including meeting attendance and voting) is only permitted for those who have re-joined the group.
Use this form to (re)join:
https://www.w3.org/2004/01/pp-impl/47318/joinInstructions to join the group are available at:
http://www.w3.org/2004/01/pp-impl/47318/instructionsThanks,
Vivien on behalf of the WebRTC WG Chairs and Staff contacts[1] http://www.w3.org/2015/07/webrtc-charter.html
[2] https://lists.w3.org/Archives/Member/w3c-ac-members/2015JulSep/0024.html
W3C ORTC CG – Editors Draft Update
Big thanks to everyone (especially Bernard) for putting in the extra work required here for our next CG meeting:
Draft Community Group Report 22 June 2015
B.1 Changes since 7 May 2015
- Addressed Philipp Hancke’s review comments, as noted in: Issue 198
- Added the “failed” state to
, as noted in: Issue 199RTCIceTransportState
- Added text relating to handling of incoming media packets prior to remote fingerprint verification, as noted in: Issue 200
- Added a complete attribute to the
dictionary, as noted in:Issue 207RTCIceCandidateComplete
- Updated the description of
RTCIceGatherer.close()
and the “closed” state, as noted in: Issue 208 - Updated Statistics API error handling to reflect proposed changes to the WebRTC 1.0 API, as noted in: Issue 214
- Updated Section 10 (RTCDtmfSender) to reflect changes in the WebRTC 1.0 API, as noted in: Issue 215
- Clarified state transitions due to consent failure, as noted in: Issue 216
- Added a reference to [FEC], as noted in: Issue 217
Vancouver WebRTC – Meetup 2 @PlentyofFish

With more than 40 members and growing, Vancouver WebRTC now has a new venue! Chris Simpson from PoF rallied to get us into their new presentation lounge, the “Aquarium”, thanks Chris!
Our next event is on June 25th from 6-8pm and we have a great evening planned with Omnistream and Perch presenting!
W3C ORCT CG Meeting 9 – June 24, 10am PDT
We are holding our ninth CG meeting on the 24th of June…
https://www.w3.org/community/ortc/
Where: Online (TBD)
When: June 24, 2015 10am PDT
Agenda
Review action items from last meeting:
– RTCIceCandidateComplete dictionary
https://github.com/openpeer/ortc/issues/207
– RTCIceGatherer.close affect on RTCIceTransport / RTCDtlsTransport
https://github.com/openpeer/ortc/issues/208
– Comments added to #200
Incoming media prior to Remote Fingerprint Verification
https://github.com/openpeer/ortc/issues/200
– Comments added to #170, Peter to send fuller proposal to list
Response to connectivity checks prior to calling iceTransport.start()?
https://github.com/openpeer/ortc/issues/170#issuecomment-105629219
– Original #188 – Priority Calculation, new bug #209
Trying to remove RTCIceTransport.createAssociatedTransport(component)
https://github.com/openpeer/ortc/issues/209
– Philipp Hancke’s Review Comments
https://github.com/openpeer/ortc/issues/198
Review open issues: https://github.com/openpeer/ortc/issues?q=is%3Aopen
Review current draft: http://ortc.org (upper right hand side)
Review implementation progress: ORTC Lib, MS Edge, Google ?
Review ORTC CG alignment with WebRTC WG and 1.0 spec.
Questions, comments?
Plan next meeting.
WebRTC Update from Justin Uberti, WebTorrent talk by Feross Aboukhadijeh & John Hiesey
Fresh out of Google IO, Justin Uberti provides a WebRTC update via WebRTC Meetup in SFO at the Twilio HQ. Slides and demos are not visible, I am attempting to get a copy of the slides. UPDATE: Most of the slides were captured via photos.
Justin talking points:
– Renewed focus on mobile
– HD bitrates and bandwidth estimation
– Goal H.264 coming to Chrome 45 via Cisco’s OpenH264 (whoa!)
– VP9 & hardware support
– Demo on Nexus 6 using VP9 and hardware encoder
What’s coming next..
– Mobile performance
– Complete call setup should be 500ms
– Encryption (we don’t hold the keys)
– ECDSA coming soon!
– HW encode on android capable of 1080p
– New Echo Cancellation via DAEC (Delay Agnostic Echo Canceller)
– Mobile Networks
– Network Handoff
– Scaling Quality
– Better performance on lossy networks
New domain for “WebRTC and Web Audio resources”
– Appr.tc
– g.co/webrtc
Q&A
Q What’s the story on spec deviation?
A We want to make sure we add promises to the spec.
Q Get Stats?
A Working on it
Q Unified plan support
A Organizationally challenged and taking back seat to encoding performance and other “on fire” must fix immediately
Q What is going to evolve in screen sharing in spec and Chrome?
A Things work “ok” for screen sharing but not great for some things like scrolling, people are also interested in using in tabs versus window. Screen refresh is not as fast as we would like but we think we have fixed that.
Q Changing framerate and resolution mid-call?
A RTPSender gives you some of these knobs (Note: Object from ORTC Spec!), which is on its way.
Q Battery life for hw encoded apps?
A 3 categories, voice only, video on sw, video on hw. Video demo was on hw at 1080p at 30% of CPU. HW video will compete with a baseband voice call on wifi.
Feross Aboukhadijeh & John Hiesey (creators of PeerCDN
Talking points:
– https://github.com/feross/webtorrent
– Using WebRTC DataChannel to stream content
– Demo: can’t see the screen
– Hosting websites in Browsers via WebTorrent
– NAT traversal via regular STUN / TURN
Q&A
Q Justin asks, what will it take to have this work with existing bittorrent clients
A They need to add WebRTC, then it will work