As newly appointed co-chair in the W3C WebRTC WG, I just participated in my first Editor’s Call, and I’m impressed.
We had to address nearly dozens of Pull Requests and Issues on the associated github repos. We managed to knock down quite a few that ended up getting merged and a few that were closed today, despite not having 1 co-chair and 1 editor present.
There were some suggestions on how we could make the processes a bit more effective, allowing everyone to understand more what’s expected of them. It’s going to take a few meetings I suspect to get a real feel for how I can be adding the most value possible.
Overall, it feels like we are all trying our best to do what the new charter has set out, to get 1.0 done before getting on with the next chapter. I am excited to be part of it and look forward to continue helping!
If you have any thoughts on how the WebRTC Working Group could be doing things differently to be more effective and efficient, I would like to hear your thoughts.
Big thanks to everyone (especially Bernard) for putting in the extra work required here for our next CG meeting:
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 199
- 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 207
- 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
With the forthcoming re-charter @W3C WebRTC Working Group, there were also a few managerial changes:
- Peter Saint Andre (@andyet fame), will be joining as co-editor
- Erik Lagerway, yours truly (co-founder @hookflash), will be joining as co-chair
- Vivien Lacourba, W3C staff, will be helping out Dominique Hazael-Massieux with increased W3C staff time in the WebRTC Working Group
I am personally flattered and over the moon excited to have been asked to co-chair the WebRTC Working Group and look forward to working with Harald and Stefan to help usher in the next era of WebRTC standards work.
We are holding our ninth CG meeting on the 24th of June…
Where: Online (TBD)
When: June 24, 2015 10am PDT
Review action items from last meeting:
– RTCIceCandidateComplete dictionary
– RTCIceGatherer.close affect on RTCIceTransport / RTCDtlsTransport
– Comments added to #200
Incoming media prior to Remote Fingerprint Verification
– Comments added to #170, Peter to send fuller proposal to list
Response to connectivity checks prior to calling iceTransport.start()?
– Original #188 – Priority Calculation, new bug #209
Trying to remove RTCIceTransport.createAssociatedTransport(component)
– Philipp Hancke’s Review Comments
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.
Plan next meeting.
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”
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
– 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 Justin asks, what will it take to have this work with existing bittorrent clients
A They need to add WebRTC, then it will work