Re: Agenda, June 23 2016

Same here.  I keep getting errors both from Calendar (from which I normally
click the link) and from the hangouts link itself (where it just tells me
it can't connect).



On Thu, Jun 30, 2016 at 10:09 AM, Bernard Aboba <Bernard.Aboba@microsoft.com
> wrote:

> I am having trouble getting into the Hangout and the phone number/passcode
> doesn't seem to work either.
>
> Any ideas?
> ________________________________________
> From: Harald Alvestrand [harald@alvestrand.no]
> Sent: Thursday, June 23, 2016 5:00 AM
> To: public-webrtc-editors@w3.org
> Subject: Agenda, June 23 2016
>
> Overarching items:
>
> - Who's present over July / August?
> - Who's going to Berlin?
> - What versions need pushing before / just after Tuesday's interim?
>
> Mediacapture-main
> ============
> Pulls
> -----
>
> #367 Convert to WebIDL contiguous mode in ReSpec ()
> Perfect time to merge?
>
> Issues
> --------
>
> #350 New permission definitions are wrong. (alvestrand)
> #359 MUST clear requirement for deviceId (aboba)
> #360 Specify relation between return from getConstraints and con (burnburn)
> #366 WebIDL requires constraints on getUserMedia to be optional
> (alvestrand)
> Editorial.
>
> #368 Remove navigator.getUserMedia from the spec. ()
> I'm too close to this to comment. Assignee?
>
> #369 Remove 'advanced' constraints from the spec. ()
> Reopening previous discussion. Assign to chair ...?
>
> #370 Restore the active/inactive events on MediaStream ()
> This is a "do we care about what's implemented" topic. Assignee?
>
>
> WebRTC-PC
> =======
>
> Pulls
> ------
> #601 Specify the synchronous and queued steps for addIceCandidat (adam-be)
> As Jan-Ivar asks: Can we land this?
>
> #624 Upscale allowed (fluffy)
> No fluffy action.
>
> #647 Clarification on RTX in Codec Capabilities/Parameters (aboba)
> Next interim.
>
> #675 RTCRtpEncodingParameters attribute to turn on/off sending C (aboba)
> Next interim.
>
> #683 Add RTCRtpCodecRtxParameters dictionary (related to #548) (aboba)
> Next interim.
>
> #695 Meaning of "Liveness checks have failed" (adam-be)
> Discussion needed? I'd like to understand concerns.
>
> #699 Revised WebRTC 1.0 RTCIceTransportState diagram (stefhak)
> Ready? Address Taylor's comments?
>
> #704 Use 'create an RTCRtpSender/Receiver/Transceiver' definitio ()
> Dom's comments need addressing
>
> #707 Mark main definition of RTCRtpSender as such [editorial] ()
> #708 Fix link to defaultIceServers [editorial] ()
> #711 Fix link to rtcpc constructor ()
> Editorial block. All merge?
>
> #712 Reference MMUSIC RID document ()
> Travis doesn't like it. MMUSIC-RID is undefined.
>
> #713 Missing destruction sequence ()
> WIP
>
> Issues
> --------
> I'd like to have someone pick up #709, since it originates from my team.
>
> #253 Assurance that requests to IdP proxy originate from the use
> (martinthomson)
> #257 ICE Candidate should have accessors for protocol-relevant i
> (alvestrand)
> #295 Guidance for extending objects vs extending Stats needed (alvestrand)
> #296 Debugging ICE problems needs more info (aboba)
> #305 Describe what happens when media changes (fluffy)
> #337 Interfacing between WebRTC spec and JSEP (burnburn)
> #457 Non-normative ICE state transition diagram (taylor-b)
> #526 NetworkError event is not defined and might not be needed (adam-be)
> #548 RTX/RED/FEC handling (aboba)
> #551 Errors when identifying a m-line in addIceCandidate() (adam-be)
> #554 We never fire the 'connectionstatechange' event (adam-be)
> #555 Sort out requirements around IdpLoginError (martinthomson)
> #561 Normatively cite webrtc-stats for sections 8.x (alvestrand)
> #562 What to do with an RTCIdentityProvider that returns rubbish
> (martinthomson)
> #566 Separate sender and receiver sets are unnecessary when we h (burnburn)
> #571 Mechanisms for populating the contents of RTCRtpSender/Rece (taylor-b)
> #578 Need to specify precisely when MID generation happens (adam-be)
> #579 Congruenting about "The negotiation-needed flag is cleared  (adam-be)
> #600 Operations queue: What is run synchronously before the oper (adam-be)
> #644 Fob on RTCRtpEncodingParameters to turn on and off sending  (aboba)
> #645 public negotiation-needed flag as readonly (adam-be)
> #654 Need JSEP reference for general RTCPeerConnection descripti (burnburn)
> #655 Update JSEP reference to 5.8 (burnburn)
> #658 Link addIceCandidate to JSEP for applying ICE candidate (burnburn)
> #661 Add informative table of all things that can cause negotiat (burnburn)
> #669 Missing destruction sequence for ice agent. (aboba)
> #671 Processing remote MediaStreamTracks without MediaStreams in
> (alvestrand)
> #678 Support assertions that identify the recipient (martinthomson)
> #684 Proper JSEP reference for sendEncodings in subsequent offer (burnburn)
> #685 Update JSEP reference for receipt of multiple RTP encodings (aboba)
> #687 Clarify reasoning behind and mitigation of privacy issues ( (stefhak)
> #688 Indicators of usage and data flow (PING review) (stefhak)
> #690 Information available prior to permission prompt (PING revi (stefhak)
> #692 Meaning of "Liveness checks have failed" for `disconnected` (adam-be)
> #698 JSEP/WebRTC mismatch on empty remote MID (adam-be)
> #700 An event for when a Circuit Breaker is triggered (stefhak)
> #703 Editorial: Almost all links to RTCRtpSender points to a par
> (dontcallmedom)
> #705 Missing sender identifier attribute (msid) (stefhak)
> #706 How does setDirection interact with active/inactive sender/ (aboba)
> #709 offerToReceiveAudio/offerToReceiveVideo remain in implement ()
> #710 Links to 'RTCPeerConnection constructor algorithm' are miss ()
>
>
>
>