- From: Stefan Håkansson LK <stefan.lk.hakansson@ericsson.com>
- Date: Thu, 6 Oct 2016 06:51:32 +0000
- To: "public-webrtc@w3.org" <public-webrtc@w3.org>
This is the summary of decisions made during the sessions at TPAC as recorded by the chairs. It has been assembled from the minutes [1][2] as well as from private notes. Anytime a reference to a slide is made, the slides to look at are those at [3]. We hope we recorded all decisions and got them right, but please review and comment where we did not or where we missed a decision. This is also an opportunity for those who could not participate to comment if there is any decision made that they are not happy with. Stefan for the chairs PS There were a lot of things that were discussed that are not mentioned below since we think no conclusion or decision was reached webrtc-pc ========= - webrtc-pc Issue #787 "Integrate RTCRtpTransceiver into set local/remote steps": will be resolved by referencing the relevant JSEP section - webrtc-pc Issue #645 "public negotiation-needed flag as readonly": Will be closed without any action, however a new Issue on 'clarify exactly when negotiationneeded fires' will be opened. - webrtc-pc Issue #746 "Need to specify what happens if 'createDataChannel' is called with an invalid ID" will be addressed by picking a DOMError to report. - webrtc-pc Issue #727 "removeTrack: throw exception if sender is not in connection's set of senders": decided to throw an exception - webrtc-pc Issue #526 "NetworkError event is not defined and might not be needed": RESOLUTION: we will change our close event to match with WebSocket’s - webrtc-pc Issue #799 "Unclear when a DTMFToneChangeEvent is fired with an empty string": The solution proposed in PR#807 will be implemented. - webrtc-pc PR#784 "interToneGap and duration fixes" will updated so that calling insertDTMF with duration and/or interToneGap will _not_ change the default need minor fixes and then be merged - webrtc-pc Issue #714 "STUN/TURN OAuth token auth parameter passing": The webrtc chairs will reach out to the IETF TRAM WG to get clarity. NOTE: this has been OBE after the meeting as Justin has given input in the Issue - webrtc-pc Issue #305 "Describe what happens when media changes": Of the four cases brought up only letterbox/pillar was discussed (the others handled by PR#624), the decision being that the UA should center, scale and crop (in that order) to make the video fit. - webrtc-pc error reporting (see e.g. Issues #822 - #830): We will (in the absence of good WebIDL support currently) define our own error reporting after the model used for mediacapture-main - webrtc-pc Issue #555 "Sort out requirements around IdpLoginError": Decided to go with option 1 (slide 68) based on the new errors to be defined (see above) - webrtc-pc Issue #720/PR #738: The solution in PR#738 was liked and adopted. There was some discussion on making getting the fingerprint an async operation, but since generateCertificate() is already async, the reason was not fully understood at the meeting and the proponent is asked to file a new Issue - webrtc-pc Issue #764 "Specify how an RTCRtpSender should treat an ended track": It was decided that an RTCRtpSender should treat ended, muted and disabled tracks in the same way: black (video)/silence (audio)/zero info (other types) should be transmitted. It was also decided that video tracks, when the RTCRtpSender is not transmitting anything, should render a "frozen frame" on the remote end if connected to a video element. The RTCRtpSender can be made to not transmit e.g. by setting RTCRtpSender.enabled = false for all encodings. - webrtc-pc Issue #698 "JSEP/WebRTC mismatch on empty remote MID" will be resolved by referencing the appropriate section in JSEP - webrtc-pc PR #624 "Upscale allowed": generally liked. Decided to make an enum, and other minor updates before adopting - webrtc-pc Issue #812/PR #818 "RTCIceGatheringState definition": PR #818 liked modulo some word smithing. PR #818 will be adopted once updated. - webrtc-pc issue #561 split of text between documents - agreed with split suggested on slides. webrtc-stats: ============ - webrtc-stats issue #57 QP Statistic - a PR will be generated based on the proposal. - webrtc-stats issue #61 Circuit Breaker (seeing from stats whether its trigger conditions were reached) - a PR will be generated based on the proposal - webrtc-stats issue #62 ICE pacing - we’ll ask the IETF ICE WG for advice here - webrtc-stats issue #25 / PR #43 Extension mechanisms - agreed that "living document with snapshots" is a procedure we can agree on, bar should be similar to IANA "expert review", Cullen to provide proposal text for actual procedures. - webrtc-stats issue #23 / PR #59 Conformance language - agreed that the proposed approach is OK, and that we can remove section 8.5 (IANA) from webrtc-pc. - webrtc-stats issue #20 action on non-supported stats - "omit" seems right. Stats without a value before some time has passed present a challenge (can’t detect non-support if value can be omitted for other reasons). - webrtc-stats issue #26 what happens when things stop - this has already been decided (freeze). - webrtc-stats issue #49 API version flag - we will not add this. Issue closed. - webrtc-stats issue #295 - touches on "get stats on objects", which we haven’t specified so far. The "get it all" approach has advantages too. Guidance on slide seems reasonable, and will be made into a PR; per-object getstats is for future work. [1] https://www.w3.org/2016/09/22-webrtc-minutes [2] https://www.w3.org/2016/09/23-webrtc-minutes [3] https://www.w3.org/2011/04/webrtc/wiki/images/1/1e/WebRTCWG-2016-09-22.pdf
Received on Thursday, 6 October 2016 06:52:44 UTC