public-webrtc-logs@w3.org from February 2017 by date

Tuesday, 28 February 2017

• Re: [webrtc-stats] Remove separation of received consent and connectivity requests. Varun Singh via GitHub
• [webrtc-stats] new commits pushed by vr000m Varun Singh via GitHub
• Re: [webrtc-pc] offerToReceiveAudio/offerToReceiveVideo remain in implementations (likely needed for compat) Luna Lu via GitHub
• Closed: [webrtc-stats] Discrepancies between RTCRtpCodecParameters and RTCCodecStats naming Varun Singh via GitHub
• Re: [webrtc-stats] moving roundTripTime from outbound to inbound Varun Singh via GitHub
• [webrtc-stats] new commits pushed by vr000m Varun Singh via GitHub
• Re: [webrtc-stats] Need priority information for MediaStreamTracks Harald Alvestrand via GitHub
• Re: [webrtc-stats] Consistent marker for "non-active" object? Harald Alvestrand via GitHub
• [webrtc-stats] Need advice for handling obsolete stats Harald Alvestrand via GitHub
• Re: [webrtc-stats] Related to outgoing and incoming bitrate estimates on a candidate pair Harald Alvestrand via GitHub
• [webrtc-stats] RTCTransportStats packet counters henbos via GitHub
• Re: [webrtc-stats] RTCIceCandidatePairStats packet counters henbos via GitHub
• Re: [webrtc-stats] Remove separation of received consent and connectivity requests. Taylor Brandstetter via GitHub
• [webrtc-stats] RTCIceCandidatePairStats packet counters henbos via GitHub
• Re: [webrtc-stats] aligning codec types with webrtc-pc Harald Alvestrand via GitHub
• [webrtc-stats] Unclear if the request encompasses consent checks or not Varun Singh via GitHub
• [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Re: [webrtc-stats] Remove separation of received consent and connectivity requests. Varun Singh via GitHub
• Re: [webrtc-stats] Remove separation of received consent and connectivity requests. Harald Alvestrand via GitHub
• Re: [webrtc-stats] Remove separation of received consent and connectivity requests. Harald Alvestrand via GitHub
• Re: [webrtc-stats] Make a RTCMediaStreamTrackStats object per track attachment Harald Alvestrand via GitHub
• Re: [webrtc-stats] frameWidth/frameHeight: use last decoded value Philipp Hancke via GitHub
• [webrtc-stats] Pull Request: Issue 97 Varun Singh via GitHub
• Re: [webrtc-pc] offerToReceiveAudio/offerToReceiveVideo remain in implementations (likely needed for compat) Philip Jägenstedt via GitHub
• [webrtc-stats] Pull Request: fixes issue with TURN URL protocol Varun Singh via GitHub
• Closed: [webrtc-stats] Need consistent name for RTT estimate on RTCRTPStreamStats and on ICECandidatePair Varun Singh via GitHub
• Re: [webrtc-stats] Need consistent name for RTT estimate on RTCRTPStreamStats and on ICECandidatePair Varun Singh via GitHub
• Re: [webrtc-stats] aligning codec types with webrtc-pc Varun Singh via GitHub
• [webrtc-stats] Pull Request: aligning codec types with webrtc-pc Varun Singh via GitHub
• Re: [webrtc-pc] offerToReceiveAudio/offerToReceiveVideo remain in implementations (likely needed for compat) Harald Alvestrand via GitHub
• [webrtc-stats] Pull Request: moving roundTripTime from outbound to inbound Varun Singh via GitHub
• [webrtc-stats] Pull Request: removed cancelled and renamed inprogress with in-progress. Varun Singh via GitHub
• Re: [webrtc-stats] Remove separation of received consent and connectivity requests. Varun Singh via GitHub
• Re: [webrtc-stats] Remove separation of received consent and connectivity requests. Varun Singh via GitHub
• Re: [webrtc-pc] offerToReceiveAudio/offerToReceiveVideo remain in implementations (likely needed for compat) Philip Jägenstedt via GitHub

Monday, 27 February 2017

• Re: [webrtc-pc] offerToReceiveAudio/offerToReceiveVideo remain in implementations (likely needed for compat) Harald Alvestrand via GitHub
• Re: [webrtc-pc] offerToReceiveAudio/offerToReceiveVideo remain in implementations (likely needed for compat) Luna Lu via GitHub
• [webrtc-stats] Pull Request: Remove separation of received consent and connectivity requests. Harald Alvestrand via GitHub
• Re: [webrtc-stats] Not clear how to differentiate between received connectivity checks and consent requests Harald Alvestrand via GitHub
• Re: [webrtc-pc] NetworkError event is not defined and might not be needed Bernard Aboba via GitHub

Sunday, 26 February 2017

• Re: [webrtc-pc] Indicators of usage and data flow (PING review) Dominique Hazael-Massieux via GitHub
• Closed: [webrtc-pc] Indicators of usage and data flow (PING review) Dominique Hazael-Massieux via GitHub
• Closed: [webrtc-pc] Information available prior to permission prompt (PING review) Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] Information available prior to permission prompt (PING review) Dominique Hazael-Massieux via GitHub
• Re: [webrtc-stats] Adding remoteTimestamp to RTCRtpStreamStats. Harald Alvestrand via GitHub

Saturday, 25 February 2017

• Re: [webrtc-pc] Clarify which timestamp RTCStats.timestamp represents. Harald Alvestrand via GitHub
• Re: [webrtc-pc] RTCStats timestamp source ambiguous stefan hakansson via GitHub
• Re: [webrtc-pc] RTCStats timestamp source ambiguous Taylor Brandstetter via GitHub
• [webrtc-pc] Pull Request: Clarify which timestamp RTCStats.timestamp represents. Taylor Brandstetter via GitHub
• [webrtc-stats] Pull Request: Adding remoteTimestamp to RTCRtpStreamStats. Taylor Brandstetter via GitHub

Friday, 24 February 2017

• Re: [webrtc-stats] Stats to keep track of sync between audio and video Stefan Holmer via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video henbos via GitHub
• [webrtc-pc] Pull Request: Make RTCDataChannel.id nullable and describe when it's set. Taylor Brandstetter via GitHub

Thursday, 23 February 2017

• Re: [webrtc-pc] RTCRtcpMuxPolicy of "negotiate" should be marked as an "at-risk" feature ShijunS via GitHub
• Re: [webrtc-pc] RTCRtcpMuxPolicy of "negotiate" should be marked as an "at-risk" feature Philip Jägenstedt via GitHub
• Re: [webrtc-pc] Advanced Peer-to-peer Example Sam Dutton via GitHub
• Re: [webrtc-pc] RTCRtcpMuxPolicy of "negotiate" should be marked as an "at-risk" feature ShijunS via GitHub
• Re: [webrtc-stats] Stat for how much time it takes to encode video henbos via GitHub
• Re: [webrtc-pc] Advanced Peer-to-peer Example stefan hakansson via GitHub
• Re: [webrtc-pc] Advanced Peer-to-peer Example stefan hakansson via GitHub
• Re: [webrtc-pc] RTCStats timestamp source ambiguous stefan hakansson via GitHub
• Re: [webrtc-pc] Specify when a data channel's ID is assigned, and what the `id` attribute returns when no ID is assigned. Bernard Aboba via GitHub
• Re: [webrtc-pc] Parameters for packetization interval stefan hakansson via GitHub
• Re: [webrtc-pc] Specify how media is centered, cropped, and scaled. Fixes #305 stefan hakansson via GitHub
• [webrtc-pc] Pull Request: Specify when random mid generation happens Adam Bergkvist via GitHub
• [webrtc-pc] Pull Request: Specify how transceivers get their mids in setLocal/Remote Adam Bergkvist via GitHub
• Re: [webrtc-pc] RTCRtcpMuxPolicy of "negotiate" should be marked as an "at-risk" feature Philip Jägenstedt via GitHub
• Re: [webrtc-pc] get/setParameters does not have a parameter for packetization interval Bernard Aboba via GitHub
• Re: [webrtc-stats] Stat for adaptation reason Robin Schriebman via GitHub
• Re: [webrtc-stats] Stat for how many adaptation changes occur for a video track Robin Schriebman via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video Robin Schriebman via GitHub
• Re: [webrtc-stats] Add stat to RTCTransportStats for ICE role (controlling or controlled) Taylor Brandstetter via GitHub

Wednesday, 22 February 2017

• Re: [webrtc-pc] Parameters for packetization interval Bernard Aboba via GitHub
• Re: [webrtc-stats] RTCIceCandidatePairStats.writable/readable: redundant? henbos via GitHub
• [webrtc-stats] RTCIceCandidatePairStats.writable/readable: redundant? henbos via GitHub
• Re: [webrtc-stats] Add stat to RTCTransportStats for ICE role (controlling or controlled) Varun Singh via GitHub

Tuesday, 21 February 2017

• Re: [webrtc-pc] "Hybrid" OAuth solution. misi via GitHub
• Re: [webrtc-pc] Separated auth dictionaries for STUN/TURN (issue 714) misi via GitHub
• Re: [webrtc-pc] Support for OAuth in TURN credentials (Issue 714 patch) misi via GitHub
• [webrtc-stats] Add stat to RTCTransportStats for ICE role (controlling or controlled) Taylor Brandstetter via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video henbos via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video Stefan Holmer via GitHub
• Re: [webrtc-stats] Stat for how much time it takes to encode video Stefan Holmer via GitHub
• Re: [webrtc-pc] Support for OAuth in TURN credentials (Issue 714 patch) Bernard Aboba via GitHub
• Re: [webrtc-pc] Separated auth dictionaries for STUN/TURN (issue 714) Bernard Aboba via GitHub

Monday, 20 February 2017

• Re: [webrtc-pc] Clarify that it is possible to send the same track in several copies. Bernard Aboba via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• Re: [webrtc-pc] Clarify that it is possible to send the same track in several copies. stefan hakansson via GitHub

Friday, 17 February 2017

• Re: [webrtc-pc] Add support for WebRTC Data Channel in Workers David Ammouial via GitHub
• Closed: [webrtc-pc] Receive a track multiple times Bernard Aboba via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• Closed: [webrtc-pc] Need Custom Error for IdP Bernard Aboba via GitHub
• [webrtc-pc] Pull Request: maxSimulcastStreams attribute in RTCRtpCodecCapability Bernard Aboba via GitHub
• Re: [webrtc-pc] "Hybrid" OAuth solution. Taylor Brandstetter via GitHub
• Re: [webrtc-pc] Switch to sender and receiver getStats selectors. jan-ivar via GitHub
• Re: [webrtc-stats] Stat for audio playout delay hlundin via GitHub
• Re: [webrtc-stats] Stat for how many audio stream packets are expanded when packets are lost (and lost and the user is speaking) hlundin via GitHub
• Re: [webrtc-stats] Stat for audio playout delay hlundin via GitHub
• Re: [webrtc-pc] "Hybrid" OAuth solution. misi via GitHub
• Re: [webrtc-stats] Stat for audio playout delay Varun Singh via GitHub
• [webrtc-pc] Pull Request: "Hybrid" OAuth solution. Taylor Brandstetter via GitHub

Thursday, 16 February 2017

• Re: [webrtc-pc] Mention that codecs can be reordered or removed but not modified. Taylor Brandstetter via GitHub
• Re: [webrtc-pc] Mention that codecs can be reordered or removed but not modified. Bernard Aboba via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• Closed: [webrtc-pc] Need IDP Invalid Token Bernard Aboba via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• Closed: [webrtc-pc] Need IDP Expired Token error Bernard Aboba via GitHub
• Re: [webrtc-pc] Specify how media is centered, cropped, and scaled. Fixes #305 Bernard Aboba via GitHub
• Re: [webrtc-pc] Clarify that it is possible to send the same track in several copies. stefan hakansson via GitHub
• Re: [webrtc-pc] Clarify that it is possible to send the same track in several copies. Bernard Aboba via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video henbos via GitHub
• Re: [webrtc-stats] Stat for how many adaptation changes occur for a video track henbos via GitHub
• Re: [webrtc-stats] Stat for how much time it takes to encode video henbos via GitHub
• Re: [webrtc-pc] Handing SDP with more than one identity stefan hakansson via GitHub
• Re: [webrtc-pc] Information available prior to permission prompt (PING review) stefan hakansson via GitHub
• Re: [webrtc-pc] Indicators of usage and data flow (PING review) stefan hakansson via GitHub
• Re: [webrtc-pc] Clarify reasoning behind and mitigation of privacy issues (PING review) stefan hakansson via GitHub
• Re: [webrtc-pc] add IdP token expired error stefan hakansson via GitHub
• Re: [webrtc-pc] add IdP invalid token error stefan hakansson via GitHub
• Re: [webrtc-pc] RTCRtcpMuxPolicy of "negotiate" should be marked as an "at-risk" feature Philip Jägenstedt via GitHub
• [webrtc-pc] Pull Request: maxSimulcastStreams attribute in RTCRtpCodecCapability Bernard Aboba via GitHub

Wednesday, 15 February 2017

• Re: [webrtc-pc] Don't fire events on a closed peer connection Bernard Aboba via GitHub
• [webrtc-pc] Pull Request: Don't fire events on a closed peer connection Bernard Aboba via GitHub
• Re: [webrtc-pc] strawman text to show how unverfieid media would work Bernard Aboba via GitHub
• Re: [webrtc-pc] sdpFmtpLine isn't very convenient Bernard Aboba via GitHub
• Re: [webrtc-pc] Mention that codecs can be reordered or removed but not modified. Bernard Aboba via GitHub
• Re: [webrtc-pc] add IdP token expired error Bernard Aboba via GitHub
• Re: [webrtc-pc] add string for extra info about idpErrors Bernard Aboba via GitHub
• Re: [webrtc-pc] add IdP invalid token error Bernard Aboba via GitHub
• Re: [webrtc-pc] add IdP token expired error Bernard Aboba via GitHub
• Re: [webrtc-pc] strawman text to show how unverfieid media would work Martin Thomson via GitHub
• Re: [webrtc-stats] Stat for adaptation reason Varun Singh via GitHub
• Re: [webrtc-stats] Stat for adaptation reason henbos via GitHub
• Re: [webrtc-stats] Stat for camera input framerate of local video tracks henbos via GitHub
• Re: [webrtc-pc] strawman text to show how unverfieid media would work stefan hakansson via GitHub
• Re: [webrtc-stats] Stat for how many adaptation changes occur for a video track Varun Singh via GitHub
• Re: [webrtc-stats] Unclear which framerate "framePerSecond" represents Varun Singh via GitHub
• Closed: [webrtc-stats] Stat for audio acceleration/expand rate? henbos via GitHub
• Re: [webrtc-stats] Stat for audio acceleration/expand rate? henbos via GitHub
• Re: [webrtc-stats] Need priority information for MediaStreamTracks Varun Singh via GitHub
• Re: [webrtc-stats] Stat for how many audio stream packets are expanded when packets are lost (and lost and the user is speaking) Varun Singh via GitHub
• Re: [webrtc-stats] Stat for how many audio stream packets are expanded when packets are lost (and lost and the user is speaking) Varun Singh via GitHub
• Re: [webrtc-stats] Stat for audio acceleration/expand rate? Varun Singh via GitHub
• Re: [webrtc-pc] strawman text to show how unverfieid media would work Cullen Jennings via GitHub
• Re: [webrtc-stats] Stat for audio acceleration/expand rate? Varun Singh via GitHub
• Re: [webrtc-stats] Stat for audio acceleration/expand rate? Varun Singh via GitHub
• Re: [webrtc-pc] strawman text to show how unverfieid media would work stefan hakansson via GitHub
• Re: [webrtc-pc] Align getAlgorithm return value with Web Crypto Philip Jägenstedt via GitHub

Tuesday, 14 February 2017

• Re: [webrtc-pc] strawman text to show how unverfieid media would work Roman Shpount via GitHub
• Re: [webrtc-pc] strawman text to show how unverfieid media would work stefan hakansson via GitHub
• Re: [webrtc-pc] strawman text to show how unverfieid media would work Roman Shpount via GitHub
• Re: [webrtc-pc] Meta: auto-publish changes to the spec Philip Jägenstedt via GitHub
• Re: [webrtc-stats] Stat for how many adaptation changes occur for a video track henbos via GitHub
• Re: [webrtc-stats] Stat for adaptation reason henbos via GitHub
• Re: [webrtc-stats] Stat for adaptation reason henbos via GitHub
• Re: [webrtc-stats] Stat for audio playout delay henbos via GitHub
• Re: [webrtc-stats] Stat for likelihood of echo ivocreusen via GitHub
• Re: [webrtc-stats] Stat for likelihood of echo henbos via GitHub
• Re: [webrtc-stats] Stat for likelihood of echo Philipp Hancke via GitHub
• Re: [webrtc-stats] Stat for likelihood of echo henbos via GitHub
• Re: [webrtc-pc] Meta: auto-publish changes to the spec Cullen Jennings via GitHub
• Re: [webrtc-stats] Stat for how many audio stream packets are expanded when packets are lost (and lost and the user is speaking) henbos via GitHub
• Re: [webrtc-stats] Stat for likelihood of echo Philipp Hancke via GitHub
• Re: [webrtc-stats] Stat for audio acceleration rate? henbos via GitHub
• Re: [webrtc-pc] strawman text to show how unverfieid media would work stefan hakansson via GitHub
• [webrtc-pc] Pull Request: Switch to sender and receiver getStats selectors. jan-ivar via GitHub
• Re: [webrtc-pc] offerToReceiveAudio/offerToReceiveVideo remain in implementations (likely needed for compat) Philip Jägenstedt via GitHub
• Re: [webrtc-pc] Meta: auto-publish changes to the spec Philip Jägenstedt via GitHub
• Re: [webrtc-pc] Don't fire events on a closed peer connection jan-ivar via GitHub

Monday, 13 February 2017

• Re: [webrtc-pc] Need to describe that codecs can be removed or reordered, but not modified Taylor Brandstetter via GitHub
• Re: [webrtc-pc] sdpFmtpLine isn't very convenient Bernard Aboba via GitHub
• Re: [webrtc-pc] Need to describe that codecs can be removed or reordered, but not modified Bernard Aboba via GitHub
• [webrtc-pc] Pull Request: add IdP token expired error Cullen Jennings via GitHub
• [webrtc-pc] Pull Request: add IdP invalid token error Cullen Jennings via GitHub
• [webrtc-pc] new commits pushed by fluffy Cullen Jennings via GitHub
• [webrtc-pc] Pull Request: add string for extra info about idpErrors Cullen Jennings via GitHub

Sunday, 12 February 2017

• Re: [webrtc-pc] Should IDP Login error cary the idpLoginUrl Cullen Jennings via GitHub
• Re: [webrtc-pc] Define how long should the IdP timeout timer should be Cullen Jennings via GitHub
• Re: [webrtc-pc] Need Custom Error for IdP Cullen Jennings via GitHub
• Re: [webrtc-pc] Handing SDP with more than one identity Cullen Jennings via GitHub
• Re: [webrtc-pc] Handing SDP with more than one identity Cullen Jennings via GitHub
• Re: [webrtc-pc] strawman text to show how unverfieid media would work Roman Shpount via GitHub
• Re: [webrtc-pc] strawman text to show how unverfieid media would work stefan hakansson via GitHub
• Re: [webrtc-pc] strawman text to show how unverfieid media would work Cullen Jennings via GitHub
• [webrtc-pc] Pull Request: strawman text to show how unverfieid media would work Cullen Jennings via GitHub
• Re: [webrtc-pc] Mention that codecs can be reordered or removed but not modified. Taylor Brandstetter via GitHub
• [webrtc-pc] Pull Request: Mention that codecs can be reordered or removed but not modified. Taylor Brandstetter via GitHub
• [webrtc-pc] Need to describe that codecs can be removed or reordered, but not modified Taylor Brandstetter via GitHub
• Re: [webrtc-pc] sdpFmtpLine isn't very convenient Taylor Brandstetter via GitHub
• Re: [webrtc-pc] get/setParameters does not have a parameter for packetization interval Bernard Aboba via GitHub
• Re: [webrtc-pc] sdpFmtpLine isn't very convenient Bernard Aboba via GitHub
• [webrtc-stats] new commits pushed by vr000m Varun Singh via GitHub
• Closed: [webrtc-stats] Need description for `remoteSource` member of `RTCMediaStreamTrackStats` Varun Singh via GitHub
• Closed: [webrtc-stats] Where is the API? Varun Singh via GitHub
• [webrtc-stats] new commits pushed by vr000m Varun Singh via GitHub
• Re: [webrtc-pc] getParameters does not have a parameter for packetization interval Varun Singh via GitHub

Saturday, 11 February 2017

• Re: [webrtc-pc] Specify how media is centered, cropped, and scaled. Fixes #305 stefan hakansson via GitHub
• [webrtc-pc] Pull Request: Specify how media is centered, cropped, and scaled. Fixes #305 Cullen Jennings via GitHub

Friday, 10 February 2017

• [webrtc-pc] sdpFmtpLine isn't very convenient Taylor Brandstetter via GitHub
• Re: [webrtc-pc] getParameters does not have a parameter for packetization interval Taylor Brandstetter via GitHub
• Re: [webrtc-stats] RTCPeerConnection.getStats: What to do with 'selector' argument? jan-ivar via GitHub
• Re: [webrtc-pc] getParameters does not have a parameter for packetization interval Bernard Aboba via GitHub
• Re: [webrtc-pc] getParameters does not have a parameter for packetization interval Taylor Brandstetter via GitHub
• Re: [webrtc-pc] getParameters does not have a parameter for packetization interval Varun Singh via GitHub
• Re: [webrtc-pc] getParameters does not have a parameter for packetization interval Taylor Brandstetter via GitHub
• [webrtc-pc] getParameters does not have a parameter for packetization interval Varun Singh via GitHub
• [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [webrtc-stats] RTCRTPStreamStats.mediaTrackId, RTCMediaStreamStats.trackIds Harald Alvestrand via GitHub
• Re: [webrtc-pc] Don't fire events on a closed peer connection henbos via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video Stefan Holmer via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video henbos via GitHub
• Re: [webrtc-stats] RTCPeerConnection.getStats: What to do with 'selector' argument? jan-ivar via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video Robin Schriebman via GitHub

Thursday, 9 February 2017

• Re: [webrtc-pc] Add Error Object Cullen Jennings via GitHub
• Re: [webrtc-pc] Need IDP Invalid Token Cullen Jennings via GitHub
• Re: [webrtc-pc] Advanced Peer-to-peer Example Sam Dutton via GitHub
• [webrtc-pc] Don't fire events on a closed peer connection jan-ivar via GitHub
• Re: [webrtc-pc] Advanced Peer-to-peer Example Harald Alvestrand via GitHub
• Closed: [webrtc-pc] Effect of a BYE on RtpReceiver.track Harald Alvestrand via GitHub
• Closed: [webrtc-pc] Inconsistencies in description of RTCDTMFToneChangeEvent.tone Harald Alvestrand via GitHub
• Closed: [webrtc-pc] OpenSource WebRTC IdP Harald Alvestrand via GitHub
• Re: [webrtc-pc] OpenSource WebRTC IdP Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [webrtc-pc] RTCRtcpMuxPolicy of "negotiate" should be marked as an "at-risk" feature Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [webrtc-pc] "track" event shouldn't fire for a "recvonly" description. Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Re: [webrtc-stats] Stat for retransmitted bytes Harald Alvestrand via GitHub
• Re: [webrtc-pc] Add an explicit stats selection algorithm. Harald Alvestrand via GitHub
• Re: [webrtc-pc] Receive a track multiple times stefan hakansson via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video Stefan Holmer via GitHub
• Re: [webrtc-pc] Mark negotitate in RTCRtcpMuxPolicy at risk stefan hakansson via GitHub
• Re: [webrtc-pc] Fix inconsistencies in description of RTCDTMFToneChangeEvent.tone stefan hakansson via GitHub
• Re: [webrtc-pc] Effect of a BYE on RtpReceiver.track stefan hakansson via GitHub
• [webrtc-pc] Pull Request: Clarify that it is possible to send the same track in several copies. stefan hakansson via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video Robin Schriebman via GitHub

Wednesday, 8 February 2017

• Re: [webrtc-stats] Stats to keep track of sync between audio and video Harald Alvestrand via GitHub
• Re: [webrtc-stats] Privacy & Security self review Hadley Beeman via GitHub
• Re: [webrtc-stats] Privacy & Security self review Harald Alvestrand via GitHub
• Re: [webrtc-stats] Privacy & Security self review Hadley Beeman via GitHub
• Re: [webrtc-pc] Receive a track multiple times Dominique Hazael-Massieux via GitHub
• [webrtc-pc] Pull Request: Mark negotitate in RTCRtcpMuxPolicy at risk Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] Guidance for extending objects vs extending Stats needed Dominique Hazael-Massieux via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video Stefan Holmer via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video henbos via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video Stefan Holmer via GitHub
• Re: [webrtc-stats] Stat for likelihood of echo henbos via GitHub
• Re: [webrtc-stats] Stat for likelihood of echo ivocreusen via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video henbos via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video henbos via GitHub
• Re: [webrtc-stats] RTCCodecStats.implementation... per-codec, per-stream? henbos via GitHub
• Closed: [webrtc-stats] RTCCodecStats.implementation... per-codec, per-stream? henbos via GitHub
• Closed: [webrtc-stats] Why is ssrc a DOMString? Harald Alvestrand via GitHub
• Re: [webrtc-stats] Why is ssrc a DOMString? Harald Alvestrand via GitHub
• [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Re: [webrtc-stats] Stat for audio playout delay henbos via GitHub
• Re: [webrtc-stats] Stat for how much time it takes to encode video henbos via GitHub
• Re: [webrtc-stats] Stat for how much time it takes to encode video henbos via GitHub
• Re: [webrtc-stats] Stat for camera input framerate of local video tracks henbos via GitHub
• Re: [webrtc-pc] Ensure that "track" event is only fired for "send" direction m-sections. Taylor Brandstetter via GitHub
• Re: [webrtc-stats] Definitions from MSE need re-targeting jan-ivar via GitHub
• Re: [webrtc-stats] Definitions from MSE need re-targeting jan-ivar via GitHub
• Re: [webrtc-stats] Stat for camera input framerate of local video tracks Varun Singh via GitHub
• Re: [webrtc-stats] Definitions from MSE need re-targeting Varun Singh via GitHub
• Re: [webrtc-stats] Stat for retransmitted bytes Varun Singh via GitHub
• Re: [webrtc-stats] Stat for audio playout delay Varun Singh via GitHub
• Re: [webrtc-stats] Stat for how much time it takes to encode video Varun Singh via GitHub
• Re: [webrtc-stats] Stat for how much time it takes to encode video Varun Singh via GitHub

Tuesday, 7 February 2017

• Re: [webrtc-pc] Processing remote MediaStreamTracks without MediaStreams info Harald Alvestrand via GitHub
• Closed: [webrtc-pc] Processing remote MediaStreamTracks without MediaStreams info Harald Alvestrand via GitHub
• Re: [webrtc-pc] Processing remote MediaStreamTracks without MediaStreams info Dominique Hazael-Massieux via GitHub
• Re: [webrtc-stats] Stat for audio playout delay Harald Alvestrand via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video Stefan Holmer via GitHub
• Re: [webrtc-stats] Stat for retransmitted bytes Harald Alvestrand via GitHub
• Re: [webrtc-stats] Stat for camera input framerate of local video tracks henbos via GitHub
• Closed: [webrtc-stats] Stat for target and actual encoding bitrate henbos via GitHub
• Re: [webrtc-stats] Stat for target and actual encoding bitrate henbos via GitHub
• Re: [webrtc-stats] Stat for target and actual encoding bitrate Harald Alvestrand via GitHub
• Re: [webrtc-stats] Definitions from MSE need re-targeting Harald Alvestrand via GitHub
• [webrtc-stats] Definitions from MSE need re-targeting Harald Alvestrand via GitHub
• Re: [webrtc-stats] Stat for camera input framerate of local video tracks Harald Alvestrand via GitHub
• Re: [webrtc-pc] Support assertions that identify the recipient Harald Alvestrand via GitHub
• [webrtc-pc] OpenSource WebRTC IdP Pierpaolo1992 via GitHub
• Re: [webrtc-pc] Support assertions that identify the recipient Pierpaolo1992 via GitHub
• Re: [webrtc-pc] Inconsistencies in description of RTCDTMFToneChangeEvent.tone Eric Shepherd via GitHub
• Re: [webrtc-stats] Stat for adaptation reason Harald Alvestrand via GitHub
• Re: [webrtc-pc] Information available prior to permission prompt (PING review) Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] Indicators of usage and data flow (PING review) Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] Clarify reasoning behind and mitigation of privacy issues (PING review) Dominique Hazael-Massieux via GitHub
• Re: [webrtc-stats] Stat for adaptation reason henbos via GitHub
• Re: [webrtc-pc] Clarify reasoning behind and mitigation of privacy issues (PING review) Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] Clarify reasoning behind and mitigation of privacy issues (PING review) Harald Alvestrand via GitHub
• Re: [webrtc-pc] Clarify reasoning behind and mitigation of privacy issues (PING review) Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] Inconsistencies in description of RTCDTMFToneChangeEvent.tone Harald Alvestrand via GitHub
• Re: [webrtc-pc] Inconsistencies in description of RTCDTMFToneChangeEvent.tone Harald Alvestrand via GitHub
• Re: [webrtc-stats] Stat for adaptation reason Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• [webrtc-pc] Pull Request: Ensure that "track" event is only fired for "send" direction m-sections. Taylor Brandstetter via GitHub
• Re: [webrtc-pc] Inconsistencies in description of RTCDTMFToneChangeEvent.tone Eric Shepherd via GitHub

Monday, 6 February 2017

• Re: [webrtc-pc] Receive a track multiple times Bernard Aboba via GitHub
• Re: [webrtc-pc] "track" event shouldn't fire for a "recvonly" description. Bernard Aboba via GitHub
• [webrtc-pc] Pull Request: Fix inconsistencies in description of RTCDTMFToneChangeEvent.tone Bernard Aboba via GitHub
• Re: [webrtc-pc] Inconsistencies in description of RTCDTMFToneChangeEvent.tone Bernard Aboba via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video Robin Schriebman via GitHub
• [webrtc-pc] Inconsistencies in description of RTCDTMFToneChangeEvent.tone Eric Shepherd via GitHub
• Re: [webrtc-stats] Stat for adaptation reason henbos via GitHub
• [webrtc-stats] Stat for adaptation reason henbos via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video henbos via GitHub
• Re: [webrtc-stats] Stat for how much time it takes to encode video henbos via GitHub
• Re: [webrtc-stats] RTCCodecStats needs `transportId` and `isRemote` to give it context Taylor Brandstetter via GitHub
• [webrtc-stats] Stat for camera input framerate of local video tracks henbos via GitHub
• Re: [webrtc-stats] Unclear which framerate "framePerSecond" represents henbos via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video henbos via GitHub
• Re: [webrtc-stats] RTCCodecStats needs `transportId` and `isRemote` to give it context Harald Alvestrand via GitHub
• Re: [webrtc-pc] Align getAlgorithm return value with Web Crypto Bernard Aboba via GitHub
• Closed: [webrtc-pc] Ambiguous wording in addIceCandidate Bernard Aboba via GitHub
• Re: [webrtc-pc] Ambiguous wording in addIceCandidate Bernard Aboba via GitHub
• Re: [webrtc-pc] Effect of a BYE on RtpReceiver.track Bernard Aboba via GitHub

Saturday, 4 February 2017

• [webrtc-pc] "track" event shouldn't fire for a "recvonly" description. Taylor Brandstetter via GitHub
• Re: [webrtc-stats] Stats to keep track of sync between audio and video Robin Schriebman via GitHub

Friday, 3 February 2017

• Re: [webrtc-stats] Unclear which framerate "framePerSecond" represents Taylor Brandstetter via GitHub
• Re: [webrtc-stats] What should `availableOutgoingBitrate`/`availableOutgoingBitrate` be for candidate pairs not in use? Taylor Brandstetter via GitHub
• [webrtc-stats] Stats to keep track of sync between audio and video henbos via GitHub
• Re: [webrtc-stats] Make a RTCMediaStreamTrackStats object per track attachment henbos via GitHub
• Re: [webrtc-stats] Unclear which framerate "framePerSecond" represents henbos via GitHub
• Re: [webrtc-stats] What should `availableOutgoingBitrate`/`availableOutgoingBitrate` be for candidate pairs not in use? Harald Alvestrand via GitHub

Thursday, 2 February 2017

• Re: [webrtc-stats] Not clear how to differentiate between received connectivity checks and consent requests Taylor Brandstetter via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [webrtc-pc] offerToReceiveAudio/offerToReceiveVideo remain in implementations (likely needed for compat) Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• Re: [webrtc-pc] Clarify wording on TypeError from addIceCandidate. Bernard Aboba via GitHub
• Re: [webrtc-pc] Meta: auto-publish changes to the spec Cullen Jennings via GitHub
• Closed: [webrtc-pc] Event when a transceiver is stopped via remote action Harald Alvestrand via GitHub
• Closed: [webrtc-pc] Change SetLocalDescription to require unchanged offer/answer string Harald Alvestrand via GitHub
• Closed: [webrtc-pc] currentRemoteDescription.sdp -- does it need to match the last SDP set via setRemoteDescription? Harald Alvestrand via GitHub
• Closed: [webrtc-pc] The RTCPeerConnectionIceErrorEvent constructor should have an optional init dict Harald Alvestrand via GitHub
• Re: [webrtc-pc] Meta: auto-publish changes to the spec Bernard Aboba via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Re: [webrtc-pc] Add offerToReceive* as legacy extensions Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Re: [webrtc-pc] Effect of a BYE on RtpReceiver.track Harald Alvestrand via GitHub
• Re: [webrtc-pc] Clarify wording on TypeError from addIceCandidate. Harald Alvestrand via GitHub
• Re: [webrtc-stats] Not clear how to differentiate between received connectivity checks and consent requests henbos via GitHub
• Re: [webrtc-stats] Why is ssrc a DOMString? Harald Alvestrand via GitHub
• [webrtc-stats] Pull Request: Changes type of RTCRTPStreamStats.ssrc from string to unsigned long. henbos via GitHub
• [webrtc-stats] Pull Request: frameWidth/frameHeight: use last decoded value Philipp Hancke via GitHub

Wednesday, 1 February 2017

• [webrtc-stats] Stat for target and actual encoding bitrate henbos via GitHub
• [webrtc-stats] Stat for retransmitted bytes henbos via GitHub
• Re: [webrtc-stats] Stat for likelihood of echo henbos via GitHub
• [webrtc-stats] Stat for likelihood of echo henbos via GitHub
• Re: [webrtc-stats] Stat for how many audio stream packets are expanded when the user is speaking henbos via GitHub
• Re: [webrtc-stats] Stat for how many audio stream packets are expanded when the user is speaking henbos via GitHub
• Re: [webrtc-stats] Stat for how many adaption changes occur for a video track henbos via GitHub
• [webrtc-stats] Stat for how many audio stream packets are expanded when the user is speaking henbos via GitHub
• [webrtc-stats] Stat for audio playout delay henbos via GitHub
• [webrtc-stats] Stat for how much time it takes to encode video henbos via GitHub
• Re: [webrtc-stats] Remove references saying "defines an API" Anne van Kesteren via GitHub

Last message date: Tuesday, 28 February 2017 20:55:05 UTC