[webrtc-pc] "Hybrid" OAuth solution.
[webrtc-pc] "track" event shouldn't fire for a "recvonly" description.
[webrtc-pc] Add an explicit stats selection algorithm.
[webrtc-pc] Add Error Object
[webrtc-pc] add IdP invalid token error
[webrtc-pc] add IdP token expired error
[webrtc-pc] Add offerToReceive* as legacy extensions
[webrtc-pc] add string for extra info about idpErrors
[webrtc-pc] Add support for WebRTC Data Channel in Workers
[webrtc-pc] Advanced Peer-to-peer Example
[webrtc-pc] Align getAlgorithm return value with Web Crypto
[webrtc-pc] Ambiguous wording in addIceCandidate
[webrtc-pc] Clarify reasoning behind and mitigation of privacy issues (PING review)
[webrtc-pc] Clarify that it is possible to send the same track in several copies.
[webrtc-pc] Clarify which timestamp RTCStats.timestamp represents.
[webrtc-pc] Clarify wording on TypeError from addIceCandidate.
[webrtc-pc] Define how long should the IdP timeout timer should be
[webrtc-pc] Don't fire events on a closed peer connection
[webrtc-pc] Effect of a BYE on RtpReceiver.track
[webrtc-pc] Ensure that "track" event is only fired for "send" direction m-sections.
[webrtc-pc] Fix inconsistencies in description of RTCDTMFToneChangeEvent.tone
[webrtc-pc] get/setParameters does not have a parameter for packetization interval
[webrtc-pc] getParameters does not have a parameter for packetization interval
[webrtc-pc] Guidance for extending objects vs extending Stats needed
[webrtc-pc] Handing SDP with more than one identity
[webrtc-pc] Inconsistencies in description of RTCDTMFToneChangeEvent.tone
[webrtc-pc] Indicators of usage and data flow (PING review)
[webrtc-pc] Information available prior to permission prompt (PING review)
[webrtc-pc] Mark negotitate in RTCRtcpMuxPolicy at risk
[webrtc-pc] Mention that codecs can be reordered or removed but not modified.
[webrtc-pc] Meta: auto-publish changes to the spec
[webrtc-pc] Need Custom Error for IdP
[webrtc-pc] Need IDP Invalid Token
[webrtc-pc] Need to describe that codecs can be removed or reordered, but not modified
[webrtc-pc] NetworkError event is not defined and might not be needed
[webrtc-pc] new commits pushed by aboba
[webrtc-pc] new commits pushed by alvestrand
[webrtc-pc] new commits pushed by fluffy
[webrtc-pc] offerToReceiveAudio/offerToReceiveVideo remain in implementations (likely needed for compat)
[webrtc-pc] OpenSource WebRTC IdP
[webrtc-pc] Parameters for packetization interval
[webrtc-pc] Processing remote MediaStreamTracks without MediaStreams info
[webrtc-pc] Pull Request: "Hybrid" OAuth solution.
[webrtc-pc] Pull Request: add IdP invalid token error
[webrtc-pc] Pull Request: add IdP token expired error
[webrtc-pc] Pull Request: add string for extra info about idpErrors
[webrtc-pc] Pull Request: Clarify that it is possible to send the same track in several copies.
[webrtc-pc] Pull Request: Clarify which timestamp RTCStats.timestamp represents.
[webrtc-pc] Pull Request: Don't fire events on a closed peer connection
[webrtc-pc] Pull Request: Ensure that "track" event is only fired for "send" direction m-sections.
[webrtc-pc] Pull Request: Fix inconsistencies in description of RTCDTMFToneChangeEvent.tone
[webrtc-pc] Pull Request: Make RTCDataChannel.id nullable and describe when it's set.
[webrtc-pc] Pull Request: Mark negotitate in RTCRtcpMuxPolicy at risk
[webrtc-pc] Pull Request: maxSimulcastStreams attribute in RTCRtpCodecCapability
[webrtc-pc] Pull Request: Mention that codecs can be reordered or removed but not modified.
[webrtc-pc] Pull Request: Specify how media is centered, cropped, and scaled. Fixes #305
[webrtc-pc] Pull Request: Specify how transceivers get their mids in setLocal/Remote
[webrtc-pc] Pull Request: Specify when random mid generation happens
[webrtc-pc] Pull Request: strawman text to show how unverfieid media would work
[webrtc-pc] Pull Request: Switch to sender and receiver getStats selectors.
[webrtc-pc] Receive a track multiple times
[webrtc-pc] RTCRtcpMuxPolicy of "negotiate" should be marked as an "at-risk" feature
[webrtc-pc] RTCStats timestamp source ambiguous
[webrtc-pc] sdpFmtpLine isn't very convenient
[webrtc-pc] Separated auth dictionaries for STUN/TURN (issue 714)
[webrtc-pc] Should IDP Login error cary the idpLoginUrl
[webrtc-pc] Specify how media is centered, cropped, and scaled. Fixes #305
[webrtc-pc] Specify when a data channel's ID is assigned, and what the `id` attribute returns when no ID is assigned.
[webrtc-pc] strawman text to show how unverfieid media would work
- Bernard Aboba via GitHub (Wednesday, 15 February)
- Martin Thomson via GitHub (Wednesday, 15 February)
- stefan hakansson via GitHub (Wednesday, 15 February)
- Cullen Jennings via GitHub (Wednesday, 15 February)
- stefan hakansson via GitHub (Wednesday, 15 February)
- Roman Shpount via GitHub (Tuesday, 14 February)
- stefan hakansson via GitHub (Tuesday, 14 February)
- Roman Shpount via GitHub (Tuesday, 14 February)
- stefan hakansson via GitHub (Tuesday, 14 February)
- Roman Shpount via GitHub (Sunday, 12 February)
- stefan hakansson via GitHub (Sunday, 12 February)
- Cullen Jennings via GitHub (Sunday, 12 February)
[webrtc-pc] Support assertions that identify the recipient
[webrtc-pc] Support for OAuth in TURN credentials (Issue 714 patch)
[webrtc-pc] Switch to sender and receiver getStats selectors.
[webrtc-stats] Add stat to RTCTransportStats for ICE role (controlling or controlled)
[webrtc-stats] Adding remoteTimestamp to RTCRtpStreamStats.
[webrtc-stats] aligning codec types with webrtc-pc
[webrtc-stats] Consistent marker for "non-active" object?
[webrtc-stats] Definitions from MSE need re-targeting
[webrtc-stats] frameWidth/frameHeight: use last decoded value
[webrtc-stats] Make a RTCMediaStreamTrackStats object per track attachment
[webrtc-stats] moving roundTripTime from outbound to inbound
[webrtc-stats] Need advice for handling obsolete stats
[webrtc-stats] Need consistent name for RTT estimate on RTCRTPStreamStats and on ICECandidatePair
[webrtc-stats] Need priority information for MediaStreamTracks
[webrtc-stats] new commits pushed by alvestrand
[webrtc-stats] new commits pushed by vr000m
[webrtc-stats] Not clear how to differentiate between received connectivity checks and consent requests
[webrtc-stats] Privacy & Security self review
[webrtc-stats] Pull Request: Adding remoteTimestamp to RTCRtpStreamStats.
[webrtc-stats] Pull Request: aligning codec types with webrtc-pc
[webrtc-stats] Pull Request: Changes type of RTCRTPStreamStats.ssrc from string to unsigned long.
[webrtc-stats] Pull Request: fixes issue with TURN URL protocol
[webrtc-stats] Pull Request: frameWidth/frameHeight: use last decoded value
[webrtc-stats] Pull Request: Issue 97
[webrtc-stats] Pull Request: moving roundTripTime from outbound to inbound
[webrtc-stats] Pull Request: Remove separation of received consent and connectivity requests.
[webrtc-stats] Pull Request: removed cancelled and renamed inprogress with in-progress.
[webrtc-stats] Related to outgoing and incoming bitrate estimates on a candidate pair
[webrtc-stats] Remove references saying "defines an API"
[webrtc-stats] Remove separation of received consent and connectivity requests.
[webrtc-stats] RTCCodecStats needs `transportId` and `isRemote` to give it context
[webrtc-stats] RTCCodecStats.implementation... per-codec, per-stream?
[webrtc-stats] RTCIceCandidatePairStats packet counters
[webrtc-stats] RTCIceCandidatePairStats.writable/readable: redundant?
[webrtc-stats] RTCPeerConnection.getStats: What to do with 'selector' argument?
[webrtc-stats] RTCTransportStats packet counters
[webrtc-stats] Stat for adaptation reason
- Robin Schriebman via GitHub (Thursday, 23 February)
- Varun Singh via GitHub (Wednesday, 15 February)
- henbos via GitHub (Wednesday, 15 February)
- henbos via GitHub (Tuesday, 14 February)
- henbos via GitHub (Tuesday, 14 February)
- Harald Alvestrand via GitHub (Tuesday, 7 February)
- henbos via GitHub (Tuesday, 7 February)
- Harald Alvestrand via GitHub (Tuesday, 7 February)
- henbos via GitHub (Monday, 6 February)
- henbos via GitHub (Monday, 6 February)
[webrtc-stats] Stat for audio acceleration rate?
[webrtc-stats] Stat for audio acceleration/expand rate?
[webrtc-stats] Stat for audio playout delay
[webrtc-stats] Stat for camera input framerate of local video tracks
[webrtc-stats] Stat for how many adaptation changes occur for a video track
[webrtc-stats] Stat for how many adaption changes occur for a video track
[webrtc-stats] Stat for how many audio stream packets are expanded when packets are lost (and lost and the user is speaking)
[webrtc-stats] Stat for how many audio stream packets are expanded when the user is speaking
[webrtc-stats] Stat for how much time it takes to encode video
[webrtc-stats] Stat for likelihood of echo
[webrtc-stats] Stat for retransmitted bytes
[webrtc-stats] Stat for target and actual encoding bitrate
[webrtc-stats] Stats to keep track of sync between audio and video
- Stefan Holmer via GitHub (Friday, 24 February)
- henbos via GitHub (Friday, 24 February)
- Robin Schriebman via GitHub (Thursday, 23 February)
- henbos via GitHub (Tuesday, 21 February)
- Stefan Holmer via GitHub (Tuesday, 21 February)
- henbos via GitHub (Thursday, 16 February)
- Stefan Holmer via GitHub (Friday, 10 February)
- henbos via GitHub (Friday, 10 February)
- Robin Schriebman via GitHub (Friday, 10 February)
- Stefan Holmer via GitHub (Thursday, 9 February)
- Robin Schriebman via GitHub (Thursday, 9 February)
- Harald Alvestrand via GitHub (Wednesday, 8 February)
- Stefan Holmer via GitHub (Wednesday, 8 February)
- henbos via GitHub (Wednesday, 8 February)
- Stefan Holmer via GitHub (Wednesday, 8 February)
- henbos via GitHub (Wednesday, 8 February)
- henbos via GitHub (Wednesday, 8 February)
- Stefan Holmer via GitHub (Tuesday, 7 February)
- Robin Schriebman via GitHub (Monday, 6 February)
- henbos via GitHub (Monday, 6 February)
- henbos via GitHub (Monday, 6 February)
- Robin Schriebman via GitHub (Saturday, 4 February)
- henbos via GitHub (Friday, 3 February)
[webrtc-stats] Unclear if the request encompasses consent checks or not
[webrtc-stats] Unclear which framerate "framePerSecond" represents
[webrtc-stats] What should `availableOutgoingBitrate`/`availableOutgoingBitrate` be for candidate pairs not in use?
[webrtc-stats] Why is ssrc a DOMString?
Closed: [webrtc-pc] "track" event shouldn't fire for a "recvonly" description.
Closed: [webrtc-pc] Ambiguous wording in addIceCandidate
Closed: [webrtc-pc] Change SetLocalDescription to require unchanged offer/answer string
Closed: [webrtc-pc] currentRemoteDescription.sdp -- does it need to match the last SDP set via setRemoteDescription?
Closed: [webrtc-pc] Effect of a BYE on RtpReceiver.track
Closed: [webrtc-pc] Event when a transceiver is stopped via remote action
Closed: [webrtc-pc] Inconsistencies in description of RTCDTMFToneChangeEvent.tone
Closed: [webrtc-pc] Indicators of usage and data flow (PING review)
Closed: [webrtc-pc] Information available prior to permission prompt (PING review)
Closed: [webrtc-pc] Need Custom Error for IdP
Closed: [webrtc-pc] Need IDP Expired Token error
Closed: [webrtc-pc] Need IDP Invalid Token
Closed: [webrtc-pc] offerToReceiveAudio/offerToReceiveVideo remain in implementations (likely needed for compat)
Closed: [webrtc-pc] OpenSource WebRTC IdP
Closed: [webrtc-pc] Processing remote MediaStreamTracks without MediaStreams info
Closed: [webrtc-pc] Receive a track multiple times
Closed: [webrtc-pc] RTCRtcpMuxPolicy of "negotiate" should be marked as an "at-risk" feature
Closed: [webrtc-pc] The RTCPeerConnectionIceErrorEvent constructor should have an optional init dict
Closed: [webrtc-stats] Discrepancies between RTCRtpCodecParameters and RTCCodecStats naming
Closed: [webrtc-stats] Need consistent name for RTT estimate on RTCRTPStreamStats and on ICECandidatePair
Closed: [webrtc-stats] Need description for `remoteSource` member of `RTCMediaStreamTrackStats`
Closed: [webrtc-stats] RTCCodecStats.implementation... per-codec, per-stream?
Closed: [webrtc-stats] RTCRTPStreamStats.mediaTrackId, RTCMediaStreamStats.trackIds
Closed: [webrtc-stats] Stat for audio acceleration/expand rate?
Closed: [webrtc-stats] Stat for target and actual encoding bitrate
Closed: [webrtc-stats] Where is the API?
Closed: [webrtc-stats] Why is ssrc a DOMString?
Last message date: Tuesday, 28 February 2017 20:55:05 UTC