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

Thursday, 30 March 2017
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Re: [webrtc-pc] Specify an AllowUnverifiedMedia RTCConfiguration property Peter Thatcher via GitHub
- Re: [webrtc-stats] Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. jan-ivar via GitHub
- Re: [webrtc-pc] strawman text to show how unverified media would work Peter Thatcher via GitHub
- Re: [webrtc-pc] Specify an AllowUnverifiedMedia RTCConfiguration property Peter Thatcher via GitHub
- Re: [webrtc-stats] Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. jan-ivar via GitHub
- [webrtc-pc] Pull Request: Add missing "closed" signaling state. Taylor Brandstetter via GitHub
- [webrtc-pc] Diagram for RTCSignalingState includes "closed" state, which doesn't exist? Taylor Brandstetter via GitHub
- Re: [webrtc-stats] Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. Cullen Jennings via GitHub
- Re: [webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible Bernard Aboba via GitHub
- Re: [webrtc-stats] Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. Harald Alvestrand via GitHub
- Re: [webrtc-pc] Don't link function names to legacy section (unless intended) stefan hakansson via GitHub
- Re: [webrtc-stats] Change log, and make tidy Harald Alvestrand via GitHub
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-stats] RoundTripTime not defined when the underlying stream can't calculate it Harald Alvestrand via GitHub
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Re: [webrtc-pc] Update for structured cloning changes in HTML Adam Bergkvist via GitHub
- [webrtc-pc] Pull Request: Don't link function names to legacy section (unless intended) Adam Bergkvist via GitHub
- Re: [webrtc-pc] Clarify when RTCRtpContributingSource.audioLevel can be null. stefan hakansson via GitHub
- Re: [webrtc-pc] Always update the RTCRtpContributingSource for SSRCs. stefan hakansson via GitHub
- Re: [webrtc-pc] Attempt to update RTCRtpContributingSource objects at playout time. stefan hakansson via GitHub
- Re: [webrtc-pc] RTP/RTCP non-mux: feature at risk stefan hakansson via GitHub
- Re: [webrtc-pc] Need to specify which members of the encodings in "sendEncodings" are actually used Bernard Aboba via GitHub
Wednesday, 29 March 2017
- Re: [webrtc-pc] RTCRtpContributingSource naming Taylor Brandstetter via GitHub
- [webrtc-pc] RTCRtpContributingSource naming Philipp Hancke via GitHub
- Re: [webrtc-pc] Need to specify which members of the encodings in "sendEncodings" are actually used Peter Thatcher via GitHub
- Re: [webrtc-stats] example 8.2: calculating fraction lost vs fractionLost stat Varun Singh via GitHub
- Re: [webrtc-pc] RTCRtpContributingSource.audioLevel not guaranteed to be in sync with audio playout Philipp Hancke via GitHub
- [webrtc-pc] Pull Request: Clarify when RTCRtpContributingSource.audioLevel can be null. Taylor Brandstetter via GitHub
- [webrtc-pc] Pull Request: Always update the RTCRtpContributingSource for SSRCs. Taylor Brandstetter via GitHub
- Re: [webrtc-pc] RTCRtpContributingSource.audioLevel not guaranteed to be in sync with audio playout Taylor Brandstetter via GitHub
- [webrtc-pc] Pull Request: Attempt to update RTCRtpContributingSource objects at playout time. Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Make legacy API optional to implement Philip Jägenstedt via GitHub
- Re: [webrtc-pc] Need to specify which members of the encodings in "sendEncodings" are actually used Taylor Brandstetter via GitHub
Tuesday, 28 March 2017
- Re: [webrtc-pc] Make legacy API optional to implement Harald Alvestrand via GitHub
- Re: [webrtc-pc] Make legacy API optional to implement Philip Jägenstedt via GitHub
- Re: [webrtc-pc] Make legacy API optional to implement Philip Jägenstedt via GitHub
Monday, 27 March 2017
- [webrtc-stats] example 8.2: calculating fraction lost vs fractionLost stat Philipp Hancke via GitHub
Sunday, 26 March 2017
- Re: [webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible Lennart Grahl via GitHub
- [webrtc-pc] Pull Request: RTP/RTCP non-mux: feature at risk Bernard Aboba via GitHub
Friday, 24 March 2017
- Re: [webrtc-pc] DTLS failures Bernard Aboba via GitHub
- Re: [webrtc-stats] Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. Harald Alvestrand via GitHub
- Re: [webrtc-stats] Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. jan-ivar via GitHub
- Re: [webrtc-stats] Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. jan-ivar via GitHub
- Re: [webrtc-stats] Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. jan-ivar via GitHub
- Re: [webrtc-stats] Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. Harald Alvestrand via GitHub
- Re: [webrtc-stats] Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. jan-ivar via GitHub
- Re: [webrtc-stats] Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. jesup via GitHub
Thursday, 23 March 2017
- Re: [webrtc-stats] Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. jan-ivar via GitHub
- Re: [webrtc-pc] DTLS failures Roman Shpount via GitHub
- Re: [webrtc-stats] Remove availableIncomingBitrate? Varun Singh via GitHub
- Re: [webrtc-pc] Make legacy API optional to implement youennf via GitHub
- Re: [webrtc-stats] Remove availableIncomingBitrate? Bernard Aboba via GitHub
- Re: [webrtc-stats] Remove availableIncomingBitrate? Bernard Aboba via GitHub
- Re: [webrtc-stats] Stat for audio playout delay Bernard Aboba via GitHub
- Re: [webrtc-pc] Make legacy API optional to implement Philip Jägenstedt via GitHub
- Re: [webrtc-pc] Sender/Receiver.rtcpTransport: feature at risk? Harald Alvestrand via GitHub
- Re: [webrtc-pc] Make legacy API optional to implement Harald Alvestrand via GitHub
- [webrtc-pc] new commits pushed by taylor-b Taylor Brandstetter via GitHub
- Closed: [webrtc-pc] Don't fire events on a closed peer connection Harald Alvestrand via GitHub
- [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Re: [webrtc-pc] Handling RTX in RTCRtpCodecCapabilities Bernard Aboba via GitHub
- Re: [webrtc-pc] Check RTCPeerConnection isClosed slot before running queued tasks stefan hakansson via GitHub
- [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible Harald Alvestrand via GitHub
- [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Re: [webrtc-pc] DTLS failures Taylor Brandstetter via GitHub
- [webrtc-pc] Pull Request: Remove last use of 'set of receivers' Adam Bergkvist via GitHub
- [webrtc-pc] editorial: Some links to setLocalDescription points to the legacy extensions Adam Bergkvist via GitHub
- [webrtc-pc] Pull Request: Check RTCPeerConnection isClosed slot before running queued tasks Adam Bergkvist via GitHub
- [webrtc-pc] Sender/Receiver.rtcpTransport: feature at risk? Bernard Aboba via GitHub
- [webrtc-pc] DTLS failures Bernard Aboba via GitHub
- Re: [webrtc-pc] Align getAlgorithm return value with Web Crypto Philip Jägenstedt via GitHub
- Closed: [webrtc-pc] Align getAlgorithm return value with Web Crypto Philip Jägenstedt via GitHub
- Re: [webrtc-pc] Align getAlgorithm return value with Web Crypto Bernard Aboba via GitHub
- Re: [webrtc-pc] RTCRtpContributingSource.audioLevel not guaranteed to be in sync with audio playout Taylor Brandstetter via GitHub
Wednesday, 22 March 2017
- Re: [webrtc-stats] Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. Taylor Brandstetter via GitHub
- Re: [webrtc-stats] Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. jan-ivar via GitHub
- Re: [webrtc-stats] Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. Harald Alvestrand via GitHub
- [webrtc-stats] Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing. jan-ivar via GitHub
- Re: [webrtc-stats] rtt undefined when no RTCP RR Varun Singh via GitHub
- Re: [webrtc-stats] rtt undefined when no RTCP RR jesup via GitHub
- Re: [webrtc-stats] rtt undefined when no RTCP RR Varun Singh via GitHub
- [webrtc-stats] Pull Request: rtt undefined when no RTCP RR Varun Singh via GitHub
- Re: [webrtc-stats] Bandwidth estimations again (Issue 97 redux) Varun Singh via GitHub
- Closed: [webrtc-stats] Timestamp in the getStats Harald Alvestrand via GitHub
- Re: [webrtc-stats] Stats report for RTCRtpContributingSource objects Harald Alvestrand via GitHub
- Re: [webrtc-stats] Stats report for RTCRtpContributingSource objects Varun Singh via GitHub
- Re: [webrtc-stats] RTCMediaStreamTrackStats members to keep track of audio and video sync. Harald Alvestrand via GitHub
- Re: [webrtc-stats] Change log, and make tidy Harald Alvestrand via GitHub
- Re: [webrtc-stats] Added RTCInboundRTPStreamStats sample counters. Harald Alvestrand via GitHub
- Re: [webrtc-stats] Add RTCOutboundRTPStreamStats.totalEncodeTime Varun Singh via GitHub
- Re: [webrtc-stats] Add RTCOutboundRTPStreamStats.totalEncodeTime Harald Alvestrand via GitHub
- Re: [webrtc-stats] Bandwidth estimations again (Issue 97 redux) Harald Alvestrand via GitHub
- [webrtc-stats] Pull Request: RTCMediaStreamTrackStats members to keep track of audio and video sync. henbos via GitHub
- [webrtc-stats] Pull Request: Change log, and make tidy Harald Alvestrand via GitHub
- Re: [webrtc-pc] Make "candidate" non-nullable in addIceCandidate parameter table. Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible Taylor Brandstetter via GitHub
- Re: [webrtc-pc] When should RTCRtpContributingSource#audioLevel be null? Bernard Aboba via GitHub
- Re: [webrtc-pc] When exactly is an SSRC RTCRtpContributingSource object updated? Bernard Aboba via GitHub
- Re: [webrtc-pc] When exactly is an SSRC RTCRtpContributingSource object updated? Bernard Aboba via GitHub
- Re: [webrtc-pc] When exactly is an SSRC RTCRtpContributingSource object updated? Taylor Brandstetter via GitHub
Tuesday, 21 March 2017
- Re: [webrtc-pc] When exactly is an SSRC RTCRtpContributingSource object updated? Harald Alvestrand via GitHub
- Re: [webrtc-pc] When should RTCRtpContributingSource#audioLevel be null? Taylor Brandstetter via GitHub
- [webrtc-pc] When exactly is an SSRC RTCRtpContributingSource object updated? Taylor Brandstetter via GitHub
- Re: [webrtc-pc] When should RTCRtpContributingSource#audioLevel be null? Harald Alvestrand via GitHub
- Re: [webrtc-pc] When should RTCRtpContributingSource#audioLevel be null? Taylor Brandstetter via GitHub
- [webrtc-stats] Pull Request: Added RTCInboundRTPStreamStats sample counters. henbos via GitHub
- Re: [webrtc-pc] When should RTCRtpContributingSource#audioLevel be null? Philip Jägenstedt via GitHub
- [webrtc-stats] Pull Request: Add RTCOutboundRTPStreamStats.totalEncodeTime henbos via GitHub
- Re: [webrtc-pc] When should RTCRtpContributingSource#audioLevel be null? Bernard Aboba via GitHub
- Re: [webrtc-pc] Note At-Risk features at front of document stefan hakansson via GitHub
- [webrtc-pc] When should RTCRtpContributingSource#audioLevel be null? Philip Jägenstedt via GitHub
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-stats] Need advice for handling obsolete stats Harald Alvestrand via GitHub
Monday, 20 March 2017
- [webrtc-pc] Update for structured cloning changes in HTML Domenic Denicola via GitHub
- Re: [webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible Lennart Grahl via GitHub
Sunday, 19 March 2017
- Re: [webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible Bernard Aboba via GitHub
- [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
Saturday, 18 March 2017
- Re: [webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible Bernard Aboba via GitHub
- Re: [webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible Lennart Grahl via GitHub
- Re: [webrtc-pc] Section 12.2.1.1: enum errorDetail definition Bernard Aboba via GitHub
- Re: [webrtc-pc] Need to specify which members of the encodings in "sendEncodings" are actually used Bernard Aboba via GitHub
- Re: [webrtc-pc] Need to specify which members of the encodings in "sendEncodings" are actually used Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Need to specify which members of the encodings in "sendEncodings" are actually used Bernard Aboba via GitHub
Friday, 17 March 2017
- Re: [webrtc-pc] Need to specify which members of the encodings in "sendEncodings" are actually used Taylor Brandstetter via GitHub
- [webrtc-pc] Pull Request: Make "candidate" non-nullable in addIceCandidate parameter table. Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible Taylor Brandstetter via GitHub
- [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
- Re: [webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible Lennart Grahl via GitHub
- Re: [webrtc-pc] Specify behavior if browser doesn't implement "negotiate" rtcpMuxPolicy. Taylor Brandstetter via GitHub
- Re: [webrtc-pc] RTCRtpContributingSource.audioLevel not guaranteed to be in sync with audio playout Taylor Brandstetter via GitHub
- [webrtc-pc] Pull Request: Update call flow in Section 11.6 Bernard Aboba via GitHub
- Re: [webrtc-pc] Make legacy API optional to implement Philipp Hancke via GitHub
- [webrtc-pc] Make legacy API optional to implement youennf via GitHub
- Re: [webrtc-pc] RTCRtpContributingSource.audioLevel not guaranteed to be in sync with audio playout jesup via GitHub
Thursday, 16 March 2017
- Re: [webrtc-pc] Can candidate.ufrag be null? Taylor Brandstetter via GitHub
- Closed: [webrtc-pc] Can candidate.ufrag be null? Taylor Brandstetter via GitHub
- Re: [webrtc-pc] RTCRtpContributingSource.audioLevel not guaranteed to be in sync with audio playout Taylor Brandstetter via GitHub
- [webrtc-stats] Stats report for RTCRtpContributingSource objects Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Can candidate.ufrag be null? Lennart Grahl via GitHub
- [webrtc-pc] RTCRtpContributingSource.audioLevel not guaranteed to be in sync with audio playout Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible Taylor Brandstetter via GitHub
- [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
- [webrtc-pc] new commits pushed by burnburn Daniel Burnett via GitHub
- [webrtc-pc] new commits pushed by taylor-b Taylor Brandstetter via GitHub
- [webrtc-pc] Note At-Risk features at front of document Daniel Burnett via GitHub
- [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
- Closed: [webrtc-pc] STUN/TURN OAuth token auth parameter passing Bernard Aboba via GitHub
- Re: [webrtc-pc] Add Taylor as editor of the spec Dominique Hazael-Massieux via GitHub
- Re: [webrtc-pc] Add Taylor as editor of the spec Bernard Aboba via GitHub
- Re: [webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible Lennart Grahl via GitHub
- Re: [webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible Lennart Grahl via GitHub
- Re: [webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible Lennart Grahl via GitHub
- Re: [webrtc-pc] Can candidate.ufrag be null? Lennart Grahl via GitHub
Wednesday, 15 March 2017
- Re: [webrtc-pc] Add Taylor as editor of the spec Bernard Aboba via GitHub
- [webrtc-pc] Pull Request: Add Taylor as editor of the spec Bernard Aboba via GitHub
- [webrtc-pc] Pull Request: Handling RTX in RTCRtpCodecCapabilities Bernard Aboba via GitHub
- [webrtc-pc] Pull Request: Clarify which "candidate" is referred to in addIceCandidate description. Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible Taylor Brandstetter via GitHub
- [webrtc-pc] new commits pushed by taylor-b Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Can candidate.ufrag be null? Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Can candidate.ufrag be null? Taylor Brandstetter via GitHub
- [webrtc-pc] Add Taylor as editor of the spec Dominique Hazael-Massieux via GitHub
- [webrtc-stats] Pull Request: Issue 97 redux Varun Singh via GitHub
- Re: [webrtc-stats] Related to outgoing and incoming bitrate estimates on a candidate pair Varun Singh via GitHub
- [webrtc-pc] Handling RTX in RTCRtpCodecCapabilities Bernard Aboba via GitHub
- Re: [webrtc-stats] Related to outgoing and incoming bitrate estimates on a candidate pair Varun Singh via GitHub
- [webrtc-pc] Can candidate.ufrag be null? Lennart Grahl via GitHub
- Re: [webrtc-stats] Related to outgoing and incoming bitrate estimates on a candidate pair Harald Alvestrand via GitHub
- [webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible Lennart Grahl via GitHub
- Re: [webrtc-stats] Add section on obsoleted stats. Varun Singh via GitHub
- Re: [webrtc-pc] Handing SDP with more than one identity stefan hakansson via GitHub
Tuesday, 14 March 2017
- Re: [webrtc-stats] RTCIceCandidatePairStats.writable/readable: redundant? Taylor Brandstetter via GitHub
- [webrtc-pc] Pull Request: Add getStats() to sender and receiver. jan-ivar via GitHub
- [webrtc-pc] getStats on sender and receiver. jan-ivar via GitHub
- Re: [webrtc-stats] Add section on obsoleted stats. Harald Alvestrand via GitHub
- [webrtc-pc] Mark Identity as feature at risk? Dominique Hazael-Massieux via GitHub
- Closed: [webrtc-stats] RTCPeerConnection.getStats: What to do with 'selector' argument? Harald Alvestrand via GitHub
- Re: [webrtc-stats] RTCPeerConnection.getStats: What to do with 'selector' argument? Harald Alvestrand via GitHub
- Re: [webrtc-stats] Make sure timing and responsibility for stat object creation is clear Harald Alvestrand via GitHub
- Closed: [webrtc-stats] Make sure timing and responsibility for stat object creation is clear Harald Alvestrand via GitHub
- Re: [webrtc-stats] Consider making (aggregate) stats more accessible Harald Alvestrand via GitHub
- Closed: [webrtc-stats] Consider making (aggregate) stats more accessible Harald Alvestrand via GitHub
- Closed: [webrtc-stats] Expose protocol used to communicate with TURN server Varun Singh via GitHub
- Closed: [webrtc-stats] Consider putting roundTripTime on RTCInboundStreamStats Varun Singh via GitHub
- Closed: [webrtc-stats] There is no ICE "cancelled" state Varun Singh via GitHub
- Re: [webrtc-stats] Packets or frames discarded on send? Harald Alvestrand via GitHub
- Re: [webrtc-stats] Unclear if the request encompasses consent checks or not Varun Singh via GitHub
- Re: [webrtc-stats] Unclear if the request encompasses consent checks or not Harald Alvestrand via GitHub
- Closed: [webrtc-stats] Unclear if the request encompasses consent checks or not Varun Singh via GitHub
- Re: [webrtc-stats] RTCIceCandidatePairStats.writable/readable: redundant? Varun Singh via GitHub
- Re: [webrtc-stats] RTCIceCandidatePairStats.writable/readable: redundant? Harald Alvestrand via GitHub
- Re: [webrtc-stats] RTCIceCandidatePairStats.writable/readable: redundant? Harald Alvestrand via GitHub
- Re: [webrtc-stats] RTCIceCandidatePairStats.writable/readable: redundant? Harald Alvestrand via GitHub
- Re: [webrtc-stats] Stat for adaptation reason Varun Singh via GitHub
- Re: [webrtc-stats] Stat for adaptation reason Harald Alvestrand via GitHub
- Re: [webrtc-stats] Stat for retransmitted bytes Harald Alvestrand via GitHub
- Re: [webrtc-stats] Add section on obsoleted stats. Harald Alvestrand via GitHub
- Re: [webrtc-stats] RoundTripTime not defined when the underlying stream can't calculate it Varun Singh via GitHub
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
Monday, 13 March 2017
- [webrtc-pc] Need to specify which members of the encodings in "sendEncodings" are actually used Taylor Brandstetter via GitHub
- Re: [webrtc-pc] "Hybrid" OAuth solution. Taylor Brandstetter via GitHub
- [webrtc-stats] Pull Request: Add section on obsoleted stats. Harald Alvestrand via GitHub
- Re: [webrtc-pc] Specify behavior if browser doesn't implement "negotiate" rtcpMuxPolicy. stefan hakansson via GitHub
- Re: [webrtc-pc] Freeing resources for incoming stream stefan hakansson via GitHub
- Re: [webrtc-pc] Remove Section 12.2.1.1 errorDetailEnum stefan hakansson via GitHub
- Re: [webrtc-pc] Remove Section 11.6 note stefan hakansson via GitHub
- Re: [webrtc-pc] Update Call Flow Browser to Browser stefan hakansson via GitHub
- Re: [webrtc-pc] Mark getAlgorithm method at risk stefan hakansson via GitHub
- Re: [webrtc-pc] "Hybrid" OAuth solution. stefan hakansson via GitHub
- [webrtc-pc] new commits pushed by adam-be Adam Bergkvist via GitHub
- [webrtc-pc] new commits pushed by adam-be Adam Bergkvist via GitHub
Saturday, 11 March 2017
- Re: [webrtc-pc] "Hybrid" OAuth solution. misi via GitHub
- [webrtc-pc] Pull Request: Remove Section 11.6 note Bernard Aboba via GitHub
Friday, 10 March 2017
- Re: [webrtc-stats] Packets or frames discarded on send? Varun Singh via GitHub
- Re: [webrtc-pc] Switching to new, consistent terminology when talking about exceptions. Dominique Hazael-Massieux via GitHub
Thursday, 9 March 2017
- Re: [webrtc-pc] Specify behavior if browser doesn't implement "negotiate" rtcpMuxPolicy. Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Specify behavior if browser doesn't implement "negotiate" rtcpMuxPolicy. Bernard Aboba via GitHub
- [webrtc-pc] Pull Request: Specify behavior if browser doesn't implement "negotiate" rtcpMuxPolicy. Taylor Brandstetter via GitHub
- [webrtc-pc] Pull Request: Update Call Flow Browser to Browser Bernard Aboba via GitHub
- Re: [webrtc-pc] "Hybrid" OAuth solution. Taylor Brandstetter via GitHub
- [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Re: [webrtc-pc] Add stats selection algorithm based on sender or receiver of selector. jan-ivar via GitHub
- [webrtc-pc] Pull Request: Freeing resources for incoming stream Bernard Aboba via GitHub
- Re: [webrtc-pc] Section 10.3: Freeing resources for incoming stream Bernard Aboba via GitHub
- [webrtc-pc] Pull Request: Remove Section 12.2.1.1 errorDetailEnum Bernard Aboba via GitHub
- [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
- [webrtc-pc] Pull Request: Mark getAlgorithm method at risk Bernard Aboba via GitHub
- Re: [webrtc-pc] Section 10.3: Freeing resources for incoming stream Harald Alvestrand via GitHub
- Re: [webrtc-pc] Generation of candidates Harald Alvestrand via GitHub
- Re: [webrtc-pc] Switching to new, consistent terminology when talking about exceptions. 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
- Closed: [webrtc-pc] Range checking needed for iceCandidatePoolSize Harald Alvestrand via GitHub
- [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-pc] Description of handling iceCandidatePoolSize in setConfiguration is out of sync with JSEP Harald Alvestrand via GitHub
- [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-pc] RTCDataChannelInit.id should use [EnforceRange] Harald Alvestrand via GitHub
- [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-pc] sdpFmtpLine isn't very convenient Harald Alvestrand via GitHub
- Re: [webrtc-pc] "Hybrid" OAuth solution. Harald Alvestrand via GitHub
- Re: [webrtc-pc] Add stats selection algorithm based on sender or receiver of selector. Harald Alvestrand via GitHub
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-stats] RTCDataChannelStats should reference RTCTransportStats Harald Alvestrand via GitHub
- Re: [webrtc-pc] RtpSender and track Harald Alvestrand via GitHub
- Re: [webrtc-pc] Section 11.6: Issue 6 Bernard Aboba via GitHub
- Re: [webrtc-pc] Changing iceCandidatePoolSize to an octet and adding EnforceRange. Bernard Aboba via GitHub
- [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
Wednesday, 8 March 2017
- Re: [webrtc-pc] Need to specify what happens if the browser doesn't implement "negotiate" rtcpMuxPolicy Taylor Brandstetter via GitHub
- [webrtc-pc] Pull Request: RtpSender and track Bernard Aboba via GitHub
- Re: [webrtc-pc] Need to specify what happens if the browser doesn't implement "negotiate" rtcpMuxPolicy Bernard Aboba via GitHub
- Re: [webrtc-stats] RoundTripTime not defined when the underlying stream can't calculate it jesup via GitHub
- [webrtc-pc] Need to specify what happens if the browser doesn't implement "negotiate" rtcpMuxPolicy Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Attaching the same track to multiple RTCRtpSenders: spec contradiction. Bernard Aboba via GitHub
- Re: [webrtc-pc] Throw InvalidModificationError if changing pool size after SLD. Bernard Aboba via GitHub
- [webrtc-pc] new commits pushed by taylor-b Taylor Brandstetter via GitHub
- Re: [webrtc-stats] Adding remoteTimestamp to RTCRtpStreamStats. Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Throw InvalidModificationError if changing pool size after SLD. Taylor Brandstetter via GitHub
- [webrtc-pc] Attaching the same track to multiple RTCRtpSenders: spec contradiction. henbos via GitHub
- Re: [webrtc-pc] Changing iceCandidatePoolSize to an octet and adding EnforceRange. jan-ivar via GitHub
- Re: [webrtc-pc] Section 10.3: Freeing resources for incoming stream Harald Alvestrand via GitHub
- Re: [webrtc-pc] Section 10.3: Freeing resources for incoming stream Andreas Pehrson via GitHub
- Re: [webrtc-pc] Section 10.3: Freeing resources for incoming stream Harald Alvestrand via GitHub
- Re: [webrtc-stats] Stat for likelihood of echo Harald Alvestrand 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) Harald Alvestrand via GitHub
- Re: [webrtc-stats] Stat for how much time it takes to encode video Harald Alvestrand via GitHub
- Re: [webrtc-stats] Stat for how much time it takes to encode video Harald Alvestrand via GitHub
- Re: [webrtc-stats] Stat for how many adaptation changes occur for a video track Harald Alvestrand via GitHub
- Re: [webrtc-stats] Stat for adaptation reason Harald Alvestrand via GitHub
- Re: [webrtc-stats] Add link from datachannel to transport Harald Alvestrand via GitHub
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-stats] Not clear how to differentiate between received connectivity checks and consent requests Harald Alvestrand via GitHub
- Re: [webrtc-stats] Adding remoteTimestamp to RTCRtpStreamStats. Harald Alvestrand via GitHub
- Re: [webrtc-stats] RoundTripTime not defined when the underlying stream can't calculate it henbos via GitHub
- Re: [webrtc-pc] Changing iceCandidatePoolSize to an octet and adding EnforceRange. jan-ivar via GitHub
- Re: [webrtc-pc] Changing iceCandidatePoolSize to an octet and adding EnforceRange. Bernard Aboba via GitHub
- Re: [webrtc-pc] Adding "[EnforceRange]" to RTCDataChannelInit.id. Bernard Aboba via GitHub
- Re: [webrtc-pc] Throw InvalidModificationError if changing pool size after SLD. Bernard Aboba via GitHub
Tuesday, 7 March 2017
- Closed: [webrtc-pc] Section 10.3: Bernard Aboba via GitHub
- [webrtc-pc] Section 10.3: Freeing resources for incoming stream Bernard Aboba via GitHub
- [webrtc-pc] Section 10.3: Bernard Aboba via GitHub
- Re: [webrtc-pc] Section 10.3: mute signal Bernard Aboba via GitHub
- Re: [webrtc-pc] Section 10.3: Issue 5 Bernard Aboba via GitHub
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-stats] Need descriptions for `label`, `protocol` and `state` members of `RTCDataChannelStats` Harald Alvestrand via GitHub
- Re: [webrtc-stats] RoundTripTime not defined when the underlying stream can't calculate it jan-ivar via GitHub
- Re: [webrtc-pc] Section 10.3: mute signal Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Section 11.6: Issue 6 Philipp Hancke via GitHub
- Re: [webrtc-pc] Section 10.3: mute signal Bernard Aboba via GitHub
- Re: [webrtc-pc] Section 10.3: Issue 5 Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Section 10.3: mute signal Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Section 11.6: Issue 6 Bernard Aboba via GitHub
- Re: [webrtc-stats] RoundTripTime not defined when the underlying stream can't calculate it henbos via GitHub
- Re: [webrtc-stats] RoundTripTime not defined when the underlying stream can't calculate it jan-ivar via GitHub
- Re: [webrtc-stats] RoundTripTime not defined when the underlying stream can't calculate it henbos via GitHub
- Re: [webrtc-pc] Align getAlgorithm return value with Web Crypto Philip Jägenstedt via GitHub
- [webrtc-pc] Pull Request: Remove connecting event from Event summary Bernard Aboba via GitHub
- [webrtc-pc] Pull Request: mute signal Bernard Aboba via GitHub
- Re: [webrtc-pc] Section 10.3: mute signal Bernard Aboba via GitHub
- Re: [webrtc-pc] Section 13: Connecting event Bernard Aboba via GitHub
- [webrtc-pc] Pull Request: Generation of candidates Bernard Aboba via GitHub
- Re: [webrtc-pc] Section 5.6: Generation of candidates Bernard Aboba via GitHub
- Re: [webrtc-stats] RoundTripTime not defined when the underlying stream can't calculate it jan-ivar via GitHub
- [webrtc-pc] Pull Request: Add clockrate, channels, sdpFmtpLine to codec capability Bernard Aboba via GitHub
Monday, 6 March 2017
- [webrtc-pc] Pull Request: Switching to new, consistent terminology when talking about exceptions. Taylor Brandstetter via GitHub
- Re: [webrtc-stats] RoundTripTime not defined when the underlying stream can't calculate it Varun Singh via GitHub
- Re: [webrtc-stats] RoundTripTime not defined when the underlying stream can't calculate it Varun Singh via GitHub
- [webrtc-stats] RoundTripTime not defined when the underlying stream can't calculate it jesup via GitHub
- [webrtc-pc] new commits pushed by dontcallmedom Dominique Hazael-Massieux via GitHub
- Re: [webrtc-pc] Add stats selection algorithm based on sender or receiver of selector. jan-ivar via GitHub
- [webrtc-stats] Pull Request: Adds definitions for RTCDataChannelStats members. Harald Alvestrand via GitHub
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
- [webrtc-stats] Packets or frames discarded on send? henbos via GitHub
- Re: [webrtc-pc] Align getAlgorithm return value with Web Crypto Harald Alvestrand via GitHub
Sunday, 5 March 2017
- [webrtc-pc] new commits pushed by taylor-b Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Align getAlgorithm return value with Web Crypto Bernard Aboba via GitHub
- Re: [webrtc-pc] Align getAlgorithm return value with Web Crypto Philip Jägenstedt via GitHub
Friday, 3 March 2017
- [webrtc-pc] Pull Request: Changing iceCandidatePoolSize to an octet and adding EnforceRange. Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Range checking needed for iceCandidatePoolSize Cullen Jennings via GitHub
- [webrtc-pc] Pull Request: Throw InvalidModificationError if changing pool size after SLD. Taylor Brandstetter via GitHub
- [webrtc-pc] Section 11.6: Issue 6 Bernard Aboba via GitHub
- [webrtc-pc] Section 10.3: Issue 5 Bernard Aboba via GitHub
- [webrtc-pc] Section 10.3: mute signal Bernard Aboba via GitHub
- [webrtc-pc] Section 5.6: Generation of candidates Bernard Aboba via GitHub
- [webrtc-pc] Description of handling iceCandidatePoolSize in setConfiguration is out of sync with JSEP Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Range checking needed for iceCandidatePoolSize Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Range checking needed for iceCandidatePoolSize Philip Jägenstedt via GitHub
- Closed: [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) Dominique Hazael-Massieux via GitHub
- Re: [webrtc-pc] Clarifying exactly what "sdpFmtpLine" represents. henbos via GitHub
Thursday, 2 March 2017
- [webrtc-pc] Range checking needed for iceCandidatePoolSize Taylor Brandstetter via GitHub
- [webrtc-pc] Pull Request: Adding "[EnforceRange]" to RTCDataChannelInit.id. Taylor Brandstetter via GitHub
- [webrtc-pc] RTCDataChannelInit.id should use [EnforceRange] Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Clarifying exactly what "sdpFmtpLine" represents. Bernard Aboba via GitHub
- [webrtc-pc] Pull Request: Clarifying exactly what "sdpFmtpLine" represents. Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Section 13: Connecting event 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
- Closed: [webrtc-pc] Advanced Peer-to-peer Example Harald Alvestrand via GitHub
- Re: [webrtc-pc] Clarify which timestamp RTCStats.timestamp represents. Harald Alvestrand via GitHub
- [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-pc] RTCStats timestamp source ambiguous Harald Alvestrand via GitHub
- Re: [webrtc-pc] Add an explicit stats selection algorithm. Harald Alvestrand via GitHub
- Closed: [webrtc-pc] Need to specify precisely when MID generation happens Harald Alvestrand via GitHub
- [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-pc] Integrate RTCRtpTransceiver into set local/remote steps Harald Alvestrand via GitHub
- [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-pc] Specify when a data channel's ID is assigned, and what the `id` attribute returns when no ID is assigned. Harald Alvestrand via GitHub
- [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-pc] Need to describe that codecs can be removed or reordered, but not modified Harald Alvestrand via GitHub
- [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-pc] Describe what happens when media changes Harald Alvestrand via GitHub
- [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Re: [webrtc-stats] Add link from datachannel to transport Harald Alvestrand via GitHub
- [webrtc-stats] Pull Request: Add link from datachannel to transport Harald Alvestrand via GitHub
- Re: [webrtc-pc] sdpFmtpLine isn't very convenient Taylor Brandstetter via GitHub
Wednesday, 1 March 2017
- Re: [webrtc-pc] Add clockrate, channels, sdpFmtpLine to codec capability Iñaki Baz Castillo via GitHub
- [webrtc-pc] Pull Request: Add clockrate, channels, sdpFmtpLine to codec capability Bernard Aboba via GitHub
- Re: [webrtc-pc] Specify when random mid generation happens Bernard Aboba via GitHub
- Re: [webrtc-pc] Label 'Warm-up example' as 'advanced p2p example' Bernard Aboba via GitHub
- Re: [webrtc-pc] Make RTCDataChannel.id nullable and describe when it's set. Bernard Aboba via GitHub
- Re: [webrtc-pc] Specify how media is centered, cropped, and scaled. Fixes #305 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] Specify how media is centered, cropped, and scaled. Fixes #305 Cullen Jennings via GitHub
- [webrtc-pc] Pull Request: Label 'Warm-up example' as 'advanced p2p example' Adam Bergkvist via GitHub
- Re: [webrtc-pc] Advanced Peer-to-peer Example Adam Bergkvist via GitHub
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-stats] Track stats: track or attachment? Harald Alvestrand via GitHub
- Re: [webrtc-stats] Remove separation of received consent and connectivity requests. Harald Alvestrand via GitHub
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-stats] "stats object" terminology is confusing Harald Alvestrand via GitHub
- Closed: [webrtc-stats] Are the stats read-only snapshots or will they change kept around? Harald Alvestrand via GitHub
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Re: [webrtc-pc] Specify how media is centered, cropped, and scaled. Fixes #305 Cullen Jennings via GitHub
- [webrtc-pc] setCodecPreferences() can't handle codecs supporting multiple clock rates/packetization-mode Bernard Aboba via GitHub

Last message date: Thursday, 30 March 2017 23:18:20 UTC
Archived on: Saturday, 6 May 2023 21:19:40 UTC

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