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

Tuesday, 31 October 2017
- Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case Nils Ohlmeier via GitHub
- Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case Lennart Grahl via GitHub
- Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case Nils Ohlmeier via GitHub
- Re: [webrtc-pc] JSEP references are out dated Vivien Lacourba via GitHub
- [webrtc-pc] JSEP references are out dated Vivien Lacourba via GitHub
- [webrtc-pc] new commits pushed by vivienlacourba Vivien Lacourba via GitHub
- Re: [webrtc-pc] Specify how RTCSctpTransport.maxMessageSize gets its value Lennart Grahl via GitHub
- [webrtc-pc] Pull Request: Specify how RTCSctpTransport.maxMessageSize gets its value Adam Bergkvist via GitHub
- [webrtc-pc] Pull Request: Add a new value to RTCIceTransportPolicy. jianjunz via GitHub
Monday, 30 October 2017
- Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case Taylor Brandstetter via GitHub
- Re: [webrtc-stats] jitterBufferOutput added henbos via GitHub
- Re: [webrtc-stats] jitterBufferDelay and concealed samples, DTX/CNG samples Varun Singh via GitHub
- Re: [webrtc-stats] jitterBufferOutput added Varun Singh via GitHub
- Re: [webrtc-stats] jitterBufferOutput added henbos via GitHub
- [webrtc-pc] new commits pushed by dontcallmedom Dominique Hazael-Massieux via GitHub
- [webrtc-pc] new commits pushed by adam-be Adam Bergkvist via GitHub
- Re: [webrtc-pc] Stats & isolated streams Dominique Hazael-Massieux via GitHub
- Re: [webrtc-pc] format & fixup Example code Jxck via GitHub
- Re: [webrtc-pc] format & fixup Example code Adam Bergkvist via GitHub
Sunday, 29 October 2017
- Re: [webrtc-stats] jitterBufferOutput added Varun Singh via GitHub
- [webrtc-stats] new commits pushed by vr000m Varun Singh via GitHub
- Closed: [webrtc-stats] RTCMediaStreamTrackStats.jitterBufferDelay: Specify unit in seconds Varun Singh via GitHub
Saturday, 28 October 2017
- Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case Lennart Grahl via GitHub
Friday, 27 October 2017
- Re: [webrtc-pc] Clarify whether RTCRtpContributingSource members are live. Taylor Brandstetter via GitHub
- Re: [webrtc-stats] RTCQualityLimitationReason and friends henbos via GitHub
- Re: [webrtc-stats] Adding "networkType" field to RTCIceCandidateStats. Harald Alvestrand via GitHub
- Re: [webrtc-stats] Adding "networkType" field to RTCIceCandidateStats. Harald Alvestrand via GitHub
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-stats] packetsLost is unsigned but can be negative according to RFC Harald Alvestrand via GitHub
- Re: [webrtc-stats] signed packetsLost Harald Alvestrand via GitHub
- Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case Lennart Grahl via GitHub
- Re: [webrtc-pc] format & fixup Example code Jxck via GitHub
- Re: [webrtc-pc] Validate protocol string in IdP operations Soares Chen via GitHub
- Re: [webrtc-pc] format & fixup Example code Adam Bergkvist via GitHub
- Re: [webrtc-pc] Adding more values to RTCIceTransportPolicy Enum jianjunz via GitHub
- Re: [webrtc-stats] RTCQualityLimitationReason and friends henbos via GitHub
- Re: [webrtc-stats] Stats for adaptation reason, for realsies henbos via GitHub
- [webrtc-stats] Pull Request: RTCQualityLimitationReason and friends henbos via GitHub
- Re: [webrtc-stats] Stats for adaptation reason, for realsies henbos via GitHub
- Re: [webrtc-stats] Stats for adaptation reason, for realsies henbos via GitHub
- Re: [webrtc-pc] format & fixup Example code Jxck via GitHub
- Re: [webrtc-stats] RTCMediaStreamTrackStats.audioLevel clarification henbos via GitHub
- [webrtc-stats] Pull Request: signed packetsLost henbos via GitHub
Thursday, 26 October 2017
- [webrtc-stats] Pull Request: jitterBufferOutput added henbos via GitHub
- Re: [webrtc-stats] jitterBufferDelay and concealed samples, DTX/CNG samples henbos via GitHub
- [webrtc-stats] Pull Request: jitterBufferDelay in seconds henbos via GitHub
- Re: [webrtc-pc] Adding more values to RTCIceTransportPolicy Enum Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Clarify whether RTCRtpContributingSource members are live. na-g via GitHub
- Re: [webrtc-pc] Throw error if data channel's buffer is filled, rather than closing. Lennart Grahl via GitHub
- Closed: [webrtc-pc] pc.close() method should close sctp transport Bernard Aboba via GitHub
- Re: [webrtc-pc] Clarify whether RTCRtpContributingSource members are live. Bernard Aboba via GitHub
- Closed: [webrtc-pc] No way to tell when SCTP association is closed Harald Alvestrand via GitHub
- Re: [webrtc-pc] offerToReceiveAudio/Video processing creates sendrecv transceivers Harald Alvestrand via GitHub
- Re: [webrtc-pc] Stats & isolated streams Harald Alvestrand via GitHub
- Re: [webrtc-pc] Need for Initial Bitrate by the Application/RtpSender? Bernard Aboba via GitHub
- Re: [webrtc-pc] Isolated Media Streams requires modification on permission algorithms in GUM and Permissions specs Harald Alvestrand via GitHub
- Re: [webrtc-pc] Isolated Media Streams requires modification on permission algorithms in GUM and Permissions specs Bernard Aboba via GitHub
- Re: [webrtc-pc] Adding more values to RTCIceTransportPolicy Enum Bernard Aboba via GitHub
- Re: [webrtc-pc] Isolated Media Streams requires modification on permission algorithms in GUM and Permissions specs Bernard Aboba via GitHub
- Closed: [webrtc-pc] Missing detail on obtaining lastOffer/lastAnswer in setLocalDescription Bernard Aboba via GitHub
- [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
- [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
- Re: [webrtc-pc] Validate protocol string in IdP operations Bernard Aboba via GitHub
- Re: [webrtc-pc] Validate protocol string in IdP operations stefan hakansson via GitHub
- Re: [webrtc-pc] Prepare status of the document for CR publication Bernard Aboba via GitHub
- Re: [webrtc-pc] Replace setDirection() with writable direction attribute Adam Bergkvist via GitHub
- [webrtc-pc] Pull Request: Replace setDirection() with writable direction attribute Adam Bergkvist via GitHub
- Re: [webrtc-pc] Isolated Media Streams requires modification on permission algorithms in GUM and Permissions specs Harald Alvestrand via GitHub
- Re: [webrtc-stats] packetsLost is unsigned but can be negative according to RFC Harald Alvestrand via GitHub
- [webrtc-pc] Isolated Media Streams requires modification on permission algorithms in GUM and Permissions specs Soares Chen via GitHub
- [webrtc-pc] getIdentityAssertion should reject with InvalidAccessError if no identity provider is set Soares Chen via GitHub
- Re: [webrtc-pc] Adding more values to RTCIceTransportPolicy Enum jianjunz via GitHub
- Re: [webrtc-pc] Clarify whether RTCRtpContributingSource members are live. Harald Alvestrand via GitHub
- Re: [webrtc-pc] Why is setDirection a method? Adam Bergkvist via GitHub
- Re: [webrtc-pc] Clarify whether RTCRtpContributingSource members are live. Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Adding more values to RTCIceTransportPolicy Enum Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Clarify whether RTCRtpContributingSource members are live. Cullen Jennings via GitHub
- Re: [webrtc-stats] RTCMediaStreamTrackStats.audioLevel clarification Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Adding more values to RTCIceTransportPolicy Enum jianjunz via GitHub
- Re: [webrtc-pc] Clarify whether RTCRtpContributingSource members are live. Taylor Brandstetter via GitHub
- Re: [webrtc-stats] Need DSCP information for incoming RTP streams Taylor Brandstetter via GitHub
Wednesday, 25 October 2017
- Re: [webrtc-stats] Audio/Video sync follow-up Robin Schriebman via GitHub
- Re: [webrtc-pc] Why is setDirection a method? Philipp Hancke via GitHub
- Re: [webrtc-pc] Why is setDirection a method? Harald Alvestrand via GitHub
- Re: [webrtc-stats] add packetsDuplicated as a result of #253 Harald Alvestrand via GitHub
- Re: [webrtc-pc] Adding more values to RTCIceTransportPolicy Enum jianjunz via GitHub
- Re: [webrtc-stats] Stats for adaptation reason, for realsies Harald Alvestrand via GitHub
- Re: [webrtc-pc] Why is setDirection a method? jan-ivar via GitHub
- Re: [webrtc-stats] Add estimatedClockSkew Harald Alvestrand via GitHub
- Re: [webrtc-stats] RTCMediaStreamTrackStats.audioLevel clarification Harald Alvestrand via GitHub
- Re: [webrtc-stats] Definitions from MSE need re-targeting Harald Alvestrand via GitHub
- Re: [webrtc-stats] Need DSCP information for incoming RTP streams Harald Alvestrand via GitHub
- Re: [webrtc-stats] Need DSCP information for outgoing RTP streams Harald Alvestrand via GitHub
- Closed: [webrtc-stats] Add "privacy" to security considerations Harald Alvestrand via GitHub
- Re: [webrtc-stats] Add "privacy" to security considerations Harald Alvestrand via GitHub
- Re: [webrtc-stats] What is fractionLost for a local incoming media stream? Harald Alvestrand via GitHub
- Re: [webrtc-pc] Adding more values to RTCIceTransportPolicy Enum Harald Alvestrand via GitHub
- Re: [webrtc-stats] jitterBufferDelay and concealed samples, DTX/CNG samples Harald Alvestrand via GitHub
- Re: [webrtc-stats] Is bytesReceived really available for RTCRemoteInboundRTPStreamStats? Harald Alvestrand via GitHub
- Re: [webrtc-stats] We need "sender" and "receiver" stats, not "track" stats Harald Alvestrand via GitHub
- Re: [webrtc-stats] RTCMediaStreamTrackStats is four dictionaries in one Harald Alvestrand via GitHub
- Re: [webrtc-stats] Interframe delay stat for video receive stream. ilnik via GitHub
- Re: [webrtc-stats] Stats for Audio network adaptation Harald Alvestrand via GitHub
- Re: [webrtc-stats] Interframe delay stat for video receive stream. Harald Alvestrand via GitHub
- Re: [webrtc-stats] Audio/Video sync follow-up Harald Alvestrand via GitHub
- Re: [webrtc-stats] TAG review: Link to design principles Harald Alvestrand via GitHub
- [webrtc-stats] Pull Request: TAG review: Link to design principles Harald Alvestrand via GitHub
- [webrtc-pc] Adding more values to RTCIceTransportPolicy Enum jianjunz via GitHub
Tuesday, 24 October 2017
- Re: [webrtc-pc] format & fixup Example code Jxck via GitHub
- Re: [webrtc-pc] Why is setDirection a method? Taylor Brandstetter via GitHub
- [webrtc-stats] new commits pushed by henbos henbos via GitHub
- Closed: [webrtc-stats] Difference between 'Id' and 'Identifier' fields henbos via GitHub
- Re: [webrtc-pc] Why is setDirection a method? Harald Alvestrand via GitHub
- Re: [webrtc-stats] Adding "networkType" field to RTCIceCandidateStats. Taylor Brandstetter via GitHub
- Re: [webrtc-stats] Move to continuous publication Harald Alvestrand via GitHub
- Closed: [webrtc-stats] Move to continuous publication Harald Alvestrand via GitHub
- [webrtc-pc] Why is setDirection a method? jan-ivar via GitHub
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Re: [webrtc-stats] Move to continuous publication Harald Alvestrand via GitHub
- Re: [webrtc-stats] "Make tidy" is broken? Vivien Lacourba via GitHub
- Closed: [webrtc-stats] "Make tidy" is broken? Vivien Lacourba via GitHub
- Re: [webrtc-pc] Clarify whether RTCRtpContributingSource members are live. na-g via GitHub
- [webrtc-stats] new commits pushed by vivienlacourba Vivien Lacourba via GitHub
- Re: [webrtc-stats] Difference between 'Id' and 'Identifier' fields Harald Alvestrand via GitHub
- [webrtc-stats] Pull Request: Add a paragraph about "identifier" when "id" is occupied Harald Alvestrand via GitHub
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
- [webrtc-stats] "Make tidy" is broken? Harald Alvestrand via GitHub
- [webrtc-pc] new commits pushed by adam-be Adam Bergkvist via GitHub
- [webrtc-stats] Pull Request: Added 'objectDeleted' attribute Harald Alvestrand via GitHub
- Re: [webrtc-stats] Add "privacy" to security considerations Harald Alvestrand via GitHub
- Re: [webrtc-stats] Complete security and privacy considerations based on self-review Harald Alvestrand via GitHub
- Re: [webrtc-pc] format & fixup Example code Adam Bergkvist via GitHub
- [webrtc-stats] new commits pushed by vivienlacourba Vivien Lacourba via GitHub
- [webrtc-stats] Pull Request: Fix respec issue: Multiple s for and Vivien Lacourba via GitHub
- [webrtc-stats] new commits pushed by vivienlacourba Vivien Lacourba via GitHub
- Re: [webrtc-stats] Adding "networkType" field to RTCIceCandidateStats. Harald Alvestrand via GitHub
- [webrtc-stats] Consider the Reporting API as a stats export mechanism Harald Alvestrand via GitHub
- Re: [webrtc-pc] format & fixup Example code Adam Bergkvist via GitHub
- Re: [webrtc-stats] Adding "networkType" field to RTCIceCandidateStats. Varun Singh via GitHub
Monday, 23 October 2017
- Re: [webrtc-stats] Adding "networkType" field to RTCIceCandidateStats. Taylor Brandstetter via GitHub
- Re: [webrtc-pc] OAUTH-POP-KEY-DISTRIBUTION IETF draft has been replaced by ACE-CWT-PROOF-OF-POSSESSION Cullen Jennings 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
- Re: [webrtc-stats] Adding "networkType" field to RTCIceCandidateStats. Taylor Brandstetter via GitHub
- Re: [webrtc-stats] RTCMediaStreamTrackStats.audioLevel clarification Taylor Brandstetter via GitHub
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
- Closed: [webrtc-stats] Add stat for SRTP decryption failures Harald Alvestrand via GitHub
- Re: [webrtc-stats] jitterBufferDelay and concealed samples, DTX/CNG samples Taylor Brandstetter via GitHub
- Re: [webrtc-stats] Added Guidelines for getStats() results caching Harald Alvestrand via GitHub
- Closed: [webrtc-stats] RTCMediaStreamTrackStats.keyFramesReceived Harald Alvestrand via GitHub
- [webrtc-stats] new commits pushed by alvestrand Harald Alvestrand via GitHub
- [webrtc-pc] OAUTH-POP-KEY-DISTRIBUTION IETF draft is has been replaced by ACE-CWT-PROOF-OF-POSSESSION Vivien Lacourba via GitHub
- Re: [webrtc-pc] Section 5.2: centering, scaling, cropping stefan hakansson via GitHub
- [webrtc-pc] Pull Request: Note about video dimension adaptation discussion added. stefan hakansson via GitHub
- Re: [webrtc-pc] format & fixup Example code Jxck via GitHub
- Re: [webrtc-stats] Adding "networkType" field to RTCIceCandidateStats. Harald Alvestrand via GitHub
- Re: [webrtc-stats] RTCMediaStreamTrackStats.audioLevel clarification hlundin via GitHub
- Re: [webrtc-stats] jitterBufferDelay and concealed samples, DTX/CNG samples hlundin via GitHub
- [webrtc-stats] Pull Request: Issue 257 network type Taylor Brandstetter via GitHub
Saturday, 21 October 2017
- Re: [webrtc-stats] upgrade to Node 8 (respec now uses async/await) Marcos Cáceres via GitHub
- Re: [webrtc-pc] Need for Initial Bitrate by the Application/RtpSender? Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Need for Initial Bitrate by the Application/RtpSender? Varun Singh via GitHub
- [webrtc-stats] Add estimatedClockSkew henbos via GitHub
- Re: [webrtc-stats] Audio/Video sync follow-up henbos via GitHub
- Re: [webrtc-stats] Audio/Video sync follow-up henbos via GitHub
- Re: [webrtc-stats] Audio/Video sync follow-up henbos via GitHub
- Re: [webrtc-stats] Audio/Video sync follow-up henbos via GitHub
- Re: [webrtc-stats] Add stat for network type of ICE candidate henbos via GitHub
- Re: [webrtc-pc] Need for Initial Bitrate by the Application/RtpSender? Taylor Brandstetter via GitHub
- Re: [webrtc-stats] jitterBufferDelay and concealed samples, DTX/CNG samples henbos via GitHub
- Re: [webrtc-pc] Need for Initial Bitrate by the Application/RtpSender? Varun Singh via GitHub
- Re: [webrtc-stats] jitterBufferDelay and concealed samples, DTX/CNG samples Taylor Brandstetter via GitHub
- Re: [webrtc-stats] RTCMediaStreamTrackStats.audioLevel clarification Taylor Brandstetter via GitHub
- Re: [webrtc-stats] RTCMediaStreamTrackStats.audioLevel clarification henbos via GitHub
Friday, 20 October 2017
- Re: [webrtc-stats] Difference between 'Id' and 'Identifier' fields henbos via GitHub
- Re: [webrtc-stats] Difference between 'Id' and 'Identifier' fields Varun Singh via GitHub
- Re: [webrtc-stats] Stats for adaptation reason, for realsies henbos via GitHub
- Re: [webrtc-stats] Stats for adaptation reason, for realsies henbos via GitHub
- [webrtc-stats] Add stat for network type of ICE candidate Taylor Brandstetter via GitHub
- Re: [webrtc-stats] Stats for adaptation reason, for realsies henbos via GitHub
- [webrtc-pc] offerToReceiveAudio/Video processing creates sendrecv transceivers docfaraday via GitHub
- Re: [webrtc-stats] Stats for adaptation reason, for realsies henbos via GitHub
- Re: [webrtc-stats] Stats for adaptation reason, for realsies Varun Singh via GitHub
- Re: [webrtc-stats] Stats for adaptation reason, for realsies henbos via GitHub
- Closed: [webrtc-stats] Stat for adaptation reason henbos via GitHub
- Re: [webrtc-stats] Stat for adaptation reason henbos via GitHub
- Closed: [webrtc-stats] Stat for how many adaptation changes occur for a video track henbos via GitHub
- Re: [webrtc-stats] Stat for how many adaptation changes occur for a video track henbos via GitHub
- [webrtc-stats] Stats for adaptation reason, for realsies henbos via GitHub
- Re: [webrtc-stats] keyFramesSent and keyFramesReceived added henbos via GitHub
- [webrtc-stats] What is fractionLost for a local incoming media stream? Varun Singh via GitHub
- Re: [webrtc-stats] add packetsDuplicated as a result of #253 Varun Singh via GitHub
- [webrtc-stats] add packetsDuplicated as a result of #253 Varun Singh via GitHub
- Re: [webrtc-stats] Added Guidelines for getStats() results caching henbos via GitHub
- Re: [webrtc-stats] packetsLost is unsigned but can be negative according to RFC Varun Singh via GitHub
- Re: [webrtc-stats] Add stats for the negotiated DTLS-SRTP and DTLS cipher suites. henbos via GitHub
- Re: [webrtc-stats] packetsLost is unsigned but can be negative according to RFC Varun Singh via GitHub
- Re: [webrtc-stats] Add "packetsFailedDecryption", to count SRTP decryption failures. Varun Singh via GitHub
- [webrtc-stats] packetsLost is unsigned but can be negative according to RFC henbos via GitHub
- Re: [webrtc-stats] upgrade to Node 8 (respec now uses async/await) Varun Singh via GitHub
- Re: [webrtc-stats] RTCMediaStreamTrackStats.concealedAudibleSamples added. Varun Singh via GitHub
- [webrtc-pc] new commits pushed by vivienlacourba Vivien Lacourba via GitHub
- [webrtc-pc] Pull Request: upgrade to Node 8 (respec 1.7 now uses async/await) Vivien Lacourba via GitHub
- Re: [webrtc-stats] upgrade to Node 8 (respec now uses async/await) Vivien Lacourba via GitHub
- Re: [webrtc-stats] upgrade to Node 8 (respec now uses async/await) Vivien Lacourba via GitHub
- Re: [webrtc-stats] upgrade to Node 8 (respec now uses async/await) Harald Alvestrand via GitHub
- [webrtc-stats] Pull Request: upgrade to Node 8 (respec now uses async/await) Vivien Lacourba via GitHub
- Re: [webrtc-stats] Privacy & Security self review Dominique Hazael-Massieux via GitHub
- Re: [webrtc-stats] Complete security and privacy considerations based on self-review Dominique Hazael-Massieux via GitHub
- [webrtc-stats] Pull Request: Complete security and privacy considerations based on self-review Dominique Hazael-Massieux via GitHub
- Re: [webrtc-pc] Let setLocalDescription use [[LastAnswer/Offer]] internal slots Harald Alvestrand via GitHub
- [webrtc-pc] Pull Request: Let setLocalDescription use [[LastAnswer/Offer]] internal slots Soares Chen via GitHub
Thursday, 19 October 2017
- Re: [webrtc-pc] format & fixup Example code Harald Alvestrand via GitHub
- Re: [webrtc-pc] format & fixup Example code Harald Alvestrand via GitHub
- Closed: [webrtc-pc] Need to describe when ICE and DTLS transport objects are created/changed Bernard Aboba via GitHub
- [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
- Closed: [webrtc-pc] When ICE restarts results in connection to a new endpoint 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
Wednesday, 18 October 2017
- Closed: [webrtc-pc] createDataChannel should use "in parallel" (editorial) jan-ivar via GitHub
- Re: [webrtc-pc] createDataChannel should use "in parallel" (editorial) jan-ivar via GitHub
- [webrtc-pc] createDataChannel should use "in parallel" (editorial) jan-ivar via GitHub
- Re: [webrtc-pc] Section 5.2: centering, scaling, cropping stefan hakansson via GitHub
- Re: [webrtc-stats] jitterBufferDelay and concealed samples, DTX/CNG samples hlundin via GitHub
Tuesday, 17 October 2017
- Re: [webrtc-stats] Add stat for SRTP decryption failures henbos via GitHub
- Re: [webrtc-pc] Section 5.2: centering, scaling, cropping Harald Alvestrand via GitHub
- Re: [webrtc-pc] Simple text for scaling issue - no letterbox allowed. Dominique Hazael-Massieux via GitHub
- Re: [webrtc-pc] Section 5.2: centering, scaling, cropping stefan hakansson via GitHub
- Re: [webrtc-pc] Section 5.2: centering, scaling, cropping Harald Alvestrand via GitHub
- Re: [webrtc-pc] Section 5.2: centering, scaling, cropping Harald Alvestrand via GitHub
Monday, 16 October 2017
- Re: [webrtc-stats] Add stat for SRTP decryption failures Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Section 5.2: centering, scaling, cropping Cullen Jennings via GitHub
- Re: [webrtc-pc] Section 5.2: centering, scaling, cropping stefan hakansson via GitHub
- Re: [webrtc-pc] Section 5.2: centering, scaling, cropping Harald Alvestrand via GitHub
- [webrtc-pc] Pull Request: Simple text for scaling issue - no letterbox allowed. Harald Alvestrand via GitHub
- Re: [webrtc-pc] Section 5.2: centering, scaling, cropping stefan hakansson via GitHub
- Re: [webrtc-pc] Section 5.2: centering, scaling, cropping Harald Alvestrand via GitHub
Sunday, 15 October 2017
- Re: [webrtc-stats] Add stat for SRTP decryption failures Philipp Hancke via GitHub
Saturday, 14 October 2017
- [webrtc-stats] Pull Request: Add "packetsFailedDecryption", to count SRTP decryption failures. Taylor Brandstetter via GitHub
Friday, 13 October 2017
- Re: [webrtc-pc] Need for Initial Bitrate by the Application/RtpSender? Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Need for Initial Bitrate by the Application/RtpSender? Varun Singh via GitHub
- Re: [webrtc-pc] Need for Initial Bitrate by the Application/RtpSender? Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Stats & isolated streams Dominique Hazael-Massieux via GitHub
- Re: [webrtc-pc] Stats & isolated streams Varun Singh via GitHub
- Re: [webrtc-pc] Resizing video (fix for Issue 1283) stefan hakansson via GitHub
- Re: [webrtc-pc] Section 5.2: centering, scaling, cropping stefan hakansson via GitHub
- Re: [webrtc-pc] Section 5.2: centering, scaling, cropping Justin Uberti via GitHub
- Re: [webrtc-pc] Need for Initial Bitrate by the Application/RtpSender? Justin Uberti via GitHub
Thursday, 12 October 2017
- Re: [webrtc-pc] Need for Initial Bitrate by the Application/RtpSender? Taylor Brandstetter via GitHub
- Re: [webrtc-stats] Add stats for the negotiated DTLS-SRTP and DTLS cipher suites. Taylor Brandstetter via GitHub
- [webrtc-pc] Need for Initial Bitrate? Varun Singh via GitHub
- Re: [webrtc-stats] jitterBufferDelay and concealed samples, DTX/CNG samples Harald Alvestrand via GitHub
- Re: [webrtc-stats] Add stats for the negotiated DTLS-SRTP and DTLS cipher suites. henbos via GitHub
- Re: [webrtc-stats] Add stats for the negotiated DTLS-SRTP and DTLS cipher suites. Harald Alvestrand via GitHub
- Re: [webrtc-stats] Add stat for SRTP decryption failures Harald Alvestrand via GitHub
- Re: [webrtc-stats] Add stat for SRTP decryption failures Varun Singh via GitHub
- [webrtc-pc] Pull Request: Add onstatechange event to SctpTransport Event table Bernard Aboba via GitHub
- [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
- Re: [webrtc-stats] Add stats for the negotiated DTLS-SRTP and DTLS cipher suites. Varun Singh via GitHub
- Re: [webrtc-stats] Add stat for SRTP decryption failures Varun Singh via GitHub
- [webrtc-pc] new commits pushed by alvestrand 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] RTCRtpSender.setParameters: Do we need more specific hardware encoder errors? Harald Alvestrand via GitHub
- Re: [webrtc-pc] Adopt "test as you commit" policy Anne van Kesteren via GitHub
- Re: [webrtc-pc] Adopt "test as you commit" policy stefan hakansson via GitHub
- Re: [webrtc-pc] Please do not use datatracker.ietf.org Anne van Kesteren via GitHub
- Closed: [webrtc-pc] Please do not use datatracker.ietf.org Dominique Hazael-Massieux via GitHub
- Re: [webrtc-pc] Please do not use datatracker.ietf.org Dominique Hazael-Massieux via GitHub
- Re: [webrtc-pc] Explain the scope of DTLS/ICE transport objects in more detail. stefan hakansson via GitHub
- Re: [webrtc-pc] Example ICE state transition appears incorrect Bernard Aboba via GitHub
- [webrtc-stats] Add stat for SRTP decryption failures Taylor Brandstetter via GitHub
Wednesday, 11 October 2017
- Re: [webrtc-pc] RTCPriorityType combines relative bitrate with QoS priority, which applications may not want. Taylor Brandstetter via GitHub
- [webrtc-pc] Pull Request: Explain the scope of DTLS/ICE transport objects in more detail. Taylor Brandstetter via GitHub
- [webrtc-pc] Example ICE state transition appears incorrect Taylor Brandstetter via GitHub
- Re: [webrtc-pc] RTCDataChannel's send() doesn't encode string data Anne van Kesteren via GitHub
- [webrtc-pc] Please do not use datatracker.ietf.org Anne van Kesteren via GitHub
- [webrtc-pc] RTCDataChannel's send() doesn't encode string data Anne van Kesteren via GitHub
- Re: [webrtc-pc] RTCIdentityProviderGlobalScope looks wrong Anne van Kesteren via GitHub
- [webrtc-pc] RTCIdentityProviderGlobalScope looks wrong Anne van Kesteren via GitHub
- [webrtc-pc] Instantiating an IdP Proxy should be defined in terms of Fetch Anne van Kesteren via GitHub
- Re: [webrtc-pc] RTCPriorityType combines relative bitrate with QoS priority, which applications may not want. stefan hakansson via GitHub
- Re: [webrtc-pc] RTCPriorityType combines relative bitrate with QoS priority, which applications may not want. Taylor Brandstetter via GitHub
- Re: [webrtc-pc] RTCPriorityType combines relative bitrate with QoS priority, which applications may not want. Bernard Aboba via GitHub
Tuesday, 10 October 2017
- [webrtc-stats] Add stats for the negotiated DTLS-SRTP and DTLS cipher suites. Taylor Brandstetter via GitHub
- Re: [webrtc-pc] RTCPriorityType combines relative bitrate with QoS priority, which applications may not want. Taylor Brandstetter via GitHub
- [webrtc-pc] RTCPriorityType combines relative bitrate with QoS priority, which applications may not want. Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Describe how transport objects are assigned to senders/receivers. Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Adopt "test as you commit" policy stefan hakansson via GitHub
- Re: [webrtc-pc] Adopt "test as you commit" policy Philip Jägenstedt via GitHub
- Re: [webrtc-pc] Adopt "test as you commit" policy stefan hakansson via GitHub
- Re: [webrtc-pc] Describe how transport objects are assigned to senders/receivers. stefan hakansson via GitHub
- Re: [webrtc-pc] Describe how transport objects are assigned to senders/receivers. Taylor Brandstetter via GitHub
- [webrtc-pc] "set an RTCSessionDescription" steps need to be broken up into multiple smaller lists of steps Taylor Brandstetter via GitHub
- [webrtc-pc] Pull Request: Describe how transport objects are assigned to senders/receivers. Taylor Brandstetter via GitHub
Monday, 9 October 2017
- Re: [webrtc-pc] Meta: auto-publish changes to the spec Cullen Jennings via GitHub
- [webrtc-pc] Pull Request: Set up automatic PR review request on webrtc.html Dominique Hazael-Massieux via GitHub
- Re: [webrtc-pc] Meta: auto-publish changes to the spec Philip Jägenstedt via GitHub
Sunday, 8 October 2017
- [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
Friday, 6 October 2017
- Closed: [webrtc-pc] How to find what keyGenAlgorithm values the browser supports? Bernard Aboba via GitHub
Thursday, 5 October 2017
- [webrtc-pc] Pull Request: pc.close() should close Data Channels and SctpTransports Bernard Aboba via GitHub
- [webrtc-pc] Pull Request: pc.close() should close Data Channels and SctpTransports Bernard Aboba via GitHub
- Closed: [webrtc-pc] pc.close() method should close sctp transport Bernard Aboba via GitHub
- Closed: [webrtc-pc] duration and interToneGap should be stored as internal slots of RTCDTMFSender Bernard Aboba via GitHub
- Closed: [webrtc-pc] Section 5.3: track.label initialization Bernard Aboba via GitHub
- Re: [webrtc-pc] Section 5.3: track.label initialization Bernard Aboba via GitHub
- Re: [webrtc-pc] toJSON Dfn needed Bernard Aboba via GitHub
- Closed: [webrtc-pc] toJSON Dfn needed Bernard Aboba via GitHub
- [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
- Closed: [webrtc-pc] Clarify offerToReceiveAudio and offerToReceiveVideo in renegotiation stefan hakansson via GitHub
- Re: [webrtc-pc] Clarify offerToReceiveAudio and offerToReceiveVideo in renegotiation stefan hakansson via GitHub
- Re: [webrtc-pc] offerToReceive* legacy behaviour spec does not match the behaviour of legacy implementations stefan hakansson via GitHub
- [webrtc-pc] new commits pushed by burnburn Daniel Burnett via GitHub
- Closed: [webrtc-pc] createDataChannel touches DOM "in the background" Bernard Aboba via GitHub
- [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
- [webrtc-pc] canInsertDTMF values not specified Bernard Aboba via GitHub
- Re: [webrtc-pc] Meta: auto-publish changes to the spec Dominique Hazael-Massieux via GitHub
- Re: [webrtc-pc] Meta: auto-publish changes to the spec Philip Jägenstedt via GitHub
- Re: [webrtc-pc] format & fixup Example code Dominique Hazael-Massieux via GitHub
- [webrtc-pc] Pull Request: format & fixup Example code Jxck via GitHub
Wednesday, 4 October 2017
- [webrtc-pc] Adopt "test as you commit" policy Philip Jägenstedt via GitHub
Tuesday, 3 October 2017
- Re: [webrtc-pc] Ambiguities with BUNDLE and ICE Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Ambiguities with BUNDLE and ICE Harald Alvestrand via GitHub
Monday, 2 October 2017
- Re: [webrtc-pc] No way to tell when SCTP association is closed Taylor Brandstetter via GitHub
- Re: [webrtc-pc] Ambiguities with BUNDLE and ICE Taylor Brandstetter via GitHub
- Re: [webrtc-stats] We need "sender" and "receiver" stats, not "track" stats jan-ivar via GitHub
- [webrtc-pc] Pull Request: setParameters: Use more specific hardware encoder errors Adam Bergkvist via GitHub
- Re: [webrtc-pc] No way to tell when SCTP association is closed stefan hakansson via GitHub
- Re: [webrtc-pc] Ambiguities with BUNDLE and ICE Harald Alvestrand via GitHub

Last message date: Tuesday, 31 October 2017 21:48:06 UTC
Archived on: Thursday, 24 March 2022 20:39:49 UTC

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