[webrtc-pc] onmute then onunmute can fire before negotiation completes
- Re: [webrtc-pc] onmute then onunmute can fire before negotiation completes
- Re: [webrtc-pc] onmute then onunmute can fire before negotiation completes
- Re: [webrtc-pc] onmute then onunmute can fire before negotiation completes
- Re: [webrtc-pc] onmute then onunmute can fire before negotiation completes
- Re: [webrtc-pc] onmute then onunmute can fire before negotiation completes
[webrtc-pc] new commits pushed by aboba
[webrtc-pc] Should rollback fire addtrack/removetrack events?
Re: [webrtc-pc] replaceTrack: Never negotiate when replacing an ended track?
[webrtc-pc] specify legacy onaddstream?
- Re: [webrtc-pc] specify legacy onaddstream?
- Re: [webrtc-pc] specify legacy onaddstream?
- Re: [webrtc-pc] specify legacy onaddstream?
- Re: [webrtc-pc] specify legacy onaddstream?
Closed: [webrtc-pc] Why is RTCRtpContributingSource.byte required-but-nullable?
Re: [webrtc-stats] RTCMediaStreamTrackStats.audioLevel clarification
[webrtc-stats] Pull Request: Additional description of audioLevel
Re: [webrtc-pc] replaceTrack: Clarify how the UA determins if negotiation is needed
[webrtc-pc] Pull Request: offerToReceive: Tweak attribute descriptions to be a bit more generic
[webrtc-pc] offerToReceive: Attribute description is only mentioning the the "true case"
[webrtc-pc] Re-adding a removed track causes InvalidSessionDescriptionError in spec
[webrtc-pc] new commits pushed by alvestrand
[webrtc-pc] Pull Request: Check that transceiver is not stopped before re-using.
- Re: [webrtc-pc] Check that transceiver is not stopped before using.
- Re: [webrtc-pc] Check that transceiver is not stopped before using.
Closed: [webrtc-pc] behaviour of offerToReceive* set to false when there is a local track
[webrtc-pc] Maximum message size slightly incorrect
[webrtc-pc] Data channel closing procedure
[webrtc-pc] Pull Request: RTPRtcSender: provided properties to get related SDP mline msid and media type
- Re: [webrtc-pc] RTPRtcSender: provided properties to get related SDP mline msid and media type
- Re: [webrtc-pc] RTPRtcSender: provided properties to get related MediaStreamTrackId and media type
- Re: [webrtc-pc] RTPRtcSender: provided properties to get related MediaStreamTrackId and media type
Re: [webrtc-pc] should offerToReceive* in createOffer be marked feature-at-risk?
Closed: [webrtc-pc] should offerToReceive* in createOffer be marked feature-at-risk?
[webrtc-pc] new commits pushed by aboba
Re: [webrtc-pc] Provide changelog for the spec in new editors draft workflow
- Re: [webrtc-pc] Provide changelog for the spec in new editors draft workflow
- Re: [webrtc-pc] Provide changelog for the spec in new editors draft workflow
Re: [webrtc-stats] Definitions from MSE need re-targeting
Closed: [webrtc-stats] Add estimatedClockSkew
[webrtc-stats] Pull Request: moved packetFailedDecryption counter to RTCInboundRTPStreamStats
[webrtc-stats] packetsFailedDecryption is not reported in a RTCP Report
[webrtc-stats] Pull Request: moved bytes and packets received counters
[webrtc-stats] Pull Request: fractionLost moved to RTCRemoteInbound
Re: [webrtc-stats] Issues raised in TAG review
Closed: [webrtc-stats] Issues raised in TAG review
[webrtc-stats] new commits pushed by henbos
Re: [webrtc-pc] OAUTH-POP-KEY-DISTRIBUTION IETF draft has been replaced by ACE-CWT-PROOF-OF-POSSESSION
[webrtc-pc] RTCCertificate Interface should (or should not) be backed up.
Re: [webrtc-stats] "record<RTCQualityLimitationReason, double> qualityLimitationDurations" has invalid type
- Re: [webrtc-stats] "record<RTCQualityLimitationReason, double> qualityLimitationDurations" has invalid type
- Re: [webrtc-stats] "record<RTCQualityLimitationReason, double> qualityLimitationDurations" has invalid type
- Re: [webrtc-stats] "record<RTCQualityLimitationReason, double> qualityLimitationDurations" has invalid type
[webrtc-pc] Pull Request: fix offerToReceive(Audio|Video): false
- Re: [webrtc-pc] fix offerToReceive(Audio|Video): false
- Re: [webrtc-pc] fix offerToReceive(Audio|Video): false
- Re: [webrtc-pc] Specify legacy behavior when offerToReceive* option is set to false
- Re: [webrtc-pc] Specify legacy behavior when offerToReceive* option is set to false
- Re: [webrtc-pc] Specify legacy behavior when offerToReceive* option is set to false
- Re: [webrtc-pc] Specify legacy behavior when offerToReceive* option is set to false
- Re: [webrtc-pc] Specify legacy behavior when offerToReceive* option is set to false
[webrtc-stats] Pull Request: qualityLimitationDurations record with DOMString key
- Re: [webrtc-stats] qualityLimitationDurations record with DOMString key
- Re: [webrtc-stats] qualityLimitationDurations record with DOMString key
- Re: [webrtc-stats] qualityLimitationDurations record with DOMString key
- Re: [webrtc-stats] qualityLimitationDurations record with DOMString key
[webrtc-stats] new commits pushed by alvestrand
Closed: [webrtc-stats] Consistent marker for "non-active" object?
[webrtc-stats] new commits pushed by henbos
[webrtc-stats] Pull Request: Use () instead of <> for record qualityLimitationDurations
Re: [webrtc-stats] Do the "audio level" stats include MediaStreamTrack volume settings?
Re: [webrtc-stats] Is keeping stats around a memory problem?
- Re: [webrtc-stats] Is keeping stats around a memory problem?
- Re: [webrtc-stats] Is keeping stats around a memory problem?
Closed: [webrtc-stats] RTCMediaStreamTrackStats.concealedAudibleSamples
Re: [webrtc-stats] RTCMediaStreamTrackStats.concealedAudibleSamples
[webrtc-stats] new commits pushed by alvestrand
Re: [webrtc-stats] Audio/Video sync follow-up
Re: [webrtc-stats] Interframe delay stat for video receive stream.
- Re: [webrtc-stats] Interframe delay stat for video receive stream.
- Re: [webrtc-stats] Interframe delay stat for video receive stream.
Re: [webrtc-stats] Stats for Audio network adaptation
- Re: [webrtc-stats] Stats for Audio network adaptation
- Re: [webrtc-stats] Stats for Audio network adaptation
Re: [webrtc-stats] Add estimatedClockSkew
Re: [webrtc-stats] Add per layer stats for SVC
[webrtc-stats] Add TLS version and EC group id to stats
- Re: [webrtc-stats] Add TLS version and EC group id to stats
- Re: [webrtc-stats] Add TLS version and EC group id to stats
- Re: [webrtc-stats] Add TLS version and EC group id to stats
- Re: [webrtc-stats] Add TLS version and EC group id to stats
Closed: [webrtc-stats] Add stats for the negotiated DTLS-SRTP and DTLS cipher suites.
[webrtc-stats] new commits pushed by vr000m
[webrtc-stats] new commits pushed by henbos
Closed: [webrtc-stats] jitterBufferDelay and concealed samples, DTX/CNG samples
[webrtc-pc] Pull Request: Rephrase RTCDataChannel.bufferedAmount description
- Re: [webrtc-pc] Rephrase RTCDataChannel.bufferedAmount description
- Re: [webrtc-pc] Rephrase RTCDataChannel.bufferedAmount description
Re: [webrtc-stats] Added 'objectDeleted' attribute
- Re: [webrtc-stats] Added 'objectDeleted' attribute
- Re: [webrtc-stats] Added 'objectDeleted' attribute
- Re: [webrtc-stats] Added 'objectDeleted' attribute
[webrtc-pc] When to fire events triggered by setRemoteDescription.
- Re: [webrtc-pc] When to fire events triggered by setRemoteDescription.
- Re: [webrtc-pc] When to fire events triggered by setRemoteDescription.
- Re: [webrtc-pc] When to fire events triggered by setRemoteDescription.
- Re: [webrtc-pc] When to fire events triggered by setRemoteDescription.
- Re: [webrtc-pc] When to fire events triggered by setRemoteDescription.
Re: [webrtc-pc] Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable?
- Re: [webrtc-pc] Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable?
- Re: [webrtc-pc] Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable?
- Re: [webrtc-pc] Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable?
- Re: [webrtc-pc] Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable?
- Re: [webrtc-pc] Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable?
- Re: [webrtc-pc] Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable?
- Re: [webrtc-pc] Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable?
- Re: [webrtc-pc] Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable?
- Re: [webrtc-pc] Why is RTCRtpSynchronizationSource.voiceActivityFlag required-but-nullable?
Re: [webrtc-pc] Why is RTCRtpContributingSource.byte required-but-nullable?
- Re: [webrtc-pc] Why is RTCRtpContributingSource.byte required-but-nullable?
- Re: [webrtc-pc] Why is RTCRtpContributingSource.byte required-but-nullable?
- Re: [webrtc-pc] Why is RTCRtpContributingSource.byte required-but-nullable?
- Re: [webrtc-pc] Why is RTCRtpContributingSource.byte required-but-nullable?
- Re: [webrtc-pc] Why is RTCRtpContributingSource.byte required-but-nullable?
- Re: [webrtc-pc] Why is RTCRtpContributingSource.byte required-but-nullable?
- Re: [webrtc-pc] Why is RTCRtpContributingSource.byte required-but-nullable?
- Re: [webrtc-pc] Why is RTCRtpContributingSource.byte required-but-nullable?
[webrtc-stats] What happens when a partial keyFrames is received?
- Re: [webrtc-stats] What happens when a partial keyFrames is received?
- Re: [webrtc-stats] What happens when a partial keyFrames is received?
Re: [webrtc-pc] RTCRtpContributingSource.timestamp needs a clearer definition
- Re: [webrtc-pc] RTCRtpContributingSource.timestamp needs a clearer definition
- Re: [webrtc-pc] RTCRtpContributingSource.timestamp needs a clearer definition
- Re: [webrtc-pc] RTCRtpContributingSource.timestamp needs a clearer definition
- Re: [webrtc-pc] RTCRtpContributingSource.timestamp needs a clearer definition
[webrtc-pc] Testing guidelines: Comments or help link?
[webrtc-pc] new commits pushed by alvestrand
Closed: [webrtc-pc] RTCPeerConnection constructor can fail - what error to return?
[webrtc-pc] new commits pushed by alvestrand
[webrtc-pc] new commits pushed by alvestrand
Closed: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case
[webrtc-pc] new commits pushed by alvestrand
Closed: [webrtc-pc] MTI specification of crypto for certs
Re: [webrtc-pc] MTI specification of crypto for certs
Closed: [webrtc-pc] More granular data channel error reporting
Re: [webrtc-pc] More granular data channel error reporting
[webrtc-pc] Pull Request: offerToReceive: Rewrite to handle two options to createOffer
- Re: [webrtc-pc] offerToReceive: Rewrite to handle two options to createOffer
- Re: [webrtc-pc] offerToReceive: Rewrite to handle two options to createOffer
- Re: [webrtc-pc] offerToReceive: Rewrite to handle two options to createOffer
- Re: [webrtc-pc] offerToReceive: Rewrite to handle two options to createOffer
- Re: [webrtc-pc] offerToReceive: Rewrite to handle two options to createOffer
- Re: [webrtc-pc] offerToReceive: Rewrite to handle two options to createOffer
[webrtc-pc] offerToReceive: Current text can't handle two options to createOffer
- Re: [webrtc-pc] offerToReceive: Current text can't handle two options to createOffer
- Closed: [webrtc-pc] offerToReceive: Current text can't handle two options to createOffer
[webrtc-pc] Pull Request: Fire removetrack/addtrack events before track events.
Re: [webrtc-pc] Ordering of stream "addtrack"/"removetrack" events vs. "track" event
Closed: [webrtc-pc] RTCTrackEvent's type parameter
Re: [webrtc-pc] RTCTrackEvent's type parameter
Re: [webrtc-pc] Minor inconsistencies around RTCSessionDescription vs. RTCSessionDescriptionInit
Re: [webrtc-pc] behaviour of offerToReceive* set to false when there is a local track
- Re: [webrtc-pc] behaviour of offerToReceive* set to false when there is a local track
- Re: [webrtc-pc] behaviour of offerToReceive* set to false when there is a local track
- Re: [webrtc-pc] behaviour of offerToReceive* set to false when there is a local track
- Re: [webrtc-pc] behaviour of offerToReceive* set to false when there is a local track
[webrtc-pc] Pull Request: fix offerToReceive(Audio|Video): false
- Re: [webrtc-pc] fix offerToReceive(Audio|Video): false
- Re: [webrtc-pc] fix offerToReceive(Audio|Video): false
- Re: [webrtc-pc] fix offerToReceive(Audio|Video): false
- Re: [webrtc-pc] fix offerToReceive(Audio|Video): false
[webrtc-pc] Pull Request: offerToReceive: s/transceiver type/transceiver kind
- Re: [webrtc-pc] offerToReceive: s/transceiver type/transceiver kind
- Re: [webrtc-pc] offerToReceive: s/transceiver type/transceiver kind
- Re: [webrtc-pc] offerToReceive: s/transceiver type/transceiver kind/ (and do it for kinds, not mediaTypes)
Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case
[webrtc-pc] RTCDataChannel.bufferedAmount description confusing
- Re: [webrtc-pc] RTCDataChannel.bufferedAmount description confusing
- Re: [webrtc-pc] RTCDataChannel.bufferedAmount description confusing
- Re: [webrtc-pc] RTCDataChannel.bufferedAmount description confusing
- Re: [webrtc-pc] RTCDataChannel.bufferedAmount description confusing
[webrtc-stats] Pull Request: Add packetsDuplicated
- Re: [webrtc-stats] Add packetsDuplicated
- Re: [webrtc-stats] Add packetsDuplicated
- Re: [webrtc-stats] Add packetsDuplicated
- Re: [webrtc-stats] Add packetsDuplicated
- Re: [webrtc-stats] Add packetsDuplicated
- Re: [webrtc-stats] Add packetsDuplicated
[webrtc-stats] Pull Request: Adding dtlsCipher and srtpCipher
- Re: [webrtc-stats] Adding dtlsCipher and srtpCipher
- Re: [webrtc-stats] Adding dtlsCipher and srtpCipher
Re: [webrtc-stats] RTCMediaStreamTrackStats.concealedAudibleSamples added.
Re: [webrtc-stats] jitterBufferEmittedCount added (jitterBufferOutput)
- Re: [webrtc-stats] jitterBufferEmittedCount added (jitterBufferOutput)
- Re: [webrtc-stats] jitterBufferEmittedCount added (jitterBufferOutput)
- Re: [webrtc-stats] jitterBufferEmittedCount added (jitterBufferOutput)
Re: [webrtc-stats] RTCQualityLimitationReason and friends
- Re: [webrtc-stats] RTCQualityLimitationReason and friends
- Re: [webrtc-stats] RTCQualityLimitationReason and friends
- Re: [webrtc-stats] RTCQualityLimitationReason and friends
Re: [webrtc-stats] Split RTCMediaStreamTrackStats into four dictionaries.
- Re: [webrtc-stats] Split RTCMediaStreamTrackStats into four dictionaries.
- Re: [webrtc-stats] Split RTCMediaStreamTrackStats into four dictionaries.
- Re: [webrtc-stats] Split RTCMediaStreamTrackStats into four dictionaries.
Re: [webrtc-stats] Pivot from "track" to "sender" and "receiver" stats.
Re: [webrtc-pc] RTCSctpTransport: Specify special cases for maxMessageSize
Re: [webrtc-pc] offerToReceive* should ignore stopped transceivers, not unstopped ones.
[webrtc-pc] Pull Request: Fix offerToReceive* bug introduced by PR #1672
Re: [webrtc-pc] addTransceiver woes
- Re: [webrtc-pc] addTransceiver woes
- Re: [webrtc-pc] addTransceiver woes
- Re: [webrtc-pc] addTransceiver woes
- Re: [webrtc-pc] addTransceiver woes
- Re: [webrtc-pc] addTransceiver woes
- Re: [webrtc-pc] addTransceiver woes
- Re: [webrtc-pc] addTransceiver woes
- Re: [webrtc-pc] addTransceiver woes
- Re: [webrtc-pc] addTransceiver woes
- Re: [webrtc-pc] addTransceiver woes
- Re: [webrtc-pc] addTransceiver woes
- Re: [webrtc-pc] addTransceiver woes
- Re: [webrtc-pc] addTransceiver woes
- Re: [webrtc-pc] addTransceiver woes
[webrtc-pc] Pull Request: remove idl comment about rtcpTransport being at risk
- Re: [webrtc-pc] remove idl comment about rtcpTransport being at risk
- Re: [webrtc-pc] remove idl comment about rtcpTransport being at risk
Re: [webrtc-pc] Add testing guideline for naming test files and adding comments
- Re: [webrtc-pc] Add testing guideline for naming test files and adding comments
- Re: [webrtc-pc] Add testing guideline for naming test files and adding comments
[webrtc-pc] replaceTrack and removeTrack
- Re: [webrtc-pc] replaceTrack and removeTrack
- Re: [webrtc-pc] replaceTrack and removeTrack: Synchronous?
- Re: [webrtc-pc] replaceTrack and removeTrack: Synchronous?
- Re: [webrtc-pc] replaceTrack and removeTrack: Synchronous?
- Re: [webrtc-pc] replaceTrack and removeTrack: Synchronous?
- Re: [webrtc-pc] replaceTrack and removeTrack: Synchronous?
- Re: [webrtc-pc] replaceTrack and removeTrack: Synchronous?
- Re: [webrtc-pc] replaceTrack and removeTrack: Synchronous?
- Re: [webrtc-pc] replaceTrack and removeTrack: Synchronous?
- Re: [webrtc-pc] replaceTrack and removeTrack: Synchronous?
- Re: [webrtc-pc] replaceTrack and removeTrack: Synchronous?
- Re: [webrtc-pc] replaceTrack and removeTrack: Synchronous?
- Re: [webrtc-pc] replaceTrack and removeTrack: Synchronous?