[webrtc-pc] offerToReceive* should ignore stopped transceivers, not unstopped ones.
[webrtc-pc] new commits pushed by alvestrand
[webrtc-pc] new commits pushed by aboba
[webrtc-pc] RTCPriorityType text is outdated
Closed: [webrtc-pc] Meta: auto-publish changes to the spec
[webrtc-pc] Pull Request: Set direction of transceiver created by offerToReceive* option
Re: [webrtc-pc] OAUTH-POP-KEY-DISTRIBUTION IETF draft has been replaced by ACE-CWT-PROOF-OF-POSSESSION
Re: [webrtc-pc] Ordering of stream "addtrack"/"removetrack" events vs. "track" event
[webrtc-pc] Pull Request: Editorial: Add IANA-HASH-FUNCTION reference
[webrtc-pc] new commits pushed by alvestrand
Closed: [webrtc-pc] Spec has reference to WEBIDL-1 and WEBIDL
Re: [webrtc-pc] Let javascript set different priorities for bitrate and DSCP markings.
[webrtc-pc] RTCPeerConnection constructor can fail - what error to return?
- Re: [webrtc-pc] RTCPeerConnection constructor can fail - what error to return?
- Re: [webrtc-pc] RTCPeerConnection constructor can fail - what error to return?
- Re: [webrtc-pc] RTCPeerConnection constructor can fail - what error to return?
- Re: [webrtc-pc] RTCPeerConnection constructor can fail - what error to return?
- Re: [webrtc-pc] RTCPeerConnection constructor can fail - what error to return?
Re: [webrtc-stats] Add stat to reflect the redundancy of FEC/RED data
Re: [webrtc-stats] Do the "audio level" stats include MediaStreamTrack volume settings?
Re: [webrtc-stats] Add estimatedClockSkew
Re: [webrtc-stats] RTCQualityLimitationReason and friends
Re: [webrtc-stats] RTCMediaStreamTrackStats is four dictionaries in one
- Re: [webrtc-stats] RTCMediaStreamTrackStats is four dictionaries in one
- Re: [webrtc-stats] RTCMediaStreamTrackStats is four dictionaries in one
Re: [webrtc-stats] Need DSCP information for incoming RTP streams
Re: [webrtc-stats] Need DSCP information for outgoing RTP streams
Re: [webrtc-stats] Privacy & Security self review
Closed: [webrtc-stats] Privacy & Security self review
[webrtc-stats] new commits pushed by vr000m
[webrtc-stats] new commits pushed by henbos
Closed: [webrtc-stats] Discuss caching and consistency of getStats() return
Re: [webrtc-stats] Complete security and privacy considerations based on self-review
Re: [webrtc-stats] Added Guidelines for getStats() results caching
Re: [webrtc-stats] RTCMediaStreamTrackStats.concealedAudibleSamples
- Re: [webrtc-stats] RTCMediaStreamTrackStats.concealedAudibleSamples
- Re: [webrtc-stats] RTCMediaStreamTrackStats.concealedAudibleSamples
Re: [webrtc-stats] RTCMediaStreamTrackStats.concealedAudibleSamples added.
[webrtc-pc] order of transceivers, senders/receivers?
- Re: [webrtc-pc] order of transceivers, senders/receivers?
- Re: [webrtc-pc] order of transceivers, senders/receivers?
- Re: [webrtc-pc] order of transceivers, senders/receivers?
- Re: [webrtc-pc] order of transceivers, senders/receivers?
- Re: [webrtc-pc] order of transceivers, senders/receivers?
- Re: [webrtc-pc] order of transceivers, senders/receivers?
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: RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries
- Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries
- Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries
- Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries
- Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries
- Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries
- Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries
- Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries
- Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries
- Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries
- Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries
Closed: [webrtc-pc] audioLevel of tracks, both send and receive?
Re: [webrtc-pc] audioLevel of tracks, both send and receive?
Re: [webrtc-pc] Add a new value to RTCIceTransportPolicy.
[webrtc-stats] Add per layer stats for SVC
[webrtc-pc] Pull Request: Set muted before SRD resolves, using new set muted algorithm.
- Re: [webrtc-pc] Set muted before SRD resolves, using new set muted algorithm.
- Re: [webrtc-pc] Set muted before SRD resolves, using new set muted algorithm.
- Re: [webrtc-pc] Set muted before SRD resolves, using new set muted algorithm.
- Re: [webrtc-pc] Set muted before SRD resolves, using new set muted algorithm.
[webrtc-pc] Pull Request: fix RTCRtpSynchronizationSource.audioLevel idl to be nullable
- Re: [webrtc-pc] fix RTCRtpSynchronizationSource.audioLevel idl to be nullable
- Re: [webrtc-pc] fix RTCRtpSynchronizationSource.audioLevel idl to be nullable
- Re: [webrtc-pc] fix RTCRtpSynchronizationSource.audioLevel idl to be nullable
[webrtc-pc] RTCRtpSynchronizationSource.audioLevel in idl should be nullable
- Re: [webrtc-pc] RTCRtpSynchronizationSource.audioLevel in idl should be nullable
- Re: [webrtc-pc] RTCRtpSynchronizationSource.audioLevel in idl should be nullable
- Closed: [webrtc-pc] RTCRtpSynchronizationSource.audioLevel in idl should be nullable
- Re: [webrtc-pc] RTCRtpSynchronizationSource.audioLevel in idl should be nullable
- Re: [webrtc-pc] RTCRtpSynchronizationSource.audioLevel in idl should be nullable
Re: [webrtc-stats] Added 'objectDeleted' attribute
[webrtc-stats] new commits pushed by vr000m
Closed: [webrtc-stats] Add stat for network type of ICE candidate
[webrtc-pc] Pull Request: 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
- 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-stats] Pull Request: Rebase of Taylor's NetworkType PR
- Re: [webrtc-stats] Rebase of Taylor's NetworkType PR
- Re: [webrtc-stats] Rebase of Taylor's NetworkType PR
- Re: [webrtc-stats] Rebase of Taylor's NetworkType PR
[webrtc-pc] Point to the Hash Function Textual Names registry
- Re: [webrtc-pc] Point to the Hash Function Textual Names registry
- Re: [webrtc-pc] Point to the Hash Function Textual Names registry
- Re: [webrtc-pc] Point to the Hash Function Textual Names registry
- Re: [webrtc-pc] Point to the Hash Function Textual Names registry
- Re: [webrtc-pc] Point to the Hash Function Textual Names registry
- Closed: [webrtc-pc] Point to the Hash Function Textual Names registry
[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] new commits pushed by burnburn
[webrtc-pc] new commits pushed by burnburn
Closed: [webrtc-pc] Adopt "test as you commit" policy
[webrtc-pc] new commits pushed by adam-be
Re: [webrtc-pc] Clarify whether RTCRtpContributingSource members are live.
[webrtc-pc] Pull Request: fix ref to webidl
Re: [webrtc-pc] canInsertDTMF transitions not specified
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?
Re: [webrtc-pc] Need for Initial Bitrate by the Application/RtpSender?
Re: [webrtc-pc] JSEP references are out dated
Re: [webrtc-pc] Document "test as you commit" policy in CONTRIBUTING.md
- Re: [webrtc-pc] Document "test as you commit" policy in CONTRIBUTING.md
- Re: [webrtc-pc] Document "test as you commit" policy in CONTRIBUTING.md
- Re: [webrtc-pc] Document "test as you commit" policy in CONTRIBUTING.md
Re: [webrtc-pc] Isolated Media Streams requires modification on permission algorithms in GUM and Permissions specs
- Re: [webrtc-pc] Isolated Media Streams requires modification on permission algorithms in GUM and Permissions specs
- Re: [webrtc-pc] Isolated Media Streams requires modification on permission algorithms in GUM and Permissions specs
[webrtc-pc] relay-first as an option for RTCIceTransportPolicy
- Re: [webrtc-pc] relay-first as an option for RTCIceTransportPolicy
- Re: [webrtc-pc] relay-first as an option for RTCIceTransportPolicy
- Closed: [webrtc-pc] relay-first as an option for RTCIceTransportPolicy
Re: [webrtc-pc] Adopt "test as you commit" policy
[webrtc-pc] Pull Request: RTCSctpTransport: Specify special cases for maxMessageSize
- Re: [webrtc-pc] RTCSctpTransport: Specify special cases for maxMessageSize
- Re: [webrtc-pc] RTCSctpTransport: Specify special cases for maxMessageSize
- Re: [webrtc-pc] RTCSctpTransport: Specify special cases for maxMessageSize
- Re: [webrtc-pc] RTCSctpTransport: Specify special cases for maxMessageSize
- Re: [webrtc-pc] RTCSctpTransport: Specify special cases for maxMessageSize
Re: [webrtc-pc] Stats & isolated streams
[webrtc-stats] Pull Request: Pivot from "track" to "sender" and "receiver" stats.
- Re: [webrtc-stats] Pivot from "track" to "sender" and "receiver" stats.
- Re: [webrtc-stats] Pivot from "track" to "sender" and "receiver" stats.
[webrtc-stats] Pull Request: Split RTCMediaStreamTrackStats into four dictionaries.
Re: [webrtc-pc] Meta: auto-publish changes to the spec
- Re: [webrtc-pc] Meta: auto-publish changes to the spec
- Re: [webrtc-pc] Meta: auto-publish changes to the spec
Re: [webrtc-stats] Adding "networkType" field to RTCIceCandidateStats.
- Re: [webrtc-stats] Adding "networkType" field to RTCIceCandidateStats.
- Re: [webrtc-stats] Adding "networkType" field to RTCIceCandidateStats.
Re: [webrtc-pc] Adding more values to RTCIceTransportPolicy Enum
- Re: [webrtc-pc] Adding more values to RTCIceTransportPolicy Enum
- Re: [webrtc-pc] Adding more values to RTCIceTransportPolicy Enum
- Re: [webrtc-pc] Adding more values to RTCIceTransportPolicy Enum
[webrtc-stats] Add stat for inputAudioLevel, before the audio filter
- Re: [webrtc-stats] Add stat for inputAudioLevel, before the audio filter
- Re: [webrtc-stats] Add stat for inputAudioLevel, before the audio filter
Re: [webrtc-stats] We need "sender" and "receiver" stats, not "track" stats
Re: [webrtc-stats] jitterBufferDelay vs playoutDelay
Closed: [webrtc-stats] jitterBufferDelay vs playoutDelay
[webrtc-stats] new commits pushed by vr000m
Closed: [webrtc-pc] setIdentityProvider should throw error if protocol contains invalid characters
[webrtc-pc] new commits pushed by aboba
[webrtc-pc] new commits pushed by aboba
[webrtc-pc] new commits pushed by alvestrand
Closed: [webrtc-pc] Sending data channel messages > maxMessageSize
[webrtc-pc] new commits pushed by alvestrand
Closed: [webrtc-pc] Why is setDirection a method?
[webrtc-pc] new commits pushed by alvestrand
Re: [webrtc-pc] Section 11: Examples
Closed: [webrtc-pc] Section 6.4: Datachannel Garbage Collection
[webrtc-pc] Pull Request: Use [[SctpTransport]] slot in RTCPeerConnection.close
[webrtc-pc] Pull Request: Add steps to create an RTCSctpTransport
Re: [webrtc-pc] Adding relativeBitrate parameter to RTCRtpEncodingParameters.
- Re: [webrtc-pc] Adding relativeBitrate parameter to RTCRtpEncodingParameters.
- Re: [webrtc-pc] Adding relativeBitrate parameter to RTCRtpEncodingParameters.
Re: [webrtc-pc] Prepare status of the document for CR publication
- Re: [webrtc-pc] Prepare status of the document for CR publication
- Re: [webrtc-pc] Prepare status of the document for CR publication
Re: [webrtc-pc] Validate protocol string in IdP operations
[webrtc-pc] [[SctpTransport]] slot needs to be defined and initialized
[webrtc-pc] RTCPeerConnection.close should only refer to one RTCSctpTransport
[webrtc-pc] new commits pushed by adam-be
Re: [webrtc-pc] Section 6.4: Datachannel Garbage Collection
Re: [webrtc-stats] What is fractionLost for a local incoming media stream?
Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case
- Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case
- Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case
- Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case
- Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case
- Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case
- Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case
- Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case