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