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

[webrtc-pc] "Hybrid" OAuth solution.

[webrtc-pc] Add an explicit stats selection algorithm.

[webrtc-pc] Add clockrate, channels, sdpFmtpLine to codec capability

[webrtc-pc] Add stats selection algorithm based on sender or receiver of selector.

[webrtc-pc] Add Taylor as editor of the spec

[webrtc-pc] Adding "[EnforceRange]" to RTCDataChannelInit.id.

[webrtc-pc] Advanced Peer-to-peer Example

[webrtc-pc] Align getAlgorithm return value with Web Crypto

[webrtc-pc] Always update the RTCRtpContributingSource for SSRCs.

[webrtc-pc] Attaching the same track to multiple RTCRtpSenders: spec contradiction.

[webrtc-pc] Attempt to update RTCRtpContributingSource objects at playout time.

[webrtc-pc] Can candidate.ufrag be null?

[webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible

[webrtc-pc] Changing iceCandidatePoolSize to an octet and adding EnforceRange.

[webrtc-pc] Check RTCPeerConnection isClosed slot before running queued tasks

[webrtc-pc] Clarify reasoning behind and mitigation of privacy issues (PING review)

[webrtc-pc] Clarify when RTCRtpContributingSource.audioLevel can be null.

[webrtc-pc] Clarify which timestamp RTCStats.timestamp represents.

[webrtc-pc] Clarifying exactly what "sdpFmtpLine" represents.

[webrtc-pc] Description of handling iceCandidatePoolSize in setConfiguration is out of sync with JSEP

[webrtc-pc] Diagram for RTCSignalingState includes "closed" state, which doesn't exist?

[webrtc-pc] Don't link function names to legacy section (unless intended)

[webrtc-pc] DTLS failures

[webrtc-pc] editorial: Some links to setLocalDescription points to the legacy extensions

[webrtc-pc] Freeing resources for incoming stream

[webrtc-pc] Generation of candidates

[webrtc-pc] getStats on sender and receiver.

[webrtc-pc] Handing SDP with more than one identity

[webrtc-pc] Handling RTX in RTCRtpCodecCapabilities

[webrtc-pc] Label 'Warm-up example' as 'advanced p2p example'

[webrtc-pc] Make "candidate" non-nullable in addIceCandidate parameter table.

[webrtc-pc] Make legacy API optional to implement

[webrtc-pc] Make RTCDataChannel.id nullable and describe when it's set.

[webrtc-pc] Mark getAlgorithm method at risk

[webrtc-pc] Mark Identity as feature at risk?

[webrtc-pc] Mention that codecs can be reordered or removed but not modified.

[webrtc-pc] Need to specify what happens if the browser doesn't implement "negotiate" rtcpMuxPolicy

[webrtc-pc] Need to specify which members of the encodings in "sendEncodings" are actually used

[webrtc-pc] new commits pushed by aboba

[webrtc-pc] new commits pushed by adam-be

[webrtc-pc] new commits pushed by alvestrand

[webrtc-pc] new commits pushed by burnburn

[webrtc-pc] new commits pushed by dontcallmedom

[webrtc-pc] new commits pushed by taylor-b

[webrtc-pc] Note At-Risk features at front of document

[webrtc-pc] Pull Request: Add clockrate, channels, sdpFmtpLine to codec capability

[webrtc-pc] Pull Request: Add getStats() to sender and receiver.

[webrtc-pc] Pull Request: Add missing "closed" signaling state.

[webrtc-pc] Pull Request: Add Taylor as editor of the spec

[webrtc-pc] Pull Request: Adding "[EnforceRange]" to RTCDataChannelInit.id.

[webrtc-pc] Pull Request: Always update the RTCRtpContributingSource for SSRCs.

[webrtc-pc] Pull Request: Attempt to update RTCRtpContributingSource objects at playout time.

[webrtc-pc] Pull Request: Changing iceCandidatePoolSize to an octet and adding EnforceRange.

[webrtc-pc] Pull Request: Check RTCPeerConnection isClosed slot before running queued tasks

[webrtc-pc] Pull Request: Clarify when RTCRtpContributingSource.audioLevel can be null.

[webrtc-pc] Pull Request: Clarify which "candidate" is referred to in addIceCandidate description.

[webrtc-pc] Pull Request: Clarifying exactly what "sdpFmtpLine" represents.

[webrtc-pc] Pull Request: Don't link function names to legacy section (unless intended)

[webrtc-pc] Pull Request: Freeing resources for incoming stream

[webrtc-pc] Pull Request: Generation of candidates

[webrtc-pc] Pull Request: Handling RTX in RTCRtpCodecCapabilities

[webrtc-pc] Pull Request: Label 'Warm-up example' as 'advanced p2p example'

[webrtc-pc] Pull Request: Make "candidate" non-nullable in addIceCandidate parameter table.

[webrtc-pc] Pull Request: Mark getAlgorithm method at risk

[webrtc-pc] Pull Request: mute signal

[webrtc-pc] Pull Request: Remove connecting event from Event summary

[webrtc-pc] Pull Request: Remove last use of 'set of receivers'

[webrtc-pc] Pull Request: Remove Section 11.6 note

[webrtc-pc] Pull Request: Remove Section 12.2.1.1 errorDetailEnum

[webrtc-pc] Pull Request: RTP/RTCP non-mux: feature at risk

[webrtc-pc] Pull Request: RtpSender and track

[webrtc-pc] Pull Request: Specify behavior if browser doesn't implement "negotiate" rtcpMuxPolicy.

[webrtc-pc] Pull Request: Switching to new, consistent terminology when talking about exceptions.

[webrtc-pc] Pull Request: Throw InvalidModificationError if changing pool size after SLD.

[webrtc-pc] Pull Request: Update Call Flow Browser to Browser

[webrtc-pc] Pull Request: Update call flow in Section 11.6

[webrtc-pc] Range checking needed for iceCandidatePoolSize

[webrtc-pc] Remove Section 11.6 note

[webrtc-pc] Remove Section 12.2.1.1 errorDetailEnum

[webrtc-pc] RTCDataChannelInit.id should use [EnforceRange]

[webrtc-pc] RTCRtpContributingSource naming

[webrtc-pc] RTCRtpContributingSource.audioLevel not guaranteed to be in sync with audio playout

[webrtc-pc] RTP/RTCP non-mux: feature at risk

[webrtc-pc] RtpSender and track

[webrtc-pc] sdpFmtpLine isn't very convenient

[webrtc-pc] Section 10.3:

[webrtc-pc] Section 10.3: Freeing resources for incoming stream

[webrtc-pc] Section 10.3: Issue 5

[webrtc-pc] Section 10.3: mute signal

[webrtc-pc] Section 11.6: Issue 6

[webrtc-pc] Section 12.2.1.1: enum errorDetail definition

[webrtc-pc] Section 13: Connecting event

[webrtc-pc] Section 5.6: Generation of candidates

[webrtc-pc] Sender/Receiver.rtcpTransport: feature at risk?

[webrtc-pc] setCodecPreferences() can't handle codecs supporting multiple clock rates/packetization-mode

[webrtc-pc] Specify an AllowUnverifiedMedia RTCConfiguration property

[webrtc-pc] Specify behavior if browser doesn't implement "negotiate" rtcpMuxPolicy.

[webrtc-pc] Specify how media is centered, cropped, and scaled. Fixes #305

[webrtc-pc] Specify when random mid generation happens

[webrtc-pc] strawman text to show how unverified media would work

[webrtc-pc] Switching to new, consistent terminology when talking about exceptions.

[webrtc-pc] Throw InvalidModificationError if changing pool size after SLD.

[webrtc-pc] Update Call Flow Browser to Browser

[webrtc-pc] Update for structured cloning changes in HTML

[webrtc-pc] When exactly is an SSRC RTCRtpContributingSource object updated?

[webrtc-pc] When should RTCRtpContributingSource#audioLevel be null?

[webrtc-stats] Add link from datachannel to transport

[webrtc-stats] Add RTCOutboundRTPStreamStats.totalEncodeTime

[webrtc-stats] Add section on obsoleted stats.

[webrtc-stats] Added RTCInboundRTPStreamStats sample counters.

[webrtc-stats] Adding remoteTimestamp to RTCRtpStreamStats.

[webrtc-stats] Bandwidth estimations again (Issue 97 redux)

[webrtc-stats] Change log, and make tidy

[webrtc-stats] Consider making (aggregate) stats more accessible

[webrtc-stats] example 8.2: calculating fraction lost vs fractionLost stat

[webrtc-stats] Make sure timing and responsibility for stat object creation is clear

[webrtc-stats] new commits pushed by alvestrand

[webrtc-stats] Packets or frames discarded on send?

[webrtc-stats] Pull Request: Add link from datachannel to transport

[webrtc-stats] Pull Request: Add RTCOutboundRTPStreamStats.totalEncodeTime

[webrtc-stats] Pull Request: Add section on obsoleted stats.

[webrtc-stats] Pull Request: Added RTCInboundRTPStreamStats sample counters.

[webrtc-stats] Pull Request: Adds definitions for RTCDataChannelStats members.

[webrtc-stats] Pull Request: Change log, and make tidy

[webrtc-stats] Pull Request: Issue 97 redux

[webrtc-stats] Pull Request: RTCMediaStreamTrackStats members to keep track of audio and video sync.

[webrtc-stats] Pull Request: rtt undefined when no RTCP RR

[webrtc-stats] Related to outgoing and incoming bitrate estimates on a candidate pair

[webrtc-stats] Remove availableIncomingBitrate?

[webrtc-stats] Remove separation of received consent and connectivity requests.

[webrtc-stats] Reuse of "inbound-rtp" and "outbound-rtp" for RTCP is confusing.

[webrtc-stats] RoundTripTime not defined when the underlying stream can't calculate it

[webrtc-stats] RTCIceCandidatePairStats.writable/readable: redundant?

[webrtc-stats] RTCMediaStreamTrackStats members to keep track of audio and video sync.

[webrtc-stats] RTCPeerConnection.getStats: What to do with 'selector' argument?

[webrtc-stats] rtt undefined when no RTCP RR

[webrtc-stats] Stat for adaptation reason

[webrtc-stats] Stat for audio playout delay

[webrtc-stats] Stat for how many adaptation changes occur for a video track

[webrtc-stats] Stat for how many audio stream packets are expanded when packets are lost (and lost and the user is speaking)

[webrtc-stats] Stat for how much time it takes to encode video

[webrtc-stats] Stat for likelihood of echo

[webrtc-stats] Stat for retransmitted bytes

[webrtc-stats] Stats report for RTCRtpContributingSource objects

[webrtc-stats] Unclear if the request encompasses consent checks or not

Closed: [webrtc-pc] Advanced Peer-to-peer Example

Closed: [webrtc-pc] Align getAlgorithm return value with Web Crypto

Closed: [webrtc-pc] Can candidate.ufrag be null?

Closed: [webrtc-pc] Candidate from onicecandidate event and addIceCandidate are incompatible

Closed: [webrtc-pc] Clarify reasoning behind and mitigation of privacy issues (PING review)

Closed: [webrtc-pc] Describe what happens when media changes

Closed: [webrtc-pc] Description of handling iceCandidatePoolSize in setConfiguration is out of sync with JSEP

Closed: [webrtc-pc] Don't fire events on a closed peer connection

Closed: [webrtc-pc] Integrate RTCRtpTransceiver into set local/remote steps

Closed: [webrtc-pc] Need to describe that codecs can be removed or reordered, but not modified

Closed: [webrtc-pc] Need to specify precisely when MID generation happens

Closed: [webrtc-pc] Range checking needed for iceCandidatePoolSize

Closed: [webrtc-pc] RTCDataChannelInit.id should use [EnforceRange]

Closed: [webrtc-pc] RTCStats timestamp source ambiguous

Closed: [webrtc-pc] sdpFmtpLine isn't very convenient

Closed: [webrtc-pc] Section 10.3:

Closed: [webrtc-pc] Specify when a data channel's ID is assigned, and what the `id` attribute returns when no ID is assigned.

Closed: [webrtc-pc] STUN/TURN OAuth token auth parameter passing

Closed: [webrtc-stats] "stats object" terminology is confusing

Closed: [webrtc-stats] Are the stats read-only snapshots or will they change kept around?

Closed: [webrtc-stats] Consider making (aggregate) stats more accessible

Closed: [webrtc-stats] Consider putting roundTripTime on RTCInboundStreamStats

Closed: [webrtc-stats] Expose protocol used to communicate with TURN server

Closed: [webrtc-stats] Make sure timing and responsibility for stat object creation is clear

Closed: [webrtc-stats] Need advice for handling obsolete stats

Closed: [webrtc-stats] Need descriptions for `label`, `protocol` and `state` members of `RTCDataChannelStats`

Closed: [webrtc-stats] Not clear how to differentiate between received connectivity checks and consent requests

Closed: [webrtc-stats] RoundTripTime not defined when the underlying stream can't calculate it

Closed: [webrtc-stats] RTCDataChannelStats should reference RTCTransportStats

Closed: [webrtc-stats] RTCPeerConnection.getStats: What to do with 'selector' argument?

Closed: [webrtc-stats] There is no ICE "cancelled" state

Closed: [webrtc-stats] Timestamp in the getStats

Closed: [webrtc-stats] Track stats: track or attachment?

Closed: [webrtc-stats] Unclear if the request encompasses consent checks or not

Last message date: Thursday, 30 March 2017 23:18:20 UTC