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

[webrtc-pc] [[SctpTransport]] slot needs to be defined and initialized

[webrtc-pc] Add a new value to RTCIceTransportPolicy.

[webrtc-pc] Add testing guideline for naming test files and adding comments

[webrtc-pc] Adding more values to RTCIceTransportPolicy Enum

[webrtc-pc] Adding relativeBitrate parameter to RTCRtpEncodingParameters.

[webrtc-pc] addTransceiver woes

[webrtc-pc] Adopt "test as you commit" policy

[webrtc-pc] audioLevel of tracks, both send and receive?

[webrtc-pc] behaviour of offerToReceive* set to false when there is a local track

[webrtc-pc] canInsertDTMF transitions not specified

[webrtc-pc] Clarify whether RTCRtpContributingSource members are live.

[webrtc-pc] Document "test as you commit" policy in CONTRIBUTING.md

[webrtc-pc] Editorial: Add IANA-HASH-FUNCTION reference

[webrtc-pc] fix RTCRtpSynchronizationSource.audioLevel idl to be nullable

[webrtc-pc] Inconsistencies in Asynchronous Task Queueing

[webrtc-pc] Isolated Media Streams requires modification on permission algorithms in GUM and Permissions specs

[webrtc-pc] JSEP references are out dated

[webrtc-pc] Let javascript set different priorities for bitrate and DSCP markings.

[webrtc-pc] Meta: auto-publish changes to the spec

[webrtc-pc] Need for Initial Bitrate by the Application/RtpSender?

[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[webrtc-pc] OAUTH-POP-KEY-DISTRIBUTION IETF draft has been replaced by ACE-CWT-PROOF-OF-POSSESSION

[webrtc-pc] offerToReceive* should ignore stopped transceivers, not unstopped ones.

[webrtc-pc] order of transceivers, senders/receivers?

[webrtc-pc] Ordering of stream "addtrack"/"removetrack" events vs. "track" event

[webrtc-pc] Point to the Hash Function Textual Names registry

[webrtc-pc] Prepare status of the document for CR publication

[webrtc-pc] Pull Request: Add steps to create an RTCSctpTransport

[webrtc-pc] Pull Request: Add testing guideline for naming test files and adding comments

[webrtc-pc] Pull Request: Editorial: Add IANA-HASH-FUNCTION reference

[webrtc-pc] Pull Request: fix ref to webidl

[webrtc-pc] Pull Request: fix RTCRtpSynchronizationSource.audioLevel idl to be nullable

[webrtc-pc] Pull Request: RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries

[webrtc-pc] Pull Request: RTCSctpTransport: Specify special cases for maxMessageSize

[webrtc-pc] Pull Request: Set direction of transceiver created by offerToReceive* option

[webrtc-pc] Pull Request: Set muted before SRD resolves, using new set muted algorithm.

[webrtc-pc] Pull Request: Use [[SctpTransport]] slot in RTCPeerConnection.close

[webrtc-pc] relay-first as an option for RTCIceTransportPolicy

[webrtc-pc] RTCPeerConnection constructor can fail - what error to return?

[webrtc-pc] RTCPeerConnection.close should only refer to one RTCSctpTransport

[webrtc-pc] RTCPriorityType text is outdated

[webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries

[webrtc-pc] RTCRtpSynchronizationSource.audioLevel in idl should be nullable

[webrtc-pc] RTCSctpTransport.maxMessageSize 0 case

[webrtc-pc] RTCSctpTransport: Specify special cases for maxMessageSize

[webrtc-pc] Section 11: Examples

[webrtc-pc] Section 6.4: Datachannel Garbage Collection

[webrtc-pc] Set direction of transceiver created by offerToReceive* option

[webrtc-pc] Set muted before SRD resolves, using new set muted algorithm.

[webrtc-pc] Specify how RTCSctpTransport.maxMessageSize gets its value

[webrtc-pc] Stats & isolated streams

[webrtc-pc] Validate protocol string in IdP operations

[webrtc-stats] Add estimatedClockSkew

[webrtc-stats] Add per layer stats for SVC

[webrtc-stats] Add stat for inputAudioLevel, before the audio filter

[webrtc-stats] Add stat to reflect the redundancy of FEC/RED data

[webrtc-stats] Added 'objectDeleted' attribute

[webrtc-stats] Added Guidelines for getStats() results caching

[webrtc-stats] Adding "networkType" field to RTCIceCandidateStats.

[webrtc-stats] Complete security and privacy considerations based on self-review

[webrtc-stats] Do the "audio level" stats include MediaStreamTrack volume settings?

[webrtc-stats] Is keeping stats around a memory problem?

[webrtc-stats] jitterBufferDelay vs playoutDelay

[webrtc-stats] Need DSCP information for incoming RTP streams

[webrtc-stats] Need DSCP information for outgoing RTP streams

[webrtc-stats] new commits pushed by henbos

[webrtc-stats] new commits pushed by vr000m

[webrtc-stats] Pivot from "track" to "sender" and "receiver" stats.

[webrtc-stats] Privacy & Security self review

[webrtc-stats] Pull Request: Pivot from "track" to "sender" and "receiver" stats.

[webrtc-stats] Pull Request: Rebase of Taylor's NetworkType PR

[webrtc-stats] Pull Request: Split RTCMediaStreamTrackStats into four dictionaries.

[webrtc-stats] Rebase of Taylor's NetworkType PR

[webrtc-stats] RTCMediaStreamTrackStats is four dictionaries in one

[webrtc-stats] RTCMediaStreamTrackStats.concealedAudibleSamples

[webrtc-stats] RTCMediaStreamTrackStats.concealedAudibleSamples added.

[webrtc-stats] RTCQualityLimitationReason and friends

[webrtc-stats] Split RTCMediaStreamTrackStats into four dictionaries.

[webrtc-stats] We need "sender" and "receiver" stats, not "track" stats

[webrtc-stats] What is fractionLost for a local incoming media stream?

Closed: [webrtc-pc] [[SctpTransportState]] slot needs to be defined and initialized

Closed: [webrtc-pc] Adopt "test as you commit" policy

Closed: [webrtc-pc] audioLevel of tracks, both send and receive?

Closed: [webrtc-pc] Meta: auto-publish changes to the spec

Closed: [webrtc-pc] Point to the Hash Function Textual Names registry

Closed: [webrtc-pc] relay-first as an option for RTCIceTransportPolicy

Closed: [webrtc-pc] RTCPeerConnection.close should only refer to the [[SctpTransport]] slot

Closed: [webrtc-pc] RTCRtpSynchronizationSource.audioLevel in idl should be nullable

Closed: [webrtc-pc] Section 6.4: Datachannel Garbage Collection

Closed: [webrtc-pc] Sending data channel messages > maxMessageSize

Closed: [webrtc-pc] setIdentityProvider should throw error if protocol contains invalid characters

Closed: [webrtc-pc] Spec has reference to WEBIDL-1 and WEBIDL

Closed: [webrtc-pc] Why is setDirection a method?

Closed: [webrtc-stats] Add stat for network type of ICE candidate

Closed: [webrtc-stats] Discuss caching and consistency of getStats() return

Closed: [webrtc-stats] jitterBufferDelay vs playoutDelay

Closed: [webrtc-stats] Privacy & Security self review

Last message date: Thursday, 30 November 2017 21:34:52 UTC