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

[webrtc-pc] "set an RTCSessionDescription" steps need to be broken up into multiple smaller lists of steps

[webrtc-pc] Adding more values to RTCIceTransportPolicy Enum

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

[webrtc-pc] Ambiguities with BUNDLE and ICE

[webrtc-pc] canInsertDTMF values not specified

[webrtc-pc] Clarify offerToReceiveAudio and offerToReceiveVideo in renegotiation

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

[webrtc-pc] createDataChannel should use "in parallel" (editorial)

[webrtc-pc] Describe how transport objects are assigned to senders/receivers.

[webrtc-pc] Example ICE state transition appears incorrect

[webrtc-pc] Explain the scope of DTLS/ICE transport objects in more detail.

[webrtc-pc] format & fixup Example code

[webrtc-pc] getIdentityAssertion should reject with InvalidAccessError if no identity provider is set

[webrtc-pc] Instantiating an IdP Proxy should be defined in terms of Fetch

[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 setLocalDescription use [[LastAnswer/Offer]] internal slots

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

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

[webrtc-pc] Need for Initial Bitrate?

[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 vivienlacourba

[webrtc-pc] No way to tell when SCTP association is closed

[webrtc-pc] OAUTH-POP-KEY-DISTRIBUTION IETF draft has been replaced by ACE-CWT-PROOF-OF-POSSESSION

[webrtc-pc] OAUTH-POP-KEY-DISTRIBUTION IETF draft is has been replaced by ACE-CWT-PROOF-OF-POSSESSION

[webrtc-pc] offerToReceive* legacy behaviour spec does not match the behaviour of legacy implementations

[webrtc-pc] offerToReceiveAudio/Video processing creates sendrecv transceivers

[webrtc-pc] Please do not use datatracker.ietf.org

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

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

[webrtc-pc] Pull Request: Add onstatechange event to SctpTransport Event table

[webrtc-pc] Pull Request: Describe how transport objects are assigned to senders/receivers.

[webrtc-pc] Pull Request: Explain the scope of DTLS/ICE transport objects in more detail.

[webrtc-pc] Pull Request: format & fixup Example code

[webrtc-pc] Pull Request: Let setLocalDescription use [[LastAnswer/Offer]] internal slots

[webrtc-pc] Pull Request: Note about video dimension adaptation discussion added.

[webrtc-pc] Pull Request: pc.close() should close Data Channels and SctpTransports

[webrtc-pc] Pull Request: Replace setDirection() with writable direction attribute

[webrtc-pc] Pull Request: Set up automatic PR review request on webrtc.html

[webrtc-pc] Pull Request: setParameters: Use more specific hardware encoder errors

[webrtc-pc] Pull Request: Simple text for scaling issue - no letterbox allowed.

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

[webrtc-pc] Pull Request: upgrade to Node 8 (respec 1.7 now uses async/await)

[webrtc-pc] Replace setDirection() with writable direction attribute

[webrtc-pc] Resizing video (fix for Issue 1283)

[webrtc-pc] RTCDataChannel's send() doesn't encode string data

[webrtc-pc] RTCIdentityProviderGlobalScope looks wrong

[webrtc-pc] RTCPriorityType combines relative bitrate with QoS priority, which applications may not want.

[webrtc-pc] RTCSctpTransport.maxMessageSize 0 case

[webrtc-pc] Section 5.2: centering, scaling, cropping

[webrtc-pc] Section 5.3: track.label initialization

[webrtc-pc] Simple text for scaling issue - no letterbox allowed.

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

[webrtc-pc] Stats & isolated streams

[webrtc-pc] Throw error if data channel's buffer is filled, rather than closing.

[webrtc-pc] toJSON Dfn needed

[webrtc-pc] Validate protocol string in IdP operations

[webrtc-pc] Why is setDirection a method?

[webrtc-stats] "Make tidy" is broken?

[webrtc-stats] Add "packetsFailedDecryption", to count SRTP decryption failures.

[webrtc-stats] Add "privacy" to security considerations

[webrtc-stats] Add estimatedClockSkew

[webrtc-stats] add packetsDuplicated as a result of #253

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

[webrtc-stats] Add stat for SRTP decryption failures

[webrtc-stats] Add stats for the negotiated DTLS-SRTP and DTLS cipher suites.

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

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

[webrtc-stats] Audio/Video sync follow-up

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

[webrtc-stats] Consider the Reporting API as a stats export mechanism

[webrtc-stats] Definitions from MSE need re-targeting

[webrtc-stats] Difference between 'Id' and 'Identifier' fields

[webrtc-stats] Interframe delay stat for video receive stream.

[webrtc-stats] Is bytesReceived really available for RTCRemoteInboundRTPStreamStats?

[webrtc-stats] jitterBufferDelay and concealed samples, DTX/CNG samples

[webrtc-stats] jitterBufferOutput added

[webrtc-stats] keyFramesSent and keyFramesReceived added

[webrtc-stats] Move to continuous publication

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

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

[webrtc-stats] new commits pushed by alvestrand

[webrtc-stats] new commits pushed by henbos

[webrtc-stats] new commits pushed by vivienlacourba

[webrtc-stats] new commits pushed by vr000m

[webrtc-stats] packetsLost is unsigned but can be negative according to RFC

[webrtc-stats] Privacy & Security self review

[webrtc-stats] Pull Request: Add "packetsFailedDecryption", to count SRTP decryption failures.

[webrtc-stats] Pull Request: Add a paragraph about "identifier" when "id" is occupied

[webrtc-stats] Pull Request: Added 'objectDeleted' attribute

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

[webrtc-stats] Pull Request: Fix respec issue: Multiple s for and

[webrtc-stats] Pull Request: Issue 257 network type

[webrtc-stats] Pull Request: jitterBufferDelay in seconds

[webrtc-stats] Pull Request: jitterBufferOutput added

[webrtc-stats] Pull Request: RTCQualityLimitationReason and friends

[webrtc-stats] Pull Request: signed packetsLost

[webrtc-stats] Pull Request: TAG review: Link to design principles

[webrtc-stats] Pull Request: upgrade to Node 8 (respec now uses async/await)

[webrtc-stats] RTCMediaStreamTrackStats is four dictionaries in one

[webrtc-stats] RTCMediaStreamTrackStats.audioLevel clarification

[webrtc-stats] RTCMediaStreamTrackStats.concealedAudibleSamples added.

[webrtc-stats] RTCQualityLimitationReason and friends

[webrtc-stats] signed packetsLost

[webrtc-stats] Stat for adaptation reason

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

[webrtc-stats] Stats for adaptation reason, for realsies

[webrtc-stats] Stats for Audio network adaptation

[webrtc-stats] TAG review: Link to design principles

[webrtc-stats] upgrade to Node 8 (respec now uses async/await)

[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] Clarify offerToReceiveAudio and offerToReceiveVideo in renegotiation

Closed: [webrtc-pc] createDataChannel should use "in parallel" (editorial)

Closed: [webrtc-pc] createDataChannel touches DOM "in the background"

Closed: [webrtc-pc] duration and interToneGap should be stored as internal slots of RTCDTMFSender

Closed: [webrtc-pc] How to find what keyGenAlgorithm values the browser supports?

Closed: [webrtc-pc] Missing detail on obtaining lastOffer/lastAnswer in setLocalDescription

Closed: [webrtc-pc] Need to describe when ICE and DTLS transport objects are created/changed

Closed: [webrtc-pc] No way to tell when SCTP association is closed

Closed: [webrtc-pc] pc.close() method should close sctp transport

Closed: [webrtc-pc] Please do not use datatracker.ietf.org

Closed: [webrtc-pc] RTCRtpSender.setParameters: Do we need more specific hardware encoder errors?

Closed: [webrtc-pc] Section 5.3: track.label initialization

Closed: [webrtc-pc] toJSON Dfn needed

Closed: [webrtc-pc] When ICE restarts results in connection to a new endpoint

Closed: [webrtc-stats] "Make tidy" is broken?

Closed: [webrtc-stats] Add "privacy" to security considerations

Closed: [webrtc-stats] Add stat for SRTP decryption failures

Closed: [webrtc-stats] Difference between 'Id' and 'Identifier' fields

Closed: [webrtc-stats] Move to continuous publication

Closed: [webrtc-stats] packetsLost is unsigned but can be negative according to RFC

Closed: [webrtc-stats] RTCMediaStreamTrackStats.jitterBufferDelay: Specify unit in seconds

Closed: [webrtc-stats] RTCMediaStreamTrackStats.keyFramesReceived

Closed: [webrtc-stats] Stat for adaptation reason

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

Last message date: Tuesday, 31 October 2017 21:48:06 UTC