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

Thursday, 30 November 2017

• Re: [webrtc-pc] RTCPeerConnection constructor can fail - what error to return? Nils Ohlmeier via GitHub
• [webrtc-pc] offerToReceive* should ignore stopped transceivers, not unstopped ones. jan-ivar via GitHub
• Re: [webrtc-pc] addTransceiver woes jan-ivar via GitHub
• Re: [webrtc-pc] addTransceiver woes Philipp Hancke via GitHub
• Re: [webrtc-pc] Add testing guideline for naming test files and adding comments Philip Jägenstedt via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [webrtc-pc] Point to the Hash Function Textual Names registry Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• Re: [webrtc-pc] RTCSctpTransport: Specify special cases for maxMessageSize Lennart Grahl via GitHub
• Re: [webrtc-pc] RTCSctpTransport: Specify special cases for maxMessageSize Harald Alvestrand via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries Philip Jägenstedt via GitHub
• Re: [webrtc-pc] RTCPeerConnection constructor can fail - what error to return? Benjamin Gruenbaum via GitHub
• Re: [webrtc-pc] Adding relativeBitrate parameter to RTCRtpEncodingParameters. stefan hakansson via GitHub
• Re: [webrtc-pc] Add testing guideline for naming test files and adding comments stefan hakansson via GitHub
• Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case Adam Bergkvist via GitHub
• Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case Adam Bergkvist via GitHub
• Re: [webrtc-pc] RTCPeerConnection constructor can fail - what error to return? Nils Ohlmeier via GitHub

Wednesday, 29 November 2017

• Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case Lennart Grahl via GitHub
• Re: [webrtc-pc] addTransceiver woes stefan hakansson via GitHub
• Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case Adam Bergkvist via GitHub
• Re: [webrtc-pc] RTCSctpTransport: Specify special cases for maxMessageSize Adam Bergkvist via GitHub
• Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case Adam Bergkvist via GitHub

Tuesday, 28 November 2017

• Re: [webrtc-pc] order of transceivers, senders/receivers? Philipp Hancke via GitHub
• Re: [webrtc-pc] order of transceivers, senders/receivers? jan-ivar via GitHub
• Re: [webrtc-pc] order of transceivers, senders/receivers? jan-ivar via GitHub
• [webrtc-pc] RTCPriorityType text is outdated Harald Alvestrand via GitHub

Monday, 27 November 2017

• Re: [webrtc-pc] Set muted before SRD resolves, using new set muted algorithm. jan-ivar via GitHub
• Re: [webrtc-pc] Set direction of transceiver created by offerToReceive* option Adam Bergkvist via GitHub
• Re: [webrtc-pc] Meta: auto-publish changes to the spec Dominique Hazael-Massieux via GitHub
• Closed: [webrtc-pc] Meta: auto-publish changes to the spec Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries Harald Alvestrand via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries henbos via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries Philip Jägenstedt via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries henbos via GitHub
• [webrtc-pc] Pull Request: Set direction of transceiver created by offerToReceive* option Adam Bergkvist via GitHub
• Re: [webrtc-pc] OAUTH-POP-KEY-DISTRIBUTION IETF draft has been replaced by ACE-CWT-PROOF-OF-POSSESSION misi via GitHub

Sunday, 26 November 2017

• Re: [webrtc-pc] Ordering of stream "addtrack"/"removetrack" events vs. "track" event henbos via GitHub

Saturday, 25 November 2017

• Re: [webrtc-pc] RTCPeerConnection constructor can fail - what error to return? Iñaki Baz Castillo via GitHub
• Re: [webrtc-pc] Set muted before SRD resolves, using new set muted algorithm. jan-ivar via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries jan-ivar via GitHub

Friday, 24 November 2017

• Re: [webrtc-pc] Meta: auto-publish changes to the spec Harald Alvestrand via GitHub
• Re: [webrtc-pc] Editorial: Add IANA-HASH-FUNCTION reference Harald Alvestrand via GitHub
• Re: [webrtc-pc] Point to the Hash Function Textual Names registry Philip Jägenstedt via GitHub
• Re: [webrtc-pc] Point to the Hash Function Textual Names registry Adam Bergkvist via GitHub
• Re: [webrtc-pc] Point to the Hash Function Textual Names registry Philip Jägenstedt via GitHub
• Re: [webrtc-pc] Point to the Hash Function Textual Names registry Adam Bergkvist via GitHub
• [webrtc-pc] Pull Request: Editorial: Add IANA-HASH-FUNCTION reference Adam Bergkvist via GitHub
• Re: [webrtc-pc] Set muted before SRD resolves, using new set muted algorithm. Adam Bergkvist via GitHub
• Re: [webrtc-pc] addTransceiver woes Adam Bergkvist via GitHub

Thursday, 23 November 2017

• Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries Philip Jägenstedt via GitHub
• Re: [webrtc-stats] RTCMediaStreamTrackStats.concealedAudibleSamples henbos via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries henbos via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries Philip Jägenstedt via GitHub
• Closed: [webrtc-pc] relay-first as an option for RTCIceTransportPolicy Harald Alvestrand via GitHub
• Re: [webrtc-pc] relay-first as an option for RTCIceTransportPolicy Harald Alvestrand via GitHub
• Re: [webrtc-pc] order of transceivers, senders/receivers? Harald Alvestrand via GitHub
• Re: [webrtc-pc] RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries Harald Alvestrand via GitHub
• Re: [webrtc-pc] Set muted before SRD resolves, using new set muted algorithm. Harald Alvestrand via GitHub
• Re: [webrtc-pc] fix RTCRtpSynchronizationSource.audioLevel idl to be nullable Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [webrtc-pc] Spec has reference to WEBIDL-1 and WEBIDL Harald Alvestrand via GitHub
• Re: [webrtc-pc] Let javascript set different priorities for bitrate and DSCP markings. Harald Alvestrand via GitHub
• Re: [webrtc-pc] RTCSctpTransport: Specify special cases for maxMessageSize Harald Alvestrand via GitHub
• Re: [webrtc-pc] Add a new value to RTCIceTransportPolicy. Harald Alvestrand via GitHub
• Re: [webrtc-pc] Adding relativeBitrate parameter to RTCRtpEncodingParameters. Harald Alvestrand via GitHub
• Re: [webrtc-pc] RTCPeerConnection constructor can fail - what error to return? Dominique Hazael-Massieux via GitHub
• [webrtc-pc] RTCPeerConnection constructor can fail - what error to return? Harald Alvestrand via GitHub
• Re: [webrtc-stats] Is keeping stats around a memory problem? Harald Alvestrand via GitHub

Wednesday, 22 November 2017

• Re: [webrtc-pc] RTCRtpSynchronizationSource.audioLevel in idl should be nullable jan-ivar via GitHub
• Re: [webrtc-stats] Add stat to reflect the redundancy of FEC/RED data Robin Schriebman via GitHub
• Re: [webrtc-stats] Do the "audio level" stats include MediaStreamTrack volume settings? Robin Schriebman via GitHub
• Re: [webrtc-stats] Add estimatedClockSkew Robin Schriebman via GitHub
• Re: [webrtc-stats] RTCMediaStreamTrackStats.concealedAudibleSamples Robin Schriebman via GitHub
• Re: [webrtc-stats] RTCQualityLimitationReason and friends Varun Singh via GitHub
• Re: [webrtc-stats] RTCMediaStreamTrackStats is four dictionaries in one henbos via GitHub
• Re: [webrtc-stats] RTCMediaStreamTrackStats is four dictionaries in one henbos via GitHub
• Re: [webrtc-stats] We need "sender" and "receiver" stats, not "track" stats Harald Alvestrand via GitHub
• Re: [webrtc-stats] RTCMediaStreamTrackStats is four dictionaries in one Harald Alvestrand via GitHub
• Re: [webrtc-stats] Need DSCP information for incoming RTP streams Harald Alvestrand via GitHub
• Re: [webrtc-stats] Need DSCP information for outgoing RTP streams Varun Singh via GitHub
• Re: [webrtc-stats] Need DSCP information for outgoing RTP streams Harald Alvestrand via GitHub
• Re: [webrtc-stats] Privacy & Security self review Harald Alvestrand via GitHub
• Closed: [webrtc-stats] Privacy & Security self review Harald Alvestrand via GitHub
• [webrtc-stats] new commits pushed by vr000m Varun Singh via GitHub
• [webrtc-stats] new commits pushed by henbos henbos via GitHub
• Closed: [webrtc-stats] Discuss caching and consistency of getStats() return henbos via GitHub
• Re: [webrtc-stats] Complete security and privacy considerations based on self-review Harald Alvestrand via GitHub
• Re: [webrtc-stats] Added Guidelines for getStats() results caching Harald Alvestrand via GitHub
• Re: [webrtc-stats] Added Guidelines for getStats() results caching Varun Singh via GitHub
• Re: [webrtc-stats] RTCMediaStreamTrackStats.concealedAudibleSamples Harald Alvestrand via GitHub
• Re: [webrtc-stats] RTCMediaStreamTrackStats.concealedAudibleSamples added. Varun Singh via GitHub
• Re: [webrtc-pc] behaviour of offerToReceive* set to false when there is a local track Philipp Hancke via GitHub
• Re: [webrtc-pc] order of transceivers, senders/receivers? Philipp Hancke via GitHub
• Re: [webrtc-pc] behaviour of offerToReceive* set to false when there is a local track stefan hakansson via GitHub
• Re: [webrtc-pc] RTCRtpSynchronizationSource.audioLevel in idl should be nullable Harald Alvestrand via GitHub
• Re: [webrtc-pc] order of transceivers, senders/receivers? Harald Alvestrand via GitHub

Tuesday, 21 November 2017

• [webrtc-pc] order of transceivers, senders/receivers? Philipp Hancke via GitHub
• Re: [webrtc-pc] behaviour of offerToReceive* set to false when there is a local track Philipp Hancke via GitHub
• Re: [webrtc-pc] behaviour of offerToReceive* set to false when there is a local track stefan hakansson via GitHub
• Closed: [webrtc-pc] RTCRtpSynchronizationSource.audioLevel in idl should be nullable Bernard Aboba via GitHub
• Re: [webrtc-pc] fix RTCRtpSynchronizationSource.audioLevel idl to be nullable Bernard Aboba via GitHub
• Re: [webrtc-pc] behaviour of offerToReceive* set to false when there is a local track Philipp Hancke via GitHub
• Re: [webrtc-pc] RTCRtpSynchronizationSource.audioLevel in idl should be nullable na-g via GitHub
• Re: [webrtc-pc] Adding more values to RTCIceTransportPolicy Enum annmarielarue via GitHub
• Re: [webrtc-pc] RTCRtpSynchronizationSource.audioLevel in idl should be nullable jan-ivar via GitHub
• [webrtc-pc] Pull Request: RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries jan-ivar via GitHub

Monday, 20 November 2017

• Re: [webrtc-pc] Clarify whether RTCRtpContributingSource members are live. jan-ivar via GitHub
• Closed: [webrtc-pc] audioLevel of tracks, both send and receive? Bernard Aboba via GitHub
• Re: [webrtc-pc] audioLevel of tracks, both send and receive? Bernard Aboba via GitHub
• Re: [webrtc-pc] Adding more values to RTCIceTransportPolicy Enum Bernard Aboba via GitHub
• Re: [webrtc-pc] Need for Initial Bitrate by the Application/RtpSender? Bernard Aboba via GitHub
• Re: [webrtc-pc] Add a new value to RTCIceTransportPolicy. Bernard Aboba via GitHub
• Re: [webrtc-stats] Add per layer stats for SVC Harald Alvestrand via GitHub
• Re: [webrtc-pc] fix RTCRtpSynchronizationSource.audioLevel idl to be nullable Dominique Hazael-Massieux via GitHub
• [webrtc-stats] Add per layer stats for SVC Sergey Silkin via GitHub

Saturday, 18 November 2017

• Re: [webrtc-pc] Add testing guideline for naming test files and adding comments Philipp Hancke via GitHub

Friday, 17 November 2017

• [webrtc-pc] Pull Request: Set muted before SRD resolves, using new set muted algorithm. jan-ivar via GitHub
• [webrtc-pc] Pull Request: fix RTCRtpSynchronizationSource.audioLevel idl to be nullable na-g via GitHub
• [webrtc-pc] RTCRtpSynchronizationSource.audioLevel in idl should be nullable na-g via GitHub

Thursday, 16 November 2017

• Re: [webrtc-pc] addTransceiver woes jan-ivar via GitHub
• Re: [webrtc-pc] Isolated Media Streams requires modification on permission algorithms in GUM and Permissions specs jan-ivar via GitHub
• Re: [webrtc-stats] Pivot from "track" to "sender" and "receiver" stats. jan-ivar via GitHub
• Re: [webrtc-stats] Added 'objectDeleted' attribute Varun Singh via GitHub
• Re: [webrtc-stats] Split RTCMediaStreamTrackStats into four dictionaries. Varun Singh via GitHub
• Re: [webrtc-stats] Pivot from "track" to "sender" and "receiver" stats. Varun Singh via GitHub
• [webrtc-stats] new commits pushed by vr000m Varun Singh via GitHub
• Closed: [webrtc-stats] Add stat for network type of ICE candidate Varun Singh via GitHub

Wednesday, 15 November 2017

• Re: [webrtc-stats] Rebase of Taylor's NetworkType PR Gary via GitHub

Tuesday, 14 November 2017

• Re: [webrtc-pc] addTransceiver woes Philipp Hancke via GitHub

Monday, 13 November 2017

• Re: [webrtc-pc] Point to the Hash Function Textual Names registry Philip Jägenstedt via GitHub
• Re: [webrtc-pc] addTransceiver woes Nils Ohlmeier via GitHub
• Re: [webrtc-pc] addTransceiver woes Nils Ohlmeier via GitHub
• Re: [webrtc-pc] addTransceiver woes Harald Alvestrand via GitHub

Sunday, 12 November 2017

• Re: [webrtc-pc] Add testing guideline for naming test files and adding comments Harald Alvestrand via GitHub

Friday, 10 November 2017

• Re: [webrtc-stats] Rebase of Taylor's NetworkType PR Gary via GitHub
• Re: [webrtc-stats] Rebase of Taylor's NetworkType PR Harald Alvestrand via GitHub

Thursday, 9 November 2017

• [webrtc-pc] Pull Request: Add testing guideline for naming test files and adding comments Soares Chen via GitHub
• [webrtc-stats] Pull Request: Rebase of Taylor's NetworkType PR Harald Alvestrand via GitHub
• Re: [webrtc-stats] Adding "networkType" field to RTCIceCandidateStats. Harald Alvestrand via GitHub
• Re: [webrtc-stats] Adding "networkType" field to RTCIceCandidateStats. Harald Alvestrand via GitHub

Wednesday, 8 November 2017

• [webrtc-pc] Point to the Hash Function Textual Names registry Philip Jägenstedt via GitHub
• Re: [webrtc-pc] addTransceiver woes Philipp Hancke via GitHub
• [webrtc-pc] addTransceiver woes Philipp Hancke via GitHub
• [webrtc-pc] new commits pushed by burnburn Daniel Burnett via GitHub
• [webrtc-pc] new commits pushed by burnburn Daniel Burnett via GitHub
• Closed: [webrtc-pc] Adopt "test as you commit" policy Adam Bergkvist via GitHub
• [webrtc-pc] new commits pushed by adam-be Adam Bergkvist via GitHub
• Re: [webrtc-pc] Document "test as you commit" policy in CONTRIBUTING.md Philip Jägenstedt via GitHub

Tuesday, 7 November 2017

• Re: [webrtc-pc] Document "test as you commit" policy in CONTRIBUTING.md Daniel Burnett via GitHub
• Re: [webrtc-stats] Is keeping stats around a memory problem? jan-ivar via GitHub
• Re: [webrtc-pc] Document "test as you commit" policy in CONTRIBUTING.md Philip Jägenstedt via GitHub
• Re: [webrtc-pc] Clarify whether RTCRtpContributingSource members are live. jan-ivar via GitHub

Monday, 6 November 2017

• Re: [webrtc-pc] Stats & isolated streams Dominique Hazael-Massieux via GitHub
• [webrtc-pc] Pull Request: fix ref to webidl Cullen Jennings via GitHub
• Re: [webrtc-pc] canInsertDTMF transitions not specified Cullen Jennings via GitHub
• Re: [webrtc-stats] Is keeping stats around a memory problem? Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] Need for Initial Bitrate by the Application/RtpSender? Cullen Jennings via GitHub
• Re: [webrtc-pc] JSEP references are out dated Cullen Jennings via GitHub
• Re: [webrtc-pc] Isolated Media Streams requires modification on permission algorithms in GUM and Permissions specs Soares Chen via GitHub
• Re: [webrtc-pc] Document "test as you commit" policy in CONTRIBUTING.md Philip Jägenstedt via GitHub
• Re: [webrtc-pc] Isolated Media Streams requires modification on permission algorithms in GUM and Permissions specs Harald Alvestrand via GitHub
• Re: [webrtc-pc] relay-first as an option for RTCIceTransportPolicy Philipp Hancke via GitHub
• [webrtc-pc] relay-first as an option for RTCIceTransportPolicy Varun Singh via GitHub
• Re: [webrtc-pc] Adopt "test as you commit" policy Philip Jägenstedt via GitHub
• Re: [webrtc-pc] RTCSctpTransport: Specify special cases for maxMessageSize Adam Bergkvist via GitHub
• [webrtc-pc] Pull Request: RTCSctpTransport: Specify special cases for maxMessageSize Adam Bergkvist via GitHub
• Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case Adam Bergkvist via GitHub
• Re: [webrtc-pc] Stats & isolated streams Harald Alvestrand via GitHub
• [webrtc-stats] Pull Request: Pivot from "track" to "sender" and "receiver" stats. jan-ivar via GitHub
• [webrtc-stats] Pull Request: Split RTCMediaStreamTrackStats into four dictionaries. jan-ivar via GitHub

Saturday, 4 November 2017

• Re: [webrtc-pc] Section 11: Examples Jxck via GitHub
• Re: [webrtc-pc] Meta: auto-publish changes to the spec Philip Jägenstedt via GitHub

Friday, 3 November 2017

• Re: [webrtc-stats] Adding "networkType" field to RTCIceCandidateStats. Gary via GitHub
• Re: [webrtc-stats] Add stat for inputAudioLevel, before the audio filter Huib Kleinhout via GitHub
• Re: [webrtc-pc] Adding more values to RTCIceTransportPolicy Enum jianjunz via GitHub
• Re: [webrtc-pc] Adding more values to RTCIceTransportPolicy Enum Nils Ohlmeier via GitHub
• Re: [webrtc-stats] Add stat for inputAudioLevel, before the audio filter Harald Alvestrand via GitHub

Thursday, 2 November 2017

• [webrtc-stats] Add stat for inputAudioLevel, before the audio filter Huib Kleinhout via GitHub
• Re: [webrtc-stats] jitterBufferDelay vs playoutDelay henbos via GitHub
• Re: [webrtc-stats] We need "sender" and "receiver" stats, not "track" stats jan-ivar via GitHub
• Re: [webrtc-stats] jitterBufferDelay vs playoutDelay Varun Singh via GitHub
• Closed: [webrtc-stats] jitterBufferDelay vs playoutDelay Varun Singh via GitHub
• [webrtc-stats] new commits pushed by vr000m Varun Singh via GitHub
• Re: [webrtc-stats] What is fractionLost for a local incoming media stream? Varun Singh via GitHub
• Closed: [webrtc-pc] setIdentityProvider should throw error if protocol contains invalid characters Bernard Aboba via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• Closed: [webrtc-pc] RTCPeerConnection.close should only refer to the [[SctpTransport]] slot Bernard Aboba via GitHub
• [webrtc-pc] new commits pushed by aboba Bernard Aboba via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [webrtc-pc] [[SctpTransportState]] slot needs to be defined and initialized Harald Alvestrand via GitHub
• Closed: [webrtc-pc] Sending data channel messages > maxMessageSize Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Closed: [webrtc-pc] Why is setDirection a method? Harald Alvestrand via GitHub
• [webrtc-pc] new commits pushed by alvestrand Harald Alvestrand via GitHub
• Re: [webrtc-pc] Prepare status of the document for CR publication Dominique Hazael-Massieux via GitHub
• Re: [webrtc-pc] Prepare status of the document for CR publication Harald Alvestrand via GitHub
• Re: [webrtc-pc] Section 11: Examples Adam Bergkvist via GitHub
• Closed: [webrtc-pc] Section 6.4: Datachannel Garbage Collection Adam Bergkvist via GitHub
• [webrtc-pc] Pull Request: Use [[SctpTransport]] slot in RTCPeerConnection.close Adam Bergkvist via GitHub
• [webrtc-pc] Pull Request: Add steps to create an RTCSctpTransport Adam Bergkvist via GitHub
• Re: [webrtc-pc] Adding relativeBitrate parameter to RTCRtpEncodingParameters. stefan hakansson via GitHub
• Re: [webrtc-pc] Prepare status of the document for CR publication stefan hakansson via GitHub
• Re: [webrtc-pc] Validate protocol string in IdP operations stefan hakansson via GitHub
• [webrtc-pc] [[SctpTransport]] slot needs to be defined and initialized Adam Bergkvist via GitHub
• [webrtc-pc] RTCPeerConnection.close should only refer to one RTCSctpTransport Adam Bergkvist via GitHub
• [webrtc-pc] new commits pushed by adam-be Adam Bergkvist via GitHub
• Re: [webrtc-pc] Section 6.4: Datachannel Garbage Collection Adam Bergkvist via GitHub
• Re: [webrtc-stats] What is fractionLost for a local incoming media stream? Varun Singh via GitHub

Wednesday, 1 November 2017

• Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case Lennart Grahl via GitHub
• Re: [webrtc-pc] RTCSctpTransport.maxMessageSize 0 case Adam Bergkvist via GitHub
• Re: [webrtc-pc] Specify how RTCSctpTransport.maxMessageSize gets its value Adam Bergkvist via GitHub
• Re: [webrtc-pc] Specify how RTCSctpTransport.maxMessageSize gets its value Adam Bergkvist via GitHub
• [webrtc-pc] new commits pushed by adam-be Adam Bergkvist via GitHub
• Re: [webrtc-pc] Inconsistencies in Asynchronous Task Queueing Adam Bergkvist via GitHub

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