Adam Bergkvist via GitHub
- [webrtc-pc] Pull Request: Set direction of transceiver created by offerToReceive* option
- [webrtc-pc] Pull Request: Editorial: Add IANA-HASH-FUNCTION reference
- [webrtc-pc] Pull Request: RTCSctpTransport: Specify special cases for maxMessageSize
- [webrtc-pc] Pull Request: Use [[SctpTransport]] slot in RTCPeerConnection.close
- [webrtc-pc] Pull Request: Add steps to create an RTCSctpTransport
- [webrtc-pc] [[SctpTransport]] slot needs to be defined and initialized
- [webrtc-pc] RTCPeerConnection.close should only refer to one RTCSctpTransport
Bernard Aboba
Cullen Jennings (fluffy)
Cullen Jennings via GitHub
Dominique Hazael-Massieux
- [minutes] TPAC F2F Nov 6-7
- Re: Recording of divs on a page
- Re: Recording of divs on a page
- WebRTC 1.0 published as Candidate Recommendation
Harald Alvestrand
- Re: A very short extension spec: DSCP codepoint control
- Re: A very short extension spec: DSCP codepoint control
- Re: A very short extension spec: DSCP codepoint control
- Re: A very short extension spec: DSCP codepoint control
- Re: A very short extension spec: DSCP codepoint control
- Re: A very short extension spec: DSCP codepoint control
- A very short extension spec: DSCP codepoint control
- Re: My personal input to the WebRTC-NV discussions
- My personal input to the WebRTC-NV discussions
Harald Alvestrand via GitHub
- [webrtc-pc] RTCPriorityType text is outdated
- [webrtc-pc] RTCPeerConnection constructor can fail - what error to return?
- [webrtc-stats] Pull Request: Rebase of Taylor's NetworkType PR
Huib Kleinhout via GitHub
jan-ivar via GitHub
- [webrtc-pc] offerToReceive* should ignore stopped transceivers, not unstopped ones.
- [webrtc-pc] Pull Request: RTCRtpContributingSource & RTCRtpSynchronizationSource -> dictionaries
- [webrtc-pc] Pull Request: Set muted before SRD resolves, using new set muted algorithm.
- [webrtc-stats] Pull Request: Pivot from "track" to "sender" and "receiver" stats.
- [webrtc-stats] Pull Request: Split RTCMediaStreamTrackStats into four dictionaries.
Lennart Grahl
na-g via GitHub
- [webrtc-pc] Pull Request: fix RTCRtpSynchronizationSource.audioLevel idl to be nullable
- [webrtc-pc] RTCRtpSynchronizationSource.audioLevel in idl should be nullable
Peter Thatcher
Philip Jägenstedt via GitHub
Philipp Hancke
Philipp Hancke via GitHub
Randell Jesup
Sergey Silkin via GitHub
Sergio Garcia Murillo
- Re: A very short extension spec: DSCP codepoint control
- Re: A very short extension spec: DSCP codepoint control
- Re: A very short extension spec: DSCP codepoint control
Silvia Pfeiffer
- Re: Recording of divs on a page
- Re: Recording of divs on a page
- Recording of divs on a page
- Re: My personal input to the WebRTC-NV discussions
- Re: My personal input to the WebRTC-NV discussions
Soares Chen via GitHub
Stefan Håkansson LK
- Re: A very short extension spec: DSCP codepoint control
- Re: A very short extension spec: DSCP codepoint control
- Adding Jan-Ivar to editing teams
- Fwd: WebRTC 1.0: Real-time Communication Between Browsers is a W3C Candidate Recommendation (Call for Implementations)
Varun Singh via GitHub
W3C Webmaster via GitHub API
- Weekly github digest (WebRTC WG specifications)
- Weekly github digest (WebRTC WG specifications)
- Weekly github digest (WebRTC WG specifications)
- Weekly github digest (WebRTC WG specifications)