- From: Bernard Aboba via GitHub <sysbot+gh@w3.org>
- Date: Mon, 10 Jul 2017 16:11:06 +0000
- To: public-ortc@w3.org
aboba has just created a new issue for https://github.com/w3c/ortc: == Sync with WebRTC 1.0 == The following merged WebRTC 1.0 PRs may relate to ORTC as well: https://github.com/w3c/webrtc-pc/pull/1455 (Label/Protocol length restrictions) https://github.com/w3c/webrtc-pc/pull/1453 (Key shortening text) https://github.com/w3c/webrtc-pc/pull/1433 (NotSupportedError for unknown ICE server schema) https://github.com/w3c/webrtc-pc/pull/1429 (ICE TCP types) https://github.com/w3c/webrtc-pc/pull/1404 (Unnecessary conditions in datachannel send) https://github.com/w3c/webrtc-pc/pull/1395 (Internal slot naming) https://github.com/w3c/webrtc-pc/pull/1388 (nullable argument to replaceTrack/setTrack) https://github.com/w3c/webrtc-pc/pull/1373 (DTMF playout algorithm for comma) https://github.com/w3c/webrtc-pc/pull/1358 (RTCDataChannel use internal slots) https://github.com/w3c/webrtc-pc/pull/1356 (createDataChannel: use TypeError) https://github.com/w3c/webrtc-pc/pull/1350 (case sensitivity of RID characters) https://github.com/w3c/webrtc-pc/pull/1348 (privacy impact of default configured ICE servers) https://github.com/w3c/webrtc-pc/pull/1338 (insertDTMF replaces tone buffer) https://github.com/w3c/webrtc-pc/pull/1337 (fix DTMF examples) https://github.com/w3c/webrtc-pc/pull/1335 (typo in DTMF playout steps) https://github.com/w3c/webrtc-pc/pull/1329 (update maxbitrate definition) https://github.com/w3c/webrtc-pc/pull/1298 (intertone gap maximum) https://github.com/w3c/webrtc-pc/pull/1239 (RTCIceConnectionEventInit: url is nullable) https://github.com/w3c/webrtc-pc/pull/1230 (use data-cite for RTCStatsType) https://github.com/w3c/webrtc-pc/pull/1226 (remove webidl defaults for RTP parameters) https://github.com/w3c/webrtc-pc/pull/1225 (units for maxframerate) https://github.com/w3c/webrtc-pc/pull/1209 (throw error if datachannel buffer is filled rather than closing) https://github.com/w3c/webrtc-pc/pull/1176 (RTCIceTransportPolicy enum descriptions) https://github.com/w3c/webrtc-pc/pull/1153 (Constructor for RTCIceCandidate) https://github.com/w3c/webrtc-pc/pull/1149 (RTCContributingSources) https://github.com/w3c/webrtc-pc/pull/1137 (RTCDataChannel.id default value) https://github.com/w3c/webrtc-pc/pull/1133 (Split getContributingSources) https://github.com/w3c/webrtc-pc/pull/1131 (USVString handling) https://github.com/w3c/webrtc-pc/pull/1115 (DTLS failures) https://github.com/w3c/webrtc-pc/pull/1109 (ptime member of RTCRtpEncodingParameters) https://github.com/w3c/webrtc-pc/pull/1100 (when RTCRtpContributingSource.audioLevel can be null) https://github.com/w3c/webrtc-pc/pull/1099 (update the RTCRtpContributingSource for SSRCs) https://github.com/w3c/webrtc-pc/pull/1098 (Attempt to update RTCRtpContributingSource objects at playout time) Please view or discuss this issue at https://github.com/w3c/ortc/issues/716 using your GitHub account
Received on Monday, 10 July 2017 16:11:12 UTC