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Weekly github digest (WebRTC WG specifications)

From: W3C Webmaster via GitHub API <sysbot+gh@w3.org>
Date: Tue, 31 Oct 2017 17:00:29 +0000
To: public-webrtc@w3.org
Message-Id: <E1e9Ztt-0005yu-OR@uranus.w3.org>



Issues
------
* w3c/webrtc-pc (+4/-4/💬35)
  4 issues created:
  - Provide changelog for the spec in new editors draft workflow (by dontcallmedom)
    https://github.com/w3c/webrtc-pc/issues/1648
  - Isolated Media Streams requires modification on permission algorithms in GUM and Permissions specs (by soareschen)
    https://github.com/w3c/webrtc-pc/issues/1646
  - getIdentityAssertion should reject with InvalidAccessError if no identity provider is set (by soareschen)
    https://github.com/w3c/webrtc-pc/issues/1645
  - Adding more values to RTCIceTransportPolicy Enum (by jianjunz)
    https://github.com/w3c/webrtc-pc/issues/1644

  10 issues received 35 new comments:
  - #1644 Adding more values to RTCIceTransportPolicy Enum (9 by alvestrand, taylor-b, jianjunz, aboba)
    https://github.com/w3c/webrtc-pc/issues/1644
  - #1533 Clarify whether RTCRtpContributingSource members are live. (7 by fluffy, taylor-b, alvestrand, aboba, na-g)
    https://github.com/w3c/webrtc-pc/issues/1533
  - #1643 Why is setDirection a method? (6 by taylor-b, adam-be, fippo, jan-ivar, alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1643
  - #1646 Isolated Media Streams requires modification on permission algorithms in GUM and Permissions specs (4 by aboba, alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1646
  - #1446 RTCSctpTransport.maxMessageSize 0 case (3 by lgrahl, taylor-b)
    https://github.com/w3c/webrtc-pc/issues/1446
  - #1613 Stats & isolated streams (2 by dontcallmedom, alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1613
  - #1248 setParameters woes (1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/1248
  - #1635 Need for Initial Bitrate by the Application/RtpSender? (1 by aboba)
    https://github.com/w3c/webrtc-pc/issues/1635
  - #942 Meta: auto-publish changes to the spec (1 by dontcallmedom)
    https://github.com/w3c/webrtc-pc/issues/942
  - #1640 offerToReceiveAudio/Video processing creates sendrecv transceivers (1 by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1640

  4 issues closed:
  - Section 9.6: protocol type https://github.com/w3c/webrtc-pc/issues/1293
  - pc.close() method should close sctp transport https://github.com/w3c/webrtc-pc/issues/1381
  - No way to tell when SCTP association is closed https://github.com/w3c/webrtc-pc/issues/1612
  - Missing detail on obtaining lastOffer/lastAnswer in setLocalDescription https://github.com/w3c/webrtc-pc/issues/1164

* w3c/webrtc-stats (+0/-5/💬29)
  18 issues received 29 new comments:
  - #256 Stats for adaptation reason, for realsies (4 by henbos, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/256
  - #193 RTCMediaStreamTrackStats.audioLevel clarification (3 by henbos, taylor-b, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/193
  - #246 jitterBufferDelay and concealed samples, DTX/CNG samples (3 by henbos, vr000m, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/246
  - #258 Add estimatedClockSkew (2 by henbos, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/258
  - #135 Need DSCP information for incoming RTP streams (2 by taylor-b, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/135
  - #222 Audio/Video sync follow-up (2 by icydragons, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/222
  - #229 Interframe delay stat for video receive stream. (2 by ilyanikolaevskiy, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/229
  - #133 Need DSCP information for outgoing RTP streams (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/133
  - #161 Definitions from MSE need re-targeting (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/161
  - #101 Move to continuous publication (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/101
  - #98 Add "privacy" to security considerations (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/98
  - #230 RTCMediaStreamTrackStats is four dictionaries in one (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/230
  - #231 We need "sender" and "receiver" stats, not "track" stats (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/231
  - #240 Stats for Audio network adaptation (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/240
  - #241 Is bytesReceived really available for RTCRemoteInboundRTPStreamStats? (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/241
  - #253 packetsLost is unsigned but can be negative according to RFC (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/253
  - #254 add packetsDuplicated as a result of #253 (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/254
  - #255 What is fractionLost for a local incoming media stream? (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/255

  5 issues closed:
  - RTCMediaStreamTrackStats.jitterBufferDelay: Specify unit in seconds https://github.com/w3c/webrtc-stats/issues/244
  - packetsLost is unsigned but can be negative according to RFC https://github.com/w3c/webrtc-stats/issues/253
  - Add "privacy" to security considerations https://github.com/w3c/webrtc-stats/issues/98
  - Difference between 'Id' and 'Identifier' fields https://github.com/w3c/webrtc-stats/issues/147
  - Move to continuous publication https://github.com/w3c/webrtc-stats/issues/101



Pull requests
-------------
* w3c/webrtc-pc (+3/-3/💬13)
  3 pull requests submitted:
  - Specify how RTCSctpTransport.maxMessageSize gets its value (by adam-be)
    https://github.com/w3c/webrtc-pc/pull/1650
  - Add a new value to RTCIceTransportPolicy. (by jianjunz)
    https://github.com/w3c/webrtc-pc/pull/1649
  - Replace setDirection() with writable direction attribute (by adam-be)
    https://github.com/w3c/webrtc-pc/pull/1647

  6 pull requests received 13 new comments:
  - #1618 format & fixup Example code (6 by adam-be, Jxck)
    https://github.com/w3c/webrtc-pc/pull/1618
  - #1608 Validate protocol string in IdP operations (3 by soareschen, aboba, stefhak)
    https://github.com/w3c/webrtc-pc/pull/1608
  - #1650 Specify how RTCSctpTransport.maxMessageSize gets its value (1 by lgrahl)
    https://github.com/w3c/webrtc-pc/pull/1650
  - #1209 Throw error if data channel's buffer is filled, rather than closing. (1 by lgrahl)
    https://github.com/w3c/webrtc-pc/pull/1209
  - #1647 Replace setDirection() with writable direction attribute (1 by adam-be)
    https://github.com/w3c/webrtc-pc/pull/1647
  - #1151 Prepare status of the document for CR publication (1 by aboba)
    https://github.com/w3c/webrtc-pc/pull/1151

  3 pull requests merged:
  - Set up automatic PR review request on webrtc.html
    https://github.com/w3c/webrtc-pc/pull/1622
  - Let setLocalDescription use [[LastAnswer/Offer]] internal slots
    https://github.com/w3c/webrtc-pc/pull/1638
  - format & fixup Example code
    https://github.com/w3c/webrtc-pc/pull/1618

* w3c/webrtc-stats (+5/-3/💬11)
  5 pull requests submitted:
  - RTCQualityLimitationReason and friends (by henbos)
    https://github.com/w3c/webrtc-stats/pull/270
  - signed packetsLost (by henbos)
    https://github.com/w3c/webrtc-stats/pull/269
  - jitterBufferOutput added (by henbos)
    https://github.com/w3c/webrtc-stats/pull/268
  - jitterBufferDelay in seconds (by henbos)
    https://github.com/w3c/webrtc-stats/pull/267
  - TAG review: Link to design principles (by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/266

  5 pull requests received 11 new comments:
  - #268 jitterBufferOutput added (4 by henbos, vr000m)
    https://github.com/w3c/webrtc-stats/pull/268
  - #259 Adding "networkType" field to RTCIceCandidateStats. (3 by taylor-b, alvestrand)
    https://github.com/w3c/webrtc-stats/pull/259
  - #270 RTCQualityLimitationReason and friends (2 by henbos)
    https://github.com/w3c/webrtc-stats/pull/270
  - #266 TAG review: Link to design principles (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/266
  - #269 signed packetsLost (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/269

  3 pull requests merged:
  - jitterBufferDelay in seconds
    https://github.com/w3c/webrtc-stats/pull/267
  - signed packetsLost
    https://github.com/w3c/webrtc-stats/pull/269
  - Add a paragraph about "identifier" when "id" is occupied
    https://github.com/w3c/webrtc-stats/pull/264


Repositories tracked by this digest:
-----------------------------------
* https://github.com/w3c/webrtc-pc
* https://github.com/w3c/webrtc-stats
Received on Tuesday, 31 October 2017 17:00:34 UTC

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