Weekly github digest (WebRTC WG specifications)

Issues
------
* w3c/webrtc-pc (+2/-7/💬4)
  2 issues created:
  - RTCSctpTransportState "new" never used (by lgrahl)
    https://github.com/w3c/webrtc-pc/issues/1874 
  - sendEncodings in "create an RTPSender" should reflect platform capabilities (by alvestrand)
    https://github.com/w3c/webrtc-pc/issues/1872 

  4 issues received 4 new comments:
  - #1872 sendEncodings in "create an RTPSender" should reflect platform capabilities (1 by jan-ivar)
    https://github.com/w3c/webrtc-pc/issues/1872 [May 2018 interim] 
  - #1745 Codify refusing to generate an empty Offer (1 by jan-ivar)
    https://github.com/w3c/webrtc-pc/issues/1745 [Needs PR] 
  - #1858 What happens when an answerer stops a transceiver that others are "bundled" on? (1 by jan-ivar)
    https://github.com/w3c/webrtc-pc/issues/1858 [May 2018 interim] 
  - #1829 Data channel minimum amount of streams supported (1 by lgrahl)
    https://github.com/w3c/webrtc-pc/issues/1829 [PR exists] 

  7 issues closed:
  - No way to add track with stream associations to a given transceiver  https://github.com/w3c/webrtc-pc/issues/1826 [April 2018 interim] [PR exists] 
  - Simulcast makes no sense for audio. https://github.com/w3c/webrtc-pc/issues/1813 [May 2018 interim] [PR exists] 
  - replaceTrack algorithm lost some important text https://github.com/w3c/webrtc-pc/issues/1855 [PR exists] 
  - Data channel minimum amount of streams supported https://github.com/w3c/webrtc-pc/issues/1829 [PR exists] 
  - Not possible to tell how old `RTCRtpContributingSource.timestamp` is https://github.com/w3c/webrtc-pc/issues/1497 [PR exists] 
  - RTCRtpContributingSource.timestamp needs a clearer definition https://github.com/w3c/webrtc-pc/issues/1690 [PR exists] 
  - Should a signalingstatechange event be fired when closing a RTCPeerConnection? https://github.com/w3c/webrtc-pc/issues/1799 

* w3c/webrtc-stats (+1/-4/💬15)
  1 issues created:
  - Identifier ambiguity in stat objects (by balazskreith)
    https://github.com/w3c/webrtc-stats/issues/350 

  9 issues received 15 new comments:
  - #345 Write a "how to add more stats" guideline document (3 by henbos, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/345 [Ready for PR] 
  - #238 Add stat to reflect the redundancy of FEC/RED data (2 by vr000m, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/238 [Submitter input needed] 
  - #271 Add stat for inputAudioLevel, before the audio filter (2 by vr000m, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/271 
  - #275 Add per layer stats for SVC (2 by ssilkin, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/275 [Submitter input needed] 
  - #347 Should WebRTC be [SecureContext] (2 by ylafon, alvestrand)
    https://github.com/w3c/webrtc-stats/issues/347 [Submitter input needed] 
  - #324 Clarify stats hierarchies (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/324 [Ready for PR] 
  - #318 Work through implications of simulcast on the receiver side (1 by alvestrand)
    https://github.com/w3c/webrtc-stats/issues/318 [Icebox] 
  - #348 add a graph showing the relationship of stats to the spec (1 by henbos)
    https://github.com/w3c/webrtc-stats/issues/348 
  - #350 Identifier ambiguity in stat objects (1 by taylor-b)
    https://github.com/w3c/webrtc-stats/issues/350 

  4 issues closed:
  - Identifier ambiguity in stat objects https://github.com/w3c/webrtc-stats/issues/350 
  - Unable to distinguish "sender" stats if multiple senders are sending the same track https://github.com/w3c/webrtc-stats/issues/342 [Icebox] 
  - Lifetime of RTPStreamStats https://github.com/w3c/webrtc-stats/issues/302 [PR exists] 
  - Consider the Reporting API as a stats export mechanism https://github.com/w3c/webrtc-stats/issues/260 [Icebox] 



Pull requests
-------------
* w3c/webrtc-pc (+1/-7/💬1)
  1 pull requests submitted:
  - Make RTCSctpTransport.maxChannels nullable & line breaks (by lgrahl)
    https://github.com/w3c/webrtc-pc/pull/1873 

  1 pull requests received 1 new comments:
  - #1867 Fire events in SLD(answer) in direction rejection edge-case. (1 by jan-ivar)
    https://github.com/w3c/webrtc-pc/pull/1867 

  7 pull requests merged:
  - Make RTCCertificate unusable by other origins
    https://github.com/w3c/webrtc-pc/pull/1870 
  - Trim [[SendEncodings]] down to what user agent supports.
    https://github.com/w3c/webrtc-pc/pull/1862 [Editors can integrate] 
  - Add sender.setStreams() method to update stream associations.
    https://github.com/w3c/webrtc-pc/pull/1860 [Editors can integrate] 
  - Data channel receive procedure
    https://github.com/w3c/webrtc-pc/pull/1849 
  - RTCSctpTransport.maxChannels & SCTP transport connected procedure
    https://github.com/w3c/webrtc-pc/pull/1848 
  - rtp media api: remove note
    https://github.com/w3c/webrtc-pc/pull/1832 
  - Define the reference clock used for RTCRtpContributingSource.timestamp
    https://github.com/w3c/webrtc-pc/pull/1854 [Editors can integrate] [Needs submitter action] 

* w3c/webrtc-stats (+2/-2/💬0)
  2 pull requests submitted:
  - Fixing some descriptions of ID references between stats objects. (by taylor-b)
    https://github.com/w3c/webrtc-stats/pull/351 
  - Define creation time for RTP objects (by alvestrand)
    https://github.com/w3c/webrtc-stats/pull/349 

  2 pull requests merged:
  - Fixing some descriptions of ID references between stats objects.
    https://github.com/w3c/webrtc-stats/pull/351 
  - Define creation time for RTP objects
    https://github.com/w3c/webrtc-stats/pull/349 


Repositories tracked by this digest:
-----------------------------------
* https://github.com/w3c/webrtc-pc
* https://github.com/w3c/webrtc-stats
* https://github.com/w3c/webrtc-charter