[1] [CITE[WebRTC]]
( ([TIME[2011-06-02 05:33:46 +09:00]] 版))
<https://sites.google.com/site/webrtc/>

[2] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
( ([TIME[2011-08-23 23:59:49 +09:00]] 版))
<http://dev.w3.org/2011/webrtc/editor/webrtc.html>

[3] [CITE[IRC logs: freenode / #whatwg / 20111027]]
( ([TIME[2011-10-30 18:45:14 +09:00]] 版))
<http://krijnhoetmer.nl/irc-logs/whatwg/20111027>

[4] [CITE[WebRTC 1.0: Real-time Communication Between Browsers]]
( ([TIME[2011-10-25 17:28:07 +09:00]] 版))
<http://www.w3.org/TR/2011/WD-webrtc-20111027/>

[5] [CITE[IRC logs: freenode / #whatwg / 20111111]]
( ([TIME[2011-11-12 00:43:00 +09:00]] 版))
<http://krijnhoetmer.nl/irc-logs/whatwg/20111111>

[6] [CITE@en[Web Applications 1.0 r6837 Prepare a WebVTT standard. Remove the WebVTT text from the HTML standard; refer to the WebVTT spec instead. Drop PeerConnection (moved to W3C WebRTC WG). Clean up some internal annotations (sorry, this affects a number of topics but is purely editorial). Fix some WebIDL issues flagged by recent updates to the WebIDL checker.Affected topics: Canvas, DOM APIs, HTML, HTML Syntax and Parsing, Microdata, Offline Web Applications, Rendering, Security, Server-Sent Events, Video Text Tracks, Video and Audio, Web Storage, Web Workers, WebSocket API]]
( ([TIME[2011-11-29 08:04:00 +09:00]] 版))
<http://html5.org/tools/web-apps-tracker?from=6836&to=6837>

[7] [CITE[''''''[''''''whatwg'''''']'''''' PeerConnection feedback]]
( ([TIME[2011-12-03 17:31:00 +09:00]] 版))
<http://lists.whatwg.org/htdig.cgi/whatwg-whatwg.org/2011-December/034037.html>

[8] [CITE[最新のWeb RTC仕様について調べてみた - IT-Walker on hatena]]
( ([TIME[2011-12-12 23:38:59 +09:00]] 版))
<http://d.hatena.ne.jp/Syunpei/20111212/1323679295>

[9] ( ([TIME[2011-01-28 06:07:25 +09:00]] 版))
<http://hixie.ch/specs/rtc-skeleton/>

[10] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
( ([TIME[2012-02-07 18:36:11 +09:00]] 版))
<http://www.w3.org/TR/2012/WD-webrtc-20120209/>

[11] [CITE@en[RTCStreamAPI - MozillaWiki]]
( ([TIME[2012-05-01 01:39:51 +09:00]] 版))
<https://wiki.mozilla.org/RTCStreamAPI>

[12] [CITE@en[Language]]
( ([[Anne van Kesteren]] 著, [TIME[2012-06-01 20:47:56 +09:00]] 版))
<http://lists.w3.org/Archives/Public/public-webrtc/2012Jun/0002.html>

[13] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
( ([TIME[2012-08-21 21:52:00 +09:00]] 版))
<http://www.w3.org/TR/2012/WD-webrtc-20120821/>

[14] [CITE[IRC logs: freenode / #whatwg / 20130329]]
( ([TIME[2013-04-02 22:21:46 +09:00]] 版))
<http://krijnhoetmer.nl/irc-logs/whatwg/20130329#l-489>

[15] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
( ([TIME[2013-09-09 22:43:31 +09:00]] 版))
<http://www.w3.org/TR/2013/WD-webrtc-20130910/>

[16] ( ([TIME[2011-01-27 21:07:25 +09:00]] 版))
<http://hixie.ch/specs/rtc-skeleton/>

[17] [CITE@en[Identifiers for WebRTC's Statistics API]]
( ([TIME[2014-10-17 22:18:52 +09:00]] 版))
<http://www.w3.org/TR/2014/WD-webrtc-stats-20141021/>

[18] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
( ([TIME[2014-12-06 01:31:17 +09:00]] 版))
<http://w3c.github.io/webrtc-pc/>

[19] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
( ([TIME[2015-02-06 18:35:56 +09:00]] 版))
<http://www.w3.org/TR/2015/WD-webrtc-20150210/>

[20] [CITE@en[Identifiers for WebRTC's Statistics API]]
([TIME[2015-02-06 17:18:22 +09:00]] 版)
<http://www.w3.org/TR/2015/WD-webrtc-stats-20150206/>

[21] [CITE@en[RFC 7478 - Web Real-Time Communication Use Cases and Requirements]]
([TIME[2015-03-24 04:13:26 +09:00]] 版)
<https://tools.ietf.org/html/rfc7478>

[22] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
([TIME[2015-06-12 09:52:58 +09:00]] 版)
<http://w3c.github.io/webrtc-pc/>

[23] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
([TIME[2015-06-12 09:52:58 +09:00]] 版)
<https://w3c.github.io/webrtc-pc/>

[24] [CITE@en[976115 – Higher latency on WebRTC DataChannels than Chrome]]
([TIME[2015-08-16 13:47:14 +09:00]] 版)
<https://bugzilla.mozilla.org/show_bug.cgi?id=976115>

[25] [CITE@en[WebRTC SFU をフルスクラッチで作ってみた]]
([TIME[2015-11-05 11:56:44 +09:00]] 版)
<https://gist.github.com/voluntas/926bc137181ea208799d>

[26] [CITE@en[RFC 7742 - WebRTC Video Processing and Codec Requirements]]
([TIME[2016-03-30 01:05:47 +09:00]] 版)
<https://tools.ietf.org/html/rfc7742>

[27] [CITE@en[Identifiers for WebRTC's Statistics API]]
( ([TIME[2016-05-25 01:05:28 +09:00]]))
<https://www.w3.org/TR/2016/WD-webrtc-stats-20160524/>

[28] [CITE@en[Identifiers for WebRTC's Statistics API]]
( ([TIME[2016-05-27 19:22:19 +09:00]]))
<https://www.w3.org/TR/2016/WD-webrtc-stats-20160527/>

[29] [CITE@en[RFC 7874 - WebRTC Audio Codec and Processing Requirements]]
( ([TIME[2016-05-27 23:06:00 +09:00]]))
<https://tools.ietf.org/html/rfc7874>

[30] [CITE@en[RFC 7875 - Additional WebRTC Audio Codecs for Interoperability]]
( ([TIME[2016-05-26 11:55:50 +09:00]]))
<https://tools.ietf.org/html/rfc7875>

[31] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
( ([TIME[2016-05-31 23:05:03 +09:00]]))
<https://www.w3.org/TR/2016/WD-webrtc-20160531/>

[32] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
([TIME[2016-08-03 18:40:23 +09:00]])
<https://www.w3.org/TR/2016/WD-webrtc-20160803/>

[33] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
([TIME[2016-09-14 04:10:11 +09:00]])
<https://www.w3.org/TR/2016/WD-webrtc-20160913/>

[34] [CITE@en[Identifiers for WebRTC's Statistics API]]
([TIME[2016-09-21 22:05:04 +09:00]])
<https://www.w3.org/TR/2016/WD-webrtc-stats-20160921/>

[35] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
([TIME[2016-11-24 08:00:20 +09:00]])
<https://www.w3.org/TR/2016/WD-webrtc-20161123/>

[36] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
([TIME[2016-11-24 13:23:34 +09:00]])
<https://www.w3.org/TR/2016/WD-webrtc-20161124/>

[37] [CITE@en[Identifiers for WebRTC's Statistics API]]
([TIME[2016-12-14 18:15:19 +09:00]])
<https://www.w3.org/TR/2016/WD-webrtc-stats-20161214/>

[38] [CITE@en[Review of WebRTC Statistics from Web Performance Working Group]]
([[Stefan Håkansson LK]]著, [TIME[2016-12-14 18:29:48 +09:00]])
<https://lists.w3.org/Archives/Public/public-web-perf/2016Dec/0006.html>

[39] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
([TIME[2017-03-13 21:38:59 +09:00]])
<https://www.w3.org/TR/2017/WD-webrtc-20170313/>

[40] [CITE@en[Merge pull request #1083 from w3c/issue-1080-patch]]
([[aboba]]著, [TIME[2017-03-17 00:37:26 +09:00]])
<https://github.com/w3c/webrtc-pc/commit/7590741a7a08d07b50cde54b35289810c3fc116b>

[41] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
([TIME[2017-05-08 22:41:06 +09:00]])
<https://www.w3.org/TR/2017/WD-webrtc-20170508/>

[42] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
([TIME[2017-05-15 21:39:11 +09:00]])
<https://www.w3.org/TR/2017/WD-webrtc-20170515/>

[43] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
([TIME[2017-06-05 16:56:10 +09:00]])
<https://www.w3.org/TR/2017/WD-webrtc-20170605/>

[44] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
([TIME[2017-08-22 18:17:12 +09:00]])
<https://www.w3.org/TR/2017/WD-webrtc-20170822/>

[45] [CITE@en[Make the document WebRTC WG only]]
([[dontcallmedom]]著, [TIME[2017-09-22 17:51:17 +09:00]])
<https://github.com/w3c/mediacapture-main/commit/494babc000742f2ab5ece76223b71acf76813fb3>

[46] [CITE@en[Re: Fwd: Call for Consensus: discontinue joint ownership of Media  Capture Task Force, leaving to WebRTC Working Group; respond by 20 Sept 2017]]
([[Dominique Hazael-Massieux]]著, [TIME[2017-09-22 01:08:47 +09:00]])
<https://lists.w3.org/Archives/Public/public-media-capture/2017Sep/0021.html>

[47] [CITE[New WebKit Features in Safari 11 | WebKit]]
([TIME[2017-09-26 23:25:00 +09:00]])
<https://webkit.org/blog/7956/new-webkit-features-in-safari-11/>

[48] [CITE@en[Deprecations and Removals in Chrome 62  |  Web  |  Google Developers]]
([TIME[2017-09-18 12:54:25 +09:00]])
<https://developers.google.com/web/updates/2017/09/chrome-62-deprecations>

[49] [CITE@en[WebRTC becomes design-complete strengthening the Web Platform as a solid actor in the telecommunications arena]]
([TIME[2017-11-02 22:48:10 +09:00]])
<https://www.w3.org/2017/11/media-advisory-webrtc10-cr.html.en>

[50] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
([TIME[2017-11-01 02:33:59 +09:00]])
<https://www.w3.org/TR/2017/CR-webrtc-20171102/>

[51] [CITE@en[Identifiers for WebRTC's Statistics API]]
([TIME[2018-01-11 02:22:28 +09:00]])
<https://www.w3.org/TR/2018/WD-webrtc-stats-20180110/>

[52] [CITE@en[Identifiers for WebRTC's Statistics API]]
([TIME[2018-01-18 03:21:12 +09:00]])
<https://www.w3.org/TR/2018/WD-webrtc-stats-20180117/>

[53] [CITE@en[Identifiers for WebRTC's Statistics API]]
([TIME[2018-02-01 06:07:41 +09:00]])
<https://www.w3.org/TR/2018/WD-webrtc-stats-20180131/>

[54] [CITE@en[WebRTC 1.0: Real-time Communication Between Browsers]]
([TIME[2017-11-01 02:33:59 +09:00]])
<https://www.w3.org/TR/2017/CR-webrtc-20171102/>

[55] [CITE@en[WebRTC becomes design-complete strengthening the Web Platform as a solid actor in the telecommunications arena]]
([TIME[2017-11-02 22:48:10 +09:00]])
<https://www.w3.org/2017/11/media-advisory-webrtc10-cr.html.en>

[56] [CITE[New WebKit Features in Safari 11.1 | WebKit]]
([TIME[2018-04-13 13:40:37 +09:00]])
<https://webkit.org/blog/8216/new-webkit-features-in-safari-11-1/>

[57] [CITE[New WebKit Features in Safari 11.1 | WebKit]]
([TIME[2018-04-24 13:08:19 +09:00]])
<https://webkit.org/blog/8216/new-webkit-features-in-safari-11-1/>

[58] [CITE@en[Identifiers for WebRTC's Statistics API]]
([TIME[2018-01-11 02:22:28 +09:00]])
<https://www.w3.org/TR/2018/WD-webrtc-stats-20180110/>

[59] [CITE@en[Identifiers for WebRTC's Statistics API]]
([TIME[2018-01-18 03:21:12 +09:00]])
<https://www.w3.org/TR/2018/WD-webrtc-stats-20180117/>

[60] [CITE@en[Identifiers for WebRTC's Statistics API]]
([TIME[2018-02-01 06:07:41 +09:00]])
<https://www.w3.org/TR/2018/WD-webrtc-stats-20180131/>

[61] [CITE@en[Identifiers for WebRTC's Statistics API]]
([TIME[2018-05-16 22:21:52 +09:00]])
<https://www.w3.org/TR/2018/WD-webrtc-stats-20180516/>

[62] [CITE@en[Identifiers for WebRTC's Statistics API]]
([TIME[2018-05-20 04:34:40 +09:00]])
<https://www.w3.org/TR/2018/WD-webrtc-stats-20180519/>

[63] [CITE@en[Identifiers for WebRTC's Statistics API]]
([TIME[2018-06-27 22:08:11 +09:00]])
<https://www.w3.org/TR/2018/WD-webrtc-stats-20180627/>

[64] [CITE@en[Identifiers for WebRTC's Statistics API]]
([TIME[2018-05-16 22:21:52 +09:00]])
<https://www.w3.org/TR/2018/WD-webrtc-stats-20180516/>

[65] [CITE@en[Identifiers for WebRTC's Statistics API]]
([TIME[2018-05-20 04:34:40 +09:00]])
<https://www.w3.org/TR/2018/WD-webrtc-stats-20180519/>

-[66] [CITE@ja[nabe@abkさんはTwitterを使っています: 「1週間でWebRTCに異様に詳しくなっているけども(配信サーバ作れるぐらい)、苦労ポイントがひどい。 ・WebRTCは「SDP, ICE, DTLS, STUN, SRTP」等の複数プロトコルの組み合わせで、最低限でも10個以上RFCを参照する必要がある……。 ・それらの網羅的なプロトコルレベルの解説が(RFC含め)皆無。」 / Twitter]], [TIME[午後11:58 · 2023年4月12日][2023-04-12T14:58:49.000Z]], [TIME[2023-04-15T13:22:29.000Z]] <https://twitter.com/nabe_abk/status/1646165984024068097>
--[67] [CITE@ja[nabe@abkさんはTwitterを使っています: 「進捗は、SRTP(セキュアなRTP)を送る手前まで前までたどり着いたところ。 そりゃWebRTCを、きちんと解説してる人なんか(ほぼ)いないよな……。だいたい、WebRTCがプロトコルレベルで分かっている人がこの世に超少ないと思われ。」 / Twitter]], [TIME[4月12日][2023-04-12T14:58:49.000Z]], [TIME[2023-04-15T13:22:29.000Z]] <https://twitter.com/nabe_abk/status/1646167601528045570>