一、官方資源
- 官方網站:http://webrtc.org(官網還是最權威的)
- 2013谷歌I/O大會對WebRTC的介紹:視頻(https://www.youtube.com/watch?v=p2HzZkd2A40),ppt(http://io13webrtc.appspot.com/#1)(講的不錯)
- 2012谷歌I/O大會對WebRTC的介紹:http://youtu.be/E8C8ouiXHHk
- WebRTC官方源碼樣例(不含移動端):http://github.com/webrtc/samples (看再多理論不如摳一遍源碼)
- WebRTC在線演示效果:http://webrtc.github.io/samples(可以清楚的看到每個接口是怎樣被調用的)
二、初學者入門
- 官方推薦的入門文章:http://html5rocks.com/en/tutorials/webrtc/basics(個人感覺講的有點繞,英文不好估計很難理解)
- 使用WebRTC搭建前端視頻聊天室——入門篇:http://chinawebrtc.org/?p=271(推薦這篇中文的入門,講的很細,它的三篇後續教程也很值得一看)
- WebRTC體系結構:http://chinawebrtc.org/?p=338(對整體的把握是很重要的)
- 通過WebRTC實現實時視頻通信:http://chinawebrtc.org/?p=462 (不錯的教程)
- 官方編譯教程:(理論後,開始實踐)
- 看看大牛的編譯實踐:
- 使用Tokbox瞬間實現在線視頻:https://dashboard.tokbox.com/quickstart#1(需要註冊申請一個sdk的key生成token,之後就很方便了)
- 國外已經有視頻教程了:http://www.pluralsight.com/courses/webrtc-fundamentals(可試看,後需會員)
- WebRTC在android端的教程:https://tech.appear.in/2015/05/25/Introduction-to-WebRTC-on-Android/
- WebRTC在iOS端的教程:https://tech.appear.in/2015/05/25/Getting-started-with-WebRTC-on-iOS/
- Play With WebRTC:http://chinawebrtc.org/?p=530
- 手把手教程:
三、高級教程
- getUserMedia解釋:http://www.html5rocks.com/en/tutorials/getusermedia/intro/
- 信令機制的解釋:http://www.html5rocks.com/en/tutorials/webrtc/infrastructure/
- 使用WebRTC搭建前端視頻聊天室——信令篇:http://chinawebrtc.org/?p=260
- 使用WebRTC搭建前端視頻聊天室——點對點通信篇:http://chinawebrtc.org/?p=273
- 使用WebRTC搭建前端視頻聊天室——數據通道篇:http://chinawebrtc.org/?p=274
- WebRTC音視頻引擎研究(1)–整體架構分析:http://chinawebrtc.org/?p=355
- WebRTC音視頻引擎研究(2)–VoiceEngine音頻編解碼器數據結構以及參數設置:http://chinawebrtc.org/?p=356
- WebRTC Native APIs[翻譯]:http://chinawebrtc.org/?p=357
- WebRTC源碼分析1——視頻顯示:http://chinawebrtc.org/?p=360
- WebRTC源碼分析2——圖像縮放與顏色空間轉換:http://chinawebrtc.org/?p=365
- WebRTC源碼分析3——jpeg編解碼:http://chinawebrtc.org/?p=366
- WebRTC源碼分析4——AVI文件讀寫:http://chinawebrtc.org/?p=371
- WebRTC源碼分析5——VoiceEngine代碼解析:http://chinawebrtc.org/?p=380
- WebRTC源碼分析6——音頻模塊結構分析:http://chinawebrtc.org/?p=379
- WebRTC源碼分析6——AudioProcessing的使用:http://chinawebrtc.org/?p=381
- webrtc 的回聲抵消(aec、aecm)算法簡介:http://chinawebrtc.org/?p=382
- 建立一個WebRtc的Android客戶端:http://chinawebrtc.org/?p=260
- WebRtc常見問題集錦:http://chinawebrtc.org/?p=327
四、源碼或示例
- 這裏面應該是最全最詳細的了:https://www.webrtc-experiment.com/
- 這裏面也有不少:http://simpl.info/
-
getUserMedia:
- ASCII碼的視頻(getUserMedia + Canvas + ASCII conversion):http://idevelop.ro/ascii-camera/
- 各種酷炫效果,還能這麼玩居然(getUserMedia + WebGL):http://webcamtoy.com
- svg濾鏡https://rawgit.com/SenorBlanco/moggy/master/filterbooth.html
- WebGl實現人臉面具:http://auduno.github.io/clmtrackr/examples/facedeform.html
- 用臉玩太空大戰:http://shinydemos.com/facekat
- 一個錄音顯示聲紋波動的demo:http://webaudiodemos.appspot.com/AudioRecorder
- 音頻Demo大集合:http://webaudiodemos.appspot.com/
- gUM + WebGL實現錄音室:http://lab.aerotwist.com/webgl/audio-room
-
RTCDataChannel
- 一個簡單的例子:http://simpl.info/dc
- 文件分享:https://sharefest.me/
- 一個js類庫:http://ozan.io/p/
- 實時通信的TogetherJS 類庫:https://togetherjs.com/
- 用WebRTC實現BitTorrent:https://github.com/feross/webtorrent
-
RTCPeerConnection
- 一個簡單的例子:http://simpl.info/pc
- 視頻聊天示例:https://apprtc.appspot.com/,源碼https://code.google.com/p/webrtc/source/browse/#svn%2Ftrunk%2Fsamples%2Fjs%2Fapprtc
- 視頻聊天示例:https://appear.in/,開發者api:https://developer.appear.in/
- https://bistri.com/
- 視頻聊天示例:https://talky.io/,源碼:https://github.com/henrikjoreteg/SimpleWebRTC
- 視頻聊天示例:https://tawk.com/
- 通過github視頻聊天:https://gittogether.com/
- 視頻聊天示例:http://codassium.com/
- 視屏聊天示例:https://vline.com/
- 視頻聊天示例:https://www.lytespark.com/
- 視頻聊天示例:https://vidtok.com/
- 視頻聊天示例:http://www.easyrtc.com/,源碼https://github.com/priologic/easyrtc
- 視頻聊天示例(印度的):https://www.miljul.in/
- http://chotis2.dit.upm.es/(可fork on GitHub)
- https://janus.conf.meetecho.com/(可fork on GitHub)
- goToMeeting在線版:https://free.gotomeeting.com/
- 嬰兒監視器:https://webrtchacks.com/baby-motion-detector/
- 電話通訊:http://zingaya.com/
五、一些api及類庫
- 官方的PeerConnection的api:http://www.webrtc.org/blog/api-description
- 官方其它的一些的api:http://www.webrtc.org/native-code/native-apis
- libjingle的文檔介紹https://developers.google.com/talk/libjingle/developer_guide?csw=1
- getUserMedia.js:https://github.com/addyosmani/getUserMedia.js
- adapter.js:https://github.com/webrtc/adapter/blob/master/adapter.js
- WebRTC的js類庫裏有些什麼:https://webrtchacks.com/whats-in-a-webrtc-javascript-library/
- Web Audio API:http://webaudio.github.io/web-audio-api/
- The PeerJS library:簡化了WebRTC傳輸數據的過程http://peerjs.com/
- 有關瀏覽器通話的js類庫:http://phono.com/
- 封裝SIP協議的js類庫:客戶端,https://code.google.com/p/sipml5/;http://jssip.net/
- 面部識別的js類庫:https://github.com/auduno/clmtrackr
- 頭部軌跡識別的js類庫:https://github.com/auduno/headtrackr/;demo,http://simpl.info/headtrackr/
- http://rtc.io/
- 開發WebRTC的工具列表(不能更全):https://webrtchacks.com/vendor-directory/
六、一些書籍
- WebRTC-APIs and RTCWEB Protocols of the HTML5 Real-Time Web, Third Edition:http://webrtcbook.com/
- Real-Time Communication with WebRTC by Salvatore Loreto & Simon Pietro Romano:https://bloggeek.me/book-webrtc-salvatore-simon/
- Getting Started with WebRTC:https://www.packtpub.com/web-development/getting-started-webrtc
七、標準及協議
- WebRTC工作小組:http://www.w3.org/2011/04/webrtc/
- w3c規定的WebRTC協議1.0http://www.w3.org/TR/webrtc/
- 媒體捕捉及媒體流協議:http://www.w3.org/TR/mediacapture-streams/
- IETF協議http://datatracker.ietf.org/wg/rtcweb/documents/
- 各大瀏覽器是否支持:http://iswebrtcreadyyet.com/
八、其它
- 國外Google group:https://groups.google.com/forum/?fromgroups#!forum/discuss-webrtc
- 國內china WebRTC社區:http://chinawebrtc.org/
九、WebRTC 1.0: Real-time Communication Between Browsers 協議文檔中文版彙總
- 第一篇是描述整個文檔的狀態和概要:http://www.iwebrtc.com/blog/webrtc-1-0-real-time-communication-between-browsers-1/
- 第二篇是整個文檔的介紹和術語:http://www.iwebrtc.com/blog/webrtc-1-0-real-time-communication-between-browsers-2/
- 第三篇從原文的4. Network Stream API開始,主要描述Network API和MediaStream接口(正式的內容從第三篇開始):http://www.iwebrtc.com/blog/webrtc-1-0-real-time-communication-between-browsers-3/
- 第四篇從原文的4.3 AudioMediaStreamTrack開始,主要描述AudioMediaStreamTrack類:http://www.iwebrtc.com/blog/webrtc-1-0-real-time-communication-between-browsers-4/
- 第五篇從原文的5.Peer-to-peer connections開始,主要描述RTCPeerConnection類。原文的第五節是整個webrtc協議的重點,RTCPeerConnection是webrtc實現的核心功能。:http://www.iwebrtc.com/blog/webrtc-1-0-real-time-communication-between-browsers-5/
- 第六篇從原文的5.1 RTCPeerConnection開始,重點描述RTCPeerConnection的屬性和方法。:http://www.iwebrtc.com/blog/webrtc-1-0-real-time-communication-between-browsers-6/
- 第七篇從原文的5.1.6 RTCPeerState Enum開始,仍然是原文的第5節的繼續。:http://www.iwebrtc.com/blog/webrtc-1-0-real-time-communication-between-browsers-7/
- 第八篇從原文的5.1.9 RTCIceServer 類型開始,講解和ICE Server交互相關的內容。:http://www.iwebrtc.com/blog/webrtc-1-0-real-time-communication-between-browsers-8/
- 第九篇從6. IANA Registrations開始,主要描述IANA Registrations相關的標準約束。:http://www.iwebrtc.com/blog/webrtc-1-0-real-time-communication-between-browsers-9/
- 第十篇從原文的7. Simple Example開始,展示了一個簡單的javascript的例子。:http://www.iwebrtc.com/blog/webrtc-1-0-real-time-communication-between-browsers-10/
- 第十一篇從原文的9. Call Flow Browser to Browser開始,描述瀏覽器到瀏覽器的呼叫建立的流程圖。(此處是重點內容):http://www.iwebrtc.com/blog/webrtc-1-0-real-time-communication-between-browsers-11/
- 第十二篇從原文的10. Call Flow Browser to MCU開始,描述瀏覽器到MCU呼叫建立的流程圖。:http://www.iwebrtc.com/blog/webrtc-1-0-real-time-communication-between-browsers-12/
- 第十三篇從原文11. Peer-to-peer Data API開始,描述創建點到點的數據傳輸通道的API。(這個很有用,可以用來傳輸語音和視頻之外的數據,比如白板、共享桌面等):http://www.iwebrtc.com/blog/webrtc-1-0-real-time-communication-between-browsers-13/
- 第十四篇從原文11.1.1 Attributes開始,接前一篇,繼續描述DataChannel的屬性等。:http://www.iwebrtc.com/blog/webrtc-1-0-real-time-communication-between-browsers-14/
- 第十五篇從原文12. Garbage collection開始,垃圾蒐集策略以及事件彙總。:http://www.iwebrtc.com/blog/webrtc-1-0-real-time-communication-between-browsers-15/
- 第十六篇從原文15. Security Considerations開始,描述安全機制、修改日誌、致謝、參考(基本上這一篇沒怎麼翻譯,大部分可以直接無視。修改日誌可以掃一眼,參考內容可以瀏覽一下)。:http://www.iwebrtc.com/blog/webrtc-1-0-real-time-communication-between-browsers-16/
十、IETF:Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 標準(譯)
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(一.介紹):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-01/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(二.基本原理):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-02/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(三.術語):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-03/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(四.核心協議):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-04/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(五.webrtc所使用RTP擴展):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-05/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(六.增強傳輸可靠性):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-06/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(七.速率控制和媒體適配):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-07/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(八.性能監控):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-08/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(九.未來擴展):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-09/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(十.信號考慮):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-10/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(十一.WebRTC API的考慮):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-11/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(十二.RTP實現):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-12/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(十三,遺留問題):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-13/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(十五,安全考慮):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-15/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(十六,致謝和參考資料):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-15-2/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(十六,致謝和參考資料):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-15-2/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(附錄A:支持的RTP拓撲圖):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-appendix-a/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(附錄A1:點對點):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-appendix-a1/
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP 中文版(附錄A2:單點多播):http://www.iwebrtc.com/blog/web-real-time-communication-webrtc-media-transport-and-use-of-rtp-appendix-a2/