WebRTC 是一套基于 Web 的实时通信解决方案。

其官网上的解释如下：
借助WebRTC，您可以在基于开放标准的应用程序中添加实时通信功能。 它支持在端点之间发送视频，语音和一般的数据信息，从而使开发人员能够构建功能强大的语音和视频通信解决方案。
该技术可在所有现代浏览器以及所有主要平台的本机客户端上使用。 WebRTC背后的技术被实现为一个开放的Web标准，并在所有主要浏览器中均以常规JavaScript API的形式提供。 对于本机客户端（例如Android和iOS应用程序），可以使用提供相同功能的库。

WebRTC 开源项目的源码在此: https://webrtc.googlesource.com/src
，上百万行的代码汗牛充栋，直接看代码会迷失在汪洋大海中。

我们还要先理解标准和协议, 知道背后的设计思想

WebRTC 相关标准

• WebRTC standard: https://www.w3.org/TR/webrtc

WebRTC 相关协议

• SDP: http://www.rfcreader.com/#rfc4566
• RTP: https://www.rfcreader.com/#rfc3550
• SRTP: https://www.rfcreader.com/#rfc3711
• RTP Profile: https://www.rfcreader.com/#rfc3551
• Datagram Transport Layer Security Version 1.2: https://www.rfcreader.com/#rfc6347
• RTCWeb Offer/Answer Protocol (ROAP): https://tools.ietf.org/html/draft-jennings-rtcweb-signaling-01
• Javascript Session Establishment Protocol (JSEP): https://tools.ietf.org/html/rfc8829
• Session Traversal Utilities for NAT (STUN): https://tools.ietf.org/html/rfc5389
• Traversal Using Relays around NAT (TURN): https://tools.ietf.org/html/rfc5766
• Interactive Connectivity Establishment (ICE): https://tools.ietf.org/html/rfc8445
• TCP Candidates with Interactive Connectivity Establishment (ICE): https://tools.ietf.org/html/rfc6544
• Trickling ICE: https://tools.ietf.org/html/draft-ivov-mmusic-trickle-ice-sip-02
• Datagram Transport Layer Security for SRTP (DTLS-SRTP): https://www.rfcreader.com/#rfc5764
• Connection-Oriented Media Transport over TLS in SDP: https://www.rfcreader.com/#rfc4572
• TCP-Based Media Transport in SDP: https://www.rfcreader.com/#rfc4145
• Web Real-Time Communication (WebRTC): Media Transport and Use of RTP: https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-26
• Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF): https://tools.ietf.org/html/rfc5104
• Extended RTP Profile for RTCP-Based Feedback (RTP/AVPF): https://tools.ietf.org/html/rfc4585
• REMB - RTCP message for Receiver Estimated Maximum Bitrate: https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03
• Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF): https://tools.ietf.org/html/rfc5104
• A Google Congestion Control Algorithm for Real-Time Communication: https://tools.ietf.org/html/draft-ietf-rmcat-gcc-02
• Framing RTP and RTCP Packets over Connection-Oriented Transport: https://tools.ietf.org/html/rfc4571
• SSRC Attributes in SDP: https://tools.ietf.org/html/rfc5576
• (RTP) Header Extension for Client-to-Mixer Audio Level Indication: https://tools.ietf.org/html/rfc6464
• RTP Retransmission Payload Format: https://tools.ietf.org/html/rfc4588
• Negotiating Media Multiplexing Using SDP: https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-54
• RTP Stream Identifier Source Description (SDES): https://tools.ietf.org/html/draft-ietf-avtext-rid-09
• WebRTC MediaStream Identification in SDP: https://tools.ietf.org/id/draft-ietf-mmusic-msid-05.html
• RTP Extensions for Transport-wide Congestion Control: https://tools.ietf.org/html/draft-ietf-avtext-rid-09
• RTP Header Extension for the RTCP Source Description Items: https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01
• A Framework for SDP Attributes when Multiplexing: https://tools.ietf.org/html/draft-ietf-mmusic-sdp-mux-attributes-19
• ULPFEC - RTP Payload Format for Generic Forward Error Correction: https://tools.ietf.org/html/rfc5109
• RED - RTP Payload for Redundant Audio Data: https://tools.ietf.org/html/rfc2198
• RTP Payload Format for H.264 Video: https://tools.ietf.org/html/rfc6184
• RTP Payload Format for Scalable Video Coding: https://tools.ietf.org/html/rfc6190
• Definition of the Opus Audio Codec: https://tools.ietf.org/html/rfc6716

参考资料

• WebRTC offical site: https://webrtc.org/

• WebRTC API

  • https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
  • https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection
  • https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel

• WebRTC tutorial and book

  • https://www.html5rocks.com/en/tutorials/webrtc/basics/
  • https://www.html5rocks.com/en/tutorials/webrtc/infrastructure/
  • https://a-wing.github.io/webrtc-book-cn/01_introduction.html

• WebRTC native codes

  • https://webrtc.googlesource.com/src/+/refs/heads/master/docs/native-code/index.md
  • https://webrtc.googlesource.com/src/+/refs/heads/master/docs/native-code/development/index.md