WebRTC Awesome

• webrtc/samples WebRTC Web demos and samples
• webrtc/apprtc
  • appr.tc
• rtctunnel/rtctunnel network tunnels over WebRTC
• jitsi/jitsi-meet Secure, Simple and Scalable Video Conferences
• obsproject/obs-studio video recording and live streaming
• peers/peerjs
  • MIT, Typescript
• peers/peerjs-server
• livekit/livekit-server Distributed audio/video rooms over WebRTC
  • Apache-2.0, Go
• WebRTC-HTTP ingestion protocol
  • WHIP
• glimesh/broadcast-box
  • supported by
    • OBS, ffmpeg
• AlexxIT/go2rtc
• https://github.com/Yahweasel/libav.js/
• node-webrtc/node-webrtc
  • WebRTC 的 Node 绑定
  • yarn add wrtc
  • Node 支持 armv7l、arm64、x64
  • Electron 支持 x64
• TURN/STUN Server
  • coturn/coturn
    • BSD-3, C TURN/STUN Server
  • pion/turn Golang, TURN Server/Client
• Golang
  • pion Golang, MIT, WebRTC
    • pion/ion Distributed RTC System by pure Go and Flutter
    • pion/dtls Golang, DTLS 1.2 Server/Client
    • pions/webrtc
      • MIT, Golang
      • WebRTC API
  • keroserene/go-webrtc
• Gateway
  • meetecho/janus-gateway
    • GPL-3.0, C
• C/C++
  • paullouisageneau/libdatachannel
    • MPL-2.0, C++
    • SSL - GnuTLS, Mbed TLS, OpenSSL
    • usrsctp
    • Plog
    • libjuice
    • libsrtp
    • nlohmann/json
    • WASM
• NodeJS
  • shinyoshiaki/werift-webrtc
    • TS 实现
    • ICE/DTLS/SCTP/RTP
  • murat-dogan/node-datachannel
    • binding libdatachannel
  • node-webrtc/node-webrtc
• Spec
  • https://w3c.github.io/p2p-webtransport/
• Ecosystem
  • bitwhip/bitwhip
    • MIT, Rust
    • CLI Native WebRTC Agent in Rust
  • Glimesh/broadcast-box
    • MIT, Go
  • obsproject/obs-studio#7926
    • OBS 支持 WebRTC 输出
  • ffmpeg 支持 WebRTC - WHIP

abbr.	stand for	meaning
DTLS	Datagram Transport Layer Security	数据报传输层安全协议
ICE	Interactive Connectivity Establishment	交互式连接建立
RTC	Real-Time Communication	实时通信
SCTP	Stream Control Transmission Protocol	流控制传输协议
SDP	Session Description Protocol	会话描述协议
SRTP	Secure Real-time Transport Protocol	安全实时传输协议
STUN	Session Traversal Utilities for NAT	NAT 会话遍历实用工具
SVC	Scalable Video Coding	可伸缩视频编码
TURN	Traversal Using Relays around NAT	NAT 中继穿越
WHIP	WebRTC HTTP Ingestion Protocol	WebRTC HTTP 输入协议

• WHIP
  • WebRTC 推流
  • https://www.ietf.org/archive/id/draft-ietf-wish-whip-01.html
    • 使用一个 HTTP POST 请求来完成一次性的 SDP (Session Description Protocol) 要约/应答过程
    • 在编码器/媒体生产者（WHIP 客户端）与广播摄取端点（媒体服务器）之间建立一个 ICE/DTLS 会话
    • 会话一旦建立，媒体流就会单向地从客户端流向服务器。
• Simulcast 同播
  • 同一个媒体源（例如，同一个摄像头或麦克风）发送多个不同编码版本的媒体流，并通过不同的 RTP 流进行传输
  • RFC 8853
    • 《在会话描述协议 (SDP) 和 RTP 会话中使用 Simulcast》

WebCodecs

• VideoEncoder, VideoDecoder, AudioWorkletm
• requestAnimationFrame
• https://caniuse.com/webcodecs
  • Chrome 94+

WebTransport

• 加密、拥塞控制、独立流、1RTT、UDP 端口复用、透明网络迁移
• HTTP/3
• https://caniuse.com/webtransport
  • Chrome 97+

参考/References

• https://quic.video/blog/replacing-webrtc/
• Explaining the WebRTC Secure Real-Time Transport Protocol (SRTP)
• A Study of WebRTC Security
• WebRTC 安全性的研究

stun

• stun.stunprotocol.org:3478
• stun.l.google.com:19302
• stun1.l.google.com:19302
• stun2.l.google.com:19302
• stun3.l.google.com:19302
• stun4.l.google.com:19302
• stun01.sipphone.com
• stun.ekiga.net
• stun.fwdnet.net
• stun.ideasip.com
• stun.iptel.org
• stun.rixtelecom.se
• stun.schlund.de
• stunserver.org
• stun.softjoys.com
• stun.voiparound.com
• stun.voipbuster.com
• stun.voipstunt.com
• stun.voxgratia.org
• stun.xten.com
• stun.fwdnet.net
• stun.voxgratia.org
• stun.xten.com
• s1.taraba.net
• s2.taraba.net
• s1.voipstation.jp
• s2.voipstation.jp
• numb.viagenie.ca
• stun.counterpath.com

---

• stuns https://gist.github.com/zziuni/3741933

turn

• turn2.l.google.com