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
• 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
  • 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

• WHIP
  • WebRTC 推流
  • I-D draft-ietf-wish-whip-01
    • 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
• 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
• caniuse.com/webtransport
  • Chrome 97+

参考/References

• 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
• stun.l.google.com
• stun1.l.google.com
• stun2.l.google.com
• stun3.l.google.com
• stun4.l.google.com
• 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 gist.github.com/zziuni/3741933

turn

• turn2.l.google.com