werift

werift (Webrtc Implementation for TypeScript)

werift is a WebRTC Implementation for TypeScript (Node.js), includes ICE/DTLS/SCTP/RTP.

install

npm install werift

requires at least Node.js 16

Documentation (WIP)

• Website
• API Reference

examples

https://github.com/shinyoshiaki/werift-webrtc/tree/master/examples

SFU

https://github.com/shinyoshiaki/node-sfu

demo

MediaChannel

yarn media

open https://shinyoshiaki.github.io/werift-webrtc/examples/mediachannel/pubsub/answer

see console & chrome://webrtc-internals/

DataChannel

run

yarn datachannel

open https://shinyoshiaki.github.io/werift-webrtc/examples/datachannel/answer

see console & chrome://webrtc-internals/

RoadMap

Work in Progress Towards 1.0

• [x] STUN
• [x] TURN
  • [x] UDP
• [x] ICE
  • [x] Vanilla ICE
  • [x] Trickle ICE
  • [x] ICE-Lite Client Side
  • [ ] ICE-Lite Server Side
• [x] DTLS
  • [x] DTLS-SRTP
  • [x] Curve25519
  • [x] P-256
• [x] DataChannel
• [x] MediaChannel
  • [x] sendonly
  • [x] recvonly
  • [x] sendrecv
  • [x] multi track
  • [x] RTX
  • [x] RED
• [x] RTP
  • [x] RFC 3550
  • [x] Parse RTP Payload Format for VP8 Video
  • [x] Parse RTP Payload Format for VP9 Video
  • [x] Parse RTP Payload Format for H264 Video
  • [x] Parse RTP Payload Format for AV1 Video
  • [x] RED (RFC 2198)
• [x] RTCP
  • [x] SR/RR
  • [x] Picture Loss Indication
  • [x] ReceiverEstimatedMaxBitrate
  • [x] GenericNack
  • [x] TransportWideCC
• [x] SRTP
• [x] SRTCP
• [x] SDP
• [x] PeerConnection
• [x] Simulcast
  • [x] recv
• [x] BWE
  • [x] sender side BWE
• [ ] Documentation
• [x] Compatibility
  • [x] Chrome
  • [x] Safari
  • [x] FireFox
  • [x] Pion
  • [x] aiortc
  • [x] sipsorcery
  • [x] webrtc-rs
• [x] Interop E2E test
  • [x] Chrome
  • ↓↓↓ https://github.com/sipsorcery/webrtc-echoes
  • [x] Pion
  • [x] aiortc
  • [x] sipsorcery
  • [x] webrtc-rs
• [ ] Unit Tests
  • [ ] follow Web Platform Tests
• [x] MediaRecorder
  • [x] OPUS
  • [x] VP8
  • [x] H264
  • [x] VP9
  • [x] AV1

Road Map Towards 2.0

• [ ] API compatible with browser RTCPeerConnection
• [ ] ICE
  • [ ] ICE restart
• [ ] SDP
  • [ ] reuse inactive m-line
• [ ] Simulcast
  • [ ] send
• [ ] support more cipher suites
• [ ] getStats
• [ ] TURN
  • [ ] TCP

reference

• aiortc https://github.com/aiortc/aiortc
• pion/webrtc https://github.com/pion/webrtc
• etc ....