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https://github.com/shinyoshiaki/werift-webrtc
WebRTC Implementation for TypeScript (Node.js), includes ICE/DTLS/SCTP/RTP/SRTP/WEBM/MP4
https://github.com/shinyoshiaki/werift-webrtc
dtls ice nodejs rtp sctp srtp typescript webm webrtc
Last synced: about 2 months ago
JSON representation
WebRTC Implementation for TypeScript (Node.js), includes ICE/DTLS/SCTP/RTP/SRTP/WEBM/MP4
- Host: GitHub
- URL: https://github.com/shinyoshiaki/werift-webrtc
- Owner: shinyoshiaki
- License: mit
- Created: 2020-02-16T07:37:24.000Z (over 4 years ago)
- Default Branch: develop
- Last Pushed: 2024-08-01T03:31:15.000Z (about 2 months ago)
- Last Synced: 2024-08-01T05:55:47.160Z (about 2 months ago)
- Topics: dtls, ice, nodejs, rtp, sctp, srtp, typescript, webm, webrtc
- Language: TypeScript
- Homepage:
- Size: 23 MB
- Stars: 458
- Watchers: 19
- Forks: 28
- Open Issues: 19
-
Metadata Files:
- Readme: README.md
- License: LICENSE
Awesome Lists containing this project
- awesome-webrtc - werift-webrtc - WebRTC Implementation for TypeScript (Node.js), includes ICE/DTLS/SCTP/RTP/SRTP/WEBM/MP4. (Node)
README
# werift
werift (**We**b**r**tc **I**mplementation **f**or **T**ypeScript)
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](https://shinyoshiaki.github.io/werift-webrtc/website/build/)
- [API Reference](https://shinyoshiaki.github.io/werift-webrtc/website/build/docs/api)
# examples
https://github.com/shinyoshiaki/werift-webrtc/tree/master/examples
### SFU
https://github.com/shinyoshiaki/node-sfu
# demo
## MediaChannel
```sh
yarn media
```
open
https://shinyoshiaki.github.io/werift-webrtc/examples/mediachannel/pubsub/answer
see console & chrome://webrtc-internals/
## DataChannel
run
```sh
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](https://github.com/web-platform-tests/wpt)
- [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 ....