Tencent-RTC / obs-trtcLinks
Demo for using OBS WHIP to publish a stream to the TRTC(Tencent Real-Time Communication) service.
☆11Updated 3 months ago
Alternatives and similar repositories for obs-trtc
Users that are interested in obs-trtc are comparing it to the libraries listed below
Sorting:
- a videoconferencing SFU plugin of Janus, which can mix all AV streams and re-publish it to another media server. But now, it's under deve…☆12Updated 6 years ago
- Utility to process H264 profile-level-id values☆53Updated 3 months ago
- play WebCodecs☆86Updated 3 years ago
- Multiparty-meeting (mediasoup) SIP gateway using Kurento☆28Updated 6 years ago
- WebRTC to RTMP and RTMP to WebRTC☆36Updated 5 years ago
- A demo application for testing mediasoup-go☆32Updated last month
- ☆60Updated 5 years ago
- ffmpeg build script, lal website document, rfc document.☆57Updated 10 months ago
- An implementation of RTP Payload Format for Flexible Forward Error Correction (FEC) - draft 11.☆34Updated 5 years ago
- webrtc insertable stream play☆15Updated 4 years ago
- Implment WebRTC H264 encoder by calling OBS's internal encoder in order to use x264 and some hardware H264 encoders for 1080P acceleratio…☆54Updated 6 years ago
- A webcodec flv websocket player☆21Updated 2 years ago
- Simplest possible plugin for Janus. Intended as a instructional sample for plugin builders.☆33Updated 4 years ago
- ☆27Updated last year
- 剥离掉音视频和其他无用模块的WebRTC库,用于P2P二进制文件传输,已应用于cdnbye移动端SDK。☆42Updated 4 years ago
- Yang Real-Time Communication,功能强大的视音频开发SDK。☆37Updated 2 years ago
- RTMP edge speed with janus-gateway☆47Updated 4 years ago
- webcodec for webOBS☆3Updated 2 years ago
- WebRTC auto build scripts for windows/mac/linux/iOS/Android☆20Updated 2 years ago
- ☆16Updated 3 years ago
- implement of mediasoup-worker with golang☆10Updated 10 months ago
- a rtmp server similar with srs but wrote by golang☆38Updated 4 years ago
- ☆34Updated 5 years ago
- A simple FLV file parser, pure C++ implementation☆10Updated 3 years ago
- Live media streaming. High performance Http, secure websocket and webrtc server. Supports H264, Opus and Mp3.☆16Updated last year
- JS Library to estimate the Mean Opinion Score (MOS) for Real Time audio & video communications☆35Updated 2 weeks ago
- ☆11Updated 7 months ago
- 使用srs_librtmp和RawQuic通过QUIC推RTMP流,实现RTMP OVER QUIC。☆94Updated 5 years ago
- copy from 金山云☆17Updated 6 years ago
- WebRTC native C/C++ sdk api based release M67, just keep WebRTC's audio/video en/decode and transfer.☆22Updated 5 years ago