wesley-fly / wtkrtc-sdk-api
WebRTC native C/C++ sdk api based release M67, just keep WebRTC's audio/video en/decode and transfer.
☆22Updated 5 years ago
Alternatives and similar repositories for wtkrtc-sdk-api
Users that are interested in wtkrtc-sdk-api are comparing it to the libraries listed below
Sorting:
- media sdk based on webrtc☆40Updated 2 years ago
- a video engine include receiver and sender base on webrtc☆12Updated 8 years ago
- The full C++ implementation of mediasoup☆33Updated 4 years ago
- LiveStream gateway for WebRTC 流媒体网关服务器☆25Updated 10 months ago
- Video Jitter Buffer derived from WebRTC☆16Updated 6 years ago
- webrtc network part for reliable udp transport☆31Updated 4 years ago
- SRT server base ZLToolKit for intergrate to ZLM☆10Updated 2 years ago
- the webrtc client for the janus webrtc gateway☆36Updated 6 years ago
- This project can be used to conduct audio call. 本程序可以实现语音通话。☆48Updated 3 years ago
- This project can be used to conduct loopback video call, you can also choose to use OPENH264 or X264 as video codec besides VP8. 本程序可以实现环…☆67Updated 3 years ago
- 使用srs_librtmp和RawQuic通过QUIC推RTMP流，实现RTMP OVER QUIC。☆94Updated 5 years ago
- 基于Chromium 82.0.4061.2 quiche库QuicTransport协议实现的原生QUIC客户端库，去掉HTTP2/SPDY，封装了类Socket接口，可以收发裸QUIC数据。☆67Updated 4 years ago
- ☆18Updated 11 years ago
- 一个移植于WebRtc项目的通用基础lib库☆26Updated 6 years ago
- a gb28181 sip sdk based on resiprocate☆30Updated 6 years ago
- C++ SDP library with ABNF strict parsing☆42Updated 2 years ago
- An implementation of RTP Payload Format for Flexible Forward Error Correction (FEC) - draft 11.☆33Updated 5 years ago
- example for how to use webrtc native c++ api☆23Updated 8 years ago
- adaptive jitter buffer implementatioin☆32Updated 4 years ago
- Yang Real-Time Communication，功能强大的视音频开发SDK。☆37Updated 2 years ago
- A demo application for testing mediasoup-go☆32Updated last week
- get vedio&audio ES from RTP packet over UDP from GBT28181 IPC☆31Updated 9 years ago
- 一个程序员的学习笔记☆22Updated 2 years ago
- cms is an industrial-strength live streaming server,support rtmp,http-flv,hls.in the future,it will support more protocol.☆12Updated 2 years ago
- FFmpeg, live555 already support RTSP, SDP, RTP, RTCP.☆16Updated last year
- Live media streaming. High performance Http, secure websocket and webrtc server. Supports H264, Opus and Mp3.☆16Updated last year
- ☆30Updated 2 years ago
- SIP UserAgent(UAS and UAC) Sample☆38Updated 8 years ago
- WebRTC auto build scripts for windows/mac/linux/iOS/Android☆20Updated 2 years ago
- ☆40Updated 6 years ago