ultravideo / uvgComm
High performance P2P-Mesh video conferencing
☆54Updated 4 months ago
Alternatives and similar repositories for uvgComm:
Users that are interested in uvgComm are comparing it to the libraries listed below
- A C++ library for multiplexing and de-multiplexing MPEG-TS☆18Updated 7 months ago
- CMake script for retrieving, building and linking Google's WebRTC into a single static library.☆32Updated 2 years ago
- Secure Real-Time Media Flow Protocol (RTMFP) Library in C++☆42Updated 3 weeks ago
- RTSP video streaming server implementation based on Live555 and FFmpeg☆43Updated 4 years ago
- WebRTC C++ library built on top of chromium webrtc.☆108Updated 6 years ago
- C++ SDP library with ABNF strict parsing☆42Updated 2 years ago
- A fast, cross-platform and modern C++ SDK for all your MPEG-2 transport stream media format needs following international specification I…☆73Updated 3 months ago
- ☆45Updated 2 months ago
- mix multi audio stream and multi video stream to one audio stream and one video stream and send it to rtmp server.☆17Updated 8 years ago
- A build script of WebRTC library from Chromium.☆39Updated last year
- mediasoup c++ server (SFU)☆14Updated 7 months ago
- a video engine include receiver and sender base on webrtc☆12Updated 7 years ago
- Media Management System: ingestion, processing, encoding, delivery, ...☆38Updated this week
- the webrtc client for the janus webrtc gateway☆37Updated 5 years ago
- Very basic sketch of rendering YUV frames via Qt/OpenGL☆56Updated 2 years ago
- webrtc build with qt☆43Updated 10 years ago
- C++ based WebRTSP implementation☆36Updated last week
- ☆45Updated 5 years ago
- SRT server base ZLToolKit for intergrate to ZLM☆10Updated 2 years ago
- VS2015 webrtc bulid solution, now open source for everyone.☆14Updated last year
- Simple C++ wrapper of the SRT protocol for building Server/Client transport solutions☆20Updated 9 months ago
- A RTSP server for streaming combined picture from multiple cameras based on FFMPEG and live555☆64Updated 6 years ago
- complete SIP signalling and RTP media service for rapid development of voice/video services and softphones☆33Updated last month
- Baresip WebRTC Demo - moved to baresip☆46Updated 2 years ago
- media sdk based on webrtc☆40Updated 2 years ago
- WebRTC precompiled builds for Linux and Windows.☆48Updated 6 years ago
- Asynchronous Audio / Video Library for H264 / MJPEG / OPUS / AAC / MP2 encoding, transcoding, recording and streaming from live sources☆63Updated 2 years ago
- WebRTC with hardware accelerated video encoding.☆98Updated 3 years ago
- Clone of https://webrtc.googlesource.com/src with Mozilla's local modifications☆44Updated 5 months ago
- An implementation of RTP Payload Format for Flexible Forward Error Correction (FEC) - draft 11.☆34Updated 5 years ago