OpenVisualCloud / Video-Conferencing-SampleLinks
Welcome to the GitHub repo for Open Visual Cloud video conferencing sample. This sample is built on top of Open WebRTC Toolkit media server and client SDK. To see the detailed information of Open WebRTC Toolkit, visit GitHub page https://github.com/open-webrtc-toolkit
☆55Updated last year
Alternatives and similar repositories for Video-Conferencing-Sample
Users that are interested in Video-Conferencing-Sample are comparing it to the libraries listed below
Sorting:
- The Symphony Media Bridge (SMB) is a media server application that handles audio, video and screen sharing media streams in an RTC confer…☆61Updated 3 months ago
- mediasoup broadcaster demo (libmediasoupclient)☆112Updated last year
- This Immersive Video project includes 2 samples which are based on OMAF and WebRTC streaming frameworks.☆135Updated 7 months ago
- Workarounds to use external H.264 video encoders in WebRTC Native C++ source code☆127Updated 7 years ago
- webrtcwork.com - Resources for those working with WebRTC (Jobs, Contractors and Tool)☆75Updated 5 months ago
- demo: webrtc to rtmp via kurento☆204Updated 3 years ago
- Simple useful interoperability tests for WebRTC libraries. If you are a WebRTC library developer we'd love to include you!☆197Updated last month
- Open WebRTC Toolkit JavaScript SDK☆209Updated 8 months ago
- A future proof, experimental WebRTC VP9 SVC SFU wit end to end encryption support☆223Updated 3 years ago
- Microservice to monitor and analyze WebRTC stacks☆36Updated 2 years ago
- Plugin for Janus forwarding RTP and RTCP packets to an external UDP receiver/decoder, e.g. a GStreamer pipeline☆87Updated last year
- Snowem is a lightweight live streaming server, based on webrtc technology. Its design mainly focuses on simplicity, scalability and high …☆87Updated 5 years ago
- rtmp to webrtc☆215Updated 2 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
- C++ based WebRTSP implementation☆37Updated 5 months ago
- Mediasoup WebRTC vanilla JS broadcast example.☆122Updated 2 years ago
- tiny/fast webRTC video conferencing gateway☆90Updated 2 weeks ago
- This repo contains the upstream webrtc stack code, with updates for Open WebRTC Toolkit.☆263Updated 8 months ago
- webrtc streamer based on gstreamer☆76Updated 6 years ago
- Open WebRTC Toolkit client SDK for native Windows/Linux/iOS applications.☆404Updated 8 months ago
- [ARCHIVED] Contents migrated to monorepo: https://github.com/Kurento/kurento☆68Updated 2 years ago
- RTMP edge speed with janus-gateway☆47Updated 4 years ago
- QrLipsync is an audio-video latency (also referred to as lipsync) measurement and validation tool.☆53Updated 3 months ago
- mediasoup client side C++ library☆314Updated 7 months ago
- Ubuntu-based container images with upstream GStreamer pre-installed☆137Updated last month
- Simple Record Demo using Mediasoup 3 and GStreamer☆209Updated 2 years ago
- srt encoder☆99Updated 2 years ago
- Recording audio/video streams from WebRTC using Medooze Media Server and GStreamer or FFmpeg☆25Updated last year
- Support WebRTC(WHIP) for FFmpeg.☆153Updated 2 weeks ago
- C++ server and client APIs for WebTransport.☆115Updated 8 months ago