amazon-kinesis-video-streams-webrtc-sdk-c - Amazon Kinesis Video Streams Webrtc SDK is for developers to install and customize realtime communication between devices and enable secure streaming of video, audio to Kinesis Video Streams

•    C

---

Create a build directory in the newly checked out repository, and execute CMake from it. mkdir -p amazon-kinesis-video-streams-webrtc-sdk-c/build; cd amazon-kinesis-video-streams-webrtc-sdk-c/build; cmake ..

webrtc webrtc-sdk kinesis-video-streams

Mirotalk - WebRTC browser-based video call, chat and screen sharing

•    Javascript

---

Mirotalk is a free browser based real-time video calls. Simple, Secure, Fast. Powered by WebRTC using google Stun and numb Turn. Mirotalk provides video quality and latency not available with traditional technology.

webrtc chatroom peer-to-peer meeting collaboration sfu webrtc-demos screensharing rtc video-call realtime-messaging conferencing video-conferencing webrtc-call webrtc-video video-chat end-to-end-encryption webrtc-signaling google-meet zoom-clone

Pion WebRTC - A pure Go implementation of the WebRTC API

•    Go

---

Pion is the Modern Stack for Web Real-Time Communication. It implements the WebRTC API. Spend more time building and less time learning a new API. Its feature include PeerConnection API support for DataChannels, Send/Receive audio and video, Renegotiation. It provides API with direct RTP/RTCP access, API also allows developer to pass their own packetizer, Opus, PCM, H264, VP8 and VP9 packetizer, Sender/Receiver Reports, Easy integration with x264, libvpx, GStreamer and ffmpeg and lot more.

audio golang streaming video webrtc p2p webassembly wasm webcam voip ortc rtp srtp webrtc-libraries webrtc-api pion pion-webrtc live-video webrtc-server

spreed-webrtc - WebRTC audio/video call and conferencing server.

•    Javascript

---

Spreed WebRTC implements a WebRTC audio/video call and conferencing server and web client. The latest source of Spreed WebRTC can be found on GitHub. If you are a user, just wanting a secure and private alternative for online communication make sure to check out the Spreedbox, providing a ready to use hardware with Spreed WebRTC included.

starrtc-android-demo - 🚀starRTC，即时通讯(IM)系统，免费IM系统（含单聊，群聊，聊天室，文件传输），免费一对一视频聊天，VOIP，语音对讲（回音消除），直播连麦，视频直播，RTSP拉流，RTMP推流，webRTC服务端，在线教育，白板，小班课，在线会议，视频会议，视频监控，局域网直连（无需服务器），兼容webRTC, 支持webRTC加速，P2P高清传输，安卓、iOS、web互通，支持门禁对讲，可视对讲，电视盒子，树莓派，海思，全志，任天堂switch，云游戏，OTT设备，物联网平台，C语言自研方案，支持二次开发成类微信，类映客等APP，✨万水千山总是情，来个star行不行✨，更多示例请访问：

•    

---

🚀starRTC，即时通讯(IM)系统，免费IM系统（含单聊，群聊，聊天室，文件传输），免费一对一视频聊天，VOIP，语音对讲（回音消除），直播连麦，视频直播，RTSP拉流，RTMP推流，webRTC服务端，在线教育，白板，小班课，在线会议，视频会议，视频监控，局域网直连（无需服务器），兼容webRTC, 支持webRTC加速，P2P高清传输，安卓、iOS、web互通，支持门禁对讲，可视对讲，电视盒子，树莓派，海思，全志，任天堂switch，云游戏，OTT设备，物联网平台，C语言自研方案，支持二次开发成类微信，类映客等APP，✨万水千山总是情，来个star行不行✨，更多示例请访问：

raspberry-pi video webrtc chatroom live free im voip janus groupchat jitsi kurento licode video-chat coturn

webtorrent-hybrid - WebTorrent (with WebRTC support in Node.js)

•    Javascript

---

In node.js, the webtorrent package only connects to normal TCP/UDP peers, not WebRTC peers. If you want to connect to all types of peers, including WebRTC peers, from node.js, you need to use this package (webtorrent-hybrid). Previous versions (0.x) of this package used wrtc, a WebRTC implementation via native extensions, the current one is based on electron-webrtc (which in turn uses electron-prebuilt) for better compatibility. It creates a hidden Electron process (which is based on Chromium, so WebRTC support is great!) and communicates with that process to enable WebRTC in Node.js. This adds a lot of overhead, so we are looking forward to using a pure JavaScript implementation, like perhaps Node-RTCPeerConnection when it's ready.

webtorrent nodejs browser torrent bittorrent hybrid webrtc bittorrent-client download mad-science streaming webrtc-data

LiveKit - Open source, distributed video/audio rooms over WebRTC

•    Go

---

LiveKit is an open source project that provides scalable, multi-user conferencing over WebRTC. It's designed to give you everything you need to build real time video/audio capabilities in your applications. It is horizontally scalable WebRTC Selective Forwarding Unit (SFU). It supports advanced features like speaker detection, simulcast, selective subscription, and moderation APIs.

sdk video webrtc media-server sfu mediaserver

janus-webrtc-gateway-docker - Perfect Docker Image for Media Streaming Expert User ( https://github

•    Dockerfile

---

This is a docker image for Janus Webrtc Gateway. Janus Gateway is still under active development phase. So, as the official docs says, some minor modification of the middleware library versions happens frequently. I try to deal with such a chage as much as I can. If you need any request about this repo, free to contact me. About the details of setup for this docker image, you should read the official docs https://janus.conf.meetecho.com/index.html carefully. I think that janus is better for webinar(web seminar), and jitsi is better for web conference system. The scalability of the current Jitsi Video Bridge(20181007) is poor because of having no local recording file(I'm not sure of this..). https://www.youtube.com/watch?v=OHHoqKCjJ0E Jitsi last-n + VP8 simulcasting has the very good performance for web conference https://jitsi.org/wp-content/uploads/2016/12/nossdav2015lastn.pdf For the video format, janus recording is per video streaming, jitsi is for mixed video conference by using chrome headlesss + ffmpeg(alsa, libxcb). From these points, janus is suitable for webinar, jitsi is for web conference. Of course, both WebRTC SFU are amazing work!! I'm using both.

docker nginx flash ffmpeg hls rtmp webrtc docker-image boringssl rtsp-server media-server media-player dash openresty libwebsockets janus nginx-rtmp janus-gateway janus-webrtc-gateway

RecordRTC.js - WebRTC JavaScript Library for Audio+Video+Screen+Canvas (2D+3D animation) Recording

•    Javasript

---

RecordRTC is WebRTC JavaScript library for audio/video as well as screen activity recording. It supports Chrome, Firefox, Opera, Android, and Microsoft Edge. Platforms: Linux, Mac and Windows.

webrtc recordrtc mediarecorder mediastreamrecorder mediarecorder-api record-audio record-video webrtc-recording record-screen video-streaming audio-streaming

webrtc - A reference gradle project that let you explore WebRTC Android in Android Studio.

•    C++

---

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose. Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

webrtc webrtc-demos webrtc-android

serverless-webrtc - A demo of using WebRTC with no signaling server.

•    Javascript

---

This is a tech demo of using WebRTC without a signaling server -- the WebRTC offer/answer exchange is performed manually by the users, for example via IM. This means that the app can run out of file:/// directly, without involving a web server. You can send text messages and files between peers.Chat is fully interoperable between all of the above (Node, Chrome, Firefox) in any combination (tested with Chrome 35 and Firefox 29).

webrtc chat

mediasoup - Cutting Edge WebRTC Video Conferencing

•    Javascript

---

mediasoup provides cutting edge WebRTC video conferencing. It is a Selective Forwarding Unit (SFU) which supports both WebRTC and plain RTP input and output. It is a Node.js module/Rust crate in server side and tiny JavaScript and C++ libraries in client side.

webrtc node-module sfu ortc server-side node video-conferencing video-chat chat instant-messaging video streaming video-streaming

holla - An abstraction over P2P video/voice/data connections using WebRTC

•    Javascript

---

Pipes a WebRTC video stream to a video element. el can be a string (id), jquery element, or dom node.

webrtc rtc calling call voip p2p data sip voice phone peer

webrtc-streamer - WebRTC streamer for V4L2 capture devices and RTSP sources

•    C++

---

This is a try to stream video sources through WebRTC using simple mechanism. It embeds a HTTP server that implements API and serve a simple HTML page that use them through AJAX.

webrtc streamer rtsp c-plus-plus v4l2 webrtc-streamer

webrtc - A pure Golang implementation of the WebRTC Native API

•    Go

---

See DESIGN.md for an overview of features and future goals. Check out the example applications to help you along your Pion WebRTC journey.

webrtc pion-webrtc ortc rtp srtp webrtc-api

glipchat - video chatroom using meteor + webrtc + react + redux

•    Javascript

---

glipchat is a real-time video chatroom application. glipchat is powered by Meteor, with React + Redux + WebRTC on the frontend. Other browsers and operating systems may not support WebRTC.

meteor electron react android webrtc cordova redux video-conferencing

wave-share - Serverless, peer-to-peer, local file sharing through sound

•    C++

---

A proof-of-concept for WebRTC signaling using sound. Works with all devices that have microphone + speakers. Runs in the browser. Nearby devices negotiate the WebRTC connection by exchanging the necessary Session Description Protocol (SDP) data via a sequence of audio tones. Upon successful negotiation, a local WebRTC connection is established between the browsers allowing data to be exchanged via LAN.

data-over-sound webrtc fsk p2p webrtc-signaling file-sharing data-transfer modem ultrasonic

starrtc-ios-demo - 🚀starRTC，免费IM（含单聊，群聊，聊天室），免费一对一视频聊天（回音消除），语音聊天，直播连麦，白板，小班课，多人会议，局域网无服务器直连，兼容webRTC, 支持webRTC加速，P2P高清传输，安卓、iOS、web互通，支持门禁可视对讲，电视盒子，树莓派，海思，全志，OTT设备，C语言自研方案，✨万水千山总是情，来个star行不行✨，更多示例请访问

•    Objective-C

---

🚀starRTC，免费IM（含单聊，群聊，聊天室），免费一对一视频聊天（回音消除），语音聊天，直播连麦，白板，小班课，多人会议，局域网无服务器直连，兼容webRTC, 支持webRTC加速，P2P高清传输，安卓、iOS、web互通，支持门禁可视对讲，电视盒子，树莓派，海思，全志，OTT设备，C语言自研方案，✨万水千山总是情，来个star行不行✨，更多示例请访问

aiortc - WebRTC and ORTC implementation for Python using asyncio

•    Python

---

aiortc is a library for Web Real-Time Communication (WebRTC) and Object Real-Time Communication (ORTC) in Python. It is built on top of asyncio, Python's standard asynchronous I/O framework. To learn more about aiortc please read the documentation.

webrtc asyncio ortc webrtc-libraries data-channel

headtrackr - Javascript library for headtracking via webcam and WebRTC/getUserMedia

•    Javascript

---

headtrackr is a javascript library for real-time face tracking and head tracking, tracking the position of a users head in relation to the computer screen, via a web camera and the webRTC/getUserMedia standard. For a demonstration see this video or try out some of the examples with a laptop that has a camera and a browser that has camera webRTC/getUserMedia support. For an overview of browsers supporting the getUserMedia standard see http://caniuse.com/stream.

webcam headtracking computer-vision facetracking