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

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.

Jitsi Meet - Secure, Simple and Scalable Video Conferences

•    Javascript

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Jitsi Meet is an open-source (Apache) WebRTC JavaScript application that uses Jitsi Videobridge to provide high quality, secure and scalable video conferences. Jitsi Meet allows very efficient collaboration. Users can stream their desktop or only some windows. It also supports shared document editing with Etherpad.

webrtc sfu video video-conferencing video-communication jitsi jitsi-meet scalable-video-conferences debian deep-video

Meething - Decentralized Real-time communications

•    Javascript

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Meething is a semi-decentralized conference bridge using modern WebRTC, GunDB and MediaSoup to produce a hybrid MESH-SFU with strong privacy, zero user data retention and powerful off-grid capabilities.

webrtc decentralized p2p meeting rtc gundb dweb videomeeting dwebrtc video-conferencing video-meeting video-chat real-time

webrtc-sdk - WebRTC Simple Calling API + Mobile SDK - A simplified approach to RTCPeerConnection for mobile and web video calling apps

•    Javascript

---

WebRTC SDK Upgraded! ES6, new camera control and 100x less code than v1. At PubNub we believe simplicity is essential for our SDK usability. We've taken a simplified approach to WebRTC Peer Connections by creating and easy-to-use SDK for developers. The ideas of simplicity should span all platforms and devices too and that's why we also support Android WebRTC mobile calling with compatibility for iOS native Objective-C based WebRTC SDK. This simple developer WebRTC SDK is powered by PubNub Data Stream Network.

webrtc webrtc-call webrtc-sdk webrtc-video webrtc-receiver webrtc-javascript-library webrtc-phone webrtc-demo webrtc-dialing

Jitsi Videobridge - WebRTC compatible video router or SFU that lets build highly scalable video conferencing infrastructure

•    Java

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Jitsi Videobridge is an XMPP server component that allows for multiuser video communication. Unlike the expensive dedicated hardware videobridges, Jitsi Videobridge does not mix the video channels into a composite video stream, but only relays the received video channels to all call participants. Therefore, while it does need to run on a server with good network bandwidth, CPU horsepower is not that critical for performance.

video-streaming video-communication xmpp xmpp-server conference webrtc

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

•    Go

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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

p2p.chat - p2p video conferencing powered by WebRTC.

•    Javascript

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⚠️ Currently Unmaintained ⚠️ This was a hobby project that has not been worked on for some time. I would like to pick this up again in the future and overhaul how the app is architected, specifically peer discovery. I advise using https://meet.jit.si if you are looking for open source, professional video conferencing. p2p.chat is a peer-to-peer video conferencing application. Think of it as an free and open source appear.in alternative.

p2p video chat

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

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

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

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

•    Dockerfile

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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

simple-peer - 📡 Simple WebRTC video/voice and data channels

•    Javascript

---

This module works in the browser with browserify.Note: If you're NOT using browserify, then use the included standalone file simplepeer.min.js. This exports a SimplePeer constructor on window.

webrtc p2p nodejs browser data-channels peer-connection data data-channel data-channel-stream peer peer-to-peer stream video voice webrtc-stream

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：

•    

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🚀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

oorja - [archived] effortless video-voice chat with realtime collaborative features

•    Javascript

---

This repository contains an alpha version, with minimal components to make things work. It was made while I was in university(2017), I'd do a lot of things differently now, particularly in terms of code organization and overall architecture. Never the less, it was super fun to make back then. I'm still keeping this repo up, up because it feels good to look back at how far you've come. oorja is a quick to use video/voice chat application with realtime collaborative features. Visit oorja.io to see it in action.

react privacy webrtc peer-to-peer collaboration

SRS (Simple Realtime Server)- Simple, high efficiency and realtime video server, supports RTMP/WebRTC/HLS/HTTP-FLV/SRT/GB28181.

•    C++

---

SRS (Simple Realtime Server) is a simple, high efficiency and realtime video server, supports RTMP/WebRTC/HLS/HTTP-FLV/SRT. SRS is industrial-strength live streaming cluster, for the best conceptual integrity and the simplest implementation. SRS provides variety of inputs, for example, Push RTMP to SRS, Push RTSP/UDP/FLV to SRS, Pull Stream to SRS. SRS can transform the RTMP to other protocols or deliveries, for example, RTMP Transcode, Snapshot, Forward to Other Servers, Transmux to HTTP-FLV, Transmux to HLS, Transmux to HDS, Transmux SRT, DVR to FLV/MP4. SRS canbe used in CDN for large stream clusters, for example, RTMP Cluster、 OriginCluster, VHOST, Reload, HTTP-FLV Cluster.

hls cdn rtmp cluster webrtc live-streaming live media-server rtmp-server srs http-streaming http-flv srt low-latency coroutine gb28181 video-streaming video-server

ION - Distributed real-time communication system, let people communicate anydevice, anytime, anywhere

•    Go

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ION is a distributed real-time communication system, let people communicate anydevice, anytime, anywhere. RTC Public PAAS/SAAS are expensive and limited flexibility. Real Destributed RTC System is not easy. Many Public PAAS/SAAS based on destributed architecture with microservices. This is not a simple cluster architecture. ION make all things easy, from front-end to back-end, will support all platforms, all scene, all protocol.

webrtc real-time-communication conference chat video-conference

Pion WebRTC - A pure Go implementation of the WebRTC API

•    Go

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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

coplay - Synchronizing video play between two peers.

•    Javascript

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Synchronizing video play between two peers. When two browsers are playing the same web video, Coplay can connect them with WebRTC (using PeerJS) and enable users to control two video players synchronously.

webrtc peerjs browser-extension video sync peer

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

•    C++

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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