Search Results for "webrtc"
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Showing 22 open source projects for "webrtc"
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Elixir WebRTC
An Elixir implementation of the W3C WebRTC API
Elixir WebRTC is an Elixir implementation of the W3C WebRTC API for building real-time communication features in Elixir applications. It gives developers a way to create peer connections, exchange media, and work with browser-compatible WebRTC behavior from the BEAM ecosystem. The project is especially useful for applications that need live audio, video, or data communication without leaving Elixir. -
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ZLMediaKit
WebRTC/RTSP/RTMP/HTTP/HLS/HTTP-FLV/WebSocket
ZLMediaKit is a high-performance C++11 media server and streaming SDK for real-time audio and video applications. It supports a wide range of protocols, including RTSP, RTMP, HLS, HTTP-FLV, WebSocket-FLV, GB28181, HTTP-TS, HTTP-fMP4, MP4, and WebRTC. The project is designed for protocol conversion, live streaming, low-latency playback, recording, forwarding, and large-scale client access. Its asynchronous, multithreaded network model makes it suitable for commercial streaming deployments with many concurrent connections. ZLMediaKit can be deployed directly as a complete media server or embedded through its standard C API as an SDK for other languages and systems. ... -
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LiveKit
End-to-end stack for WebRTC. SFU media server and SDKs
LiveKit is an open-source project that provides a scalable, multi-user conferencing system based on WebRTC, designed to offer real-time video, audio, and data capabilities for developers. -
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MediaDevices
Go implementation of the MediaDevices API
mediadevices is a Go library developed by the Pion WebRTC team that enables real-time access to audio and video devices for building native Go applications involving media streaming and conferencing. It provides a cross-platform, unified API for capturing and manipulating media streams and is often used in combination with Pion WebRTC for peer-to-peer communications. Its support for device enumeration, media constraints, and frame processing makes it a powerful building block for custom voice and video solutions in Go. -
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Billd-Live
Live broadcast room built on Vue3 + WebRTC + Nodejs + SRS
Billd-live is a full-stack live streaming platform that enables users to create and participate in real-time video broadcasts through a web-based interface. Built using technologies such as Vue3, WebRTC, Node.js, and FFmpeg, it provides a complete solution for live streaming similar to modern platforms. Users can start live sessions as hosts, while viewers can join and watch streams directly in the browser. The system supports multiple streaming protocols and includes features such as real-time interaction, multi-user connections, and content distribution. ... -
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smart rtmpd
Multi-system-supported and easy-to-maintain streaming media server
...The server is designed to be lightweight and easy to deploy, requiring minimal configuration and no external dependencies. It supports a wide range of protocols including RTMP, HLS, DASH, RTSP, and WebRTC, making it suitable for diverse streaming environments. The system is built with scalability in mind, supporting clustering and cascading setups for large-scale deployments. It also includes features such as authentication, logging, and remote configuration through web interfaces. Overall, it provides a powerful yet simple solution for building streaming infrastructure. -
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go2rtc
Ultimate camera streaming application
go2rtc is a lightweight, zero-dependency streaming server designed to unify and convert video streams across a wide range of protocols and devices, particularly in smart home and surveillance environments. Written in Go, it provides real-time streaming capabilities with extremely low latency by supporting protocols such as RTSP, WebRTC, RTMP, HTTP, and HomeKit, while also enabling seamless transcoding using FFmpeg when needed. The application can ingest streams from IP cameras, USB devices, or cloud-based sources and redistribute them to multiple clients or platforms, including browsers and smart home systems like Home Assistant. Its architecture emphasizes flexibility, allowing users to mix multiple input sources, negotiate codecs dynamically, and even enable two-way audio communication with supported devices. go2rtc also includes features for publishing streams to external platforms like YouTube or Telegram, making it useful beyond surveillance scenarios. -
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P2P Media Loader
An open-source engine for P2P streaming of live and on demand video
p2p-media-loader is a JavaScript library that enables peer-to-peer delivery of video content using WebRTC, reducing server load and improving scalability for streaming platforms. It works seamlessly with HLS and DASH formats by extending existing players like hls.js and dash.js, allowing viewers to share video segments with one another. The library is optimized for browser environments and is ideal for developers looking to build decentralized or bandwidth-efficient media applications. -
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Fishjam
General purpose media server. Supports WebRTC, HLS, RTSP, SIP
...The GitHub repository refers to the original open-source Fishjam media server, which was a general-purpose media server written in Elixir and is no longer actively maintained. The broader Fishjam project has since evolved into a platform for WebRTC-based live video, audio, and messaging infrastructure. It is designed for use cases such as video conferencing, live streaming, classrooms, telehealth, gameplay, collaboration, and interactive media products. Fishjam focuses on reducing the amount of WebRTC infrastructure that teams need to build themselves. Its main value is giving developers a practical foundation for real-time communication products with sub-second media delivery. -
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AnalysisAVP
Encode decode, rgb yuv h264 aac flv mp4 rtmp
...It covers foundational topics such as encoding, decoding, color formats like RGB and YUV, and widely used codecs including H.264 and AAC. The project also explores media container formats like MP4 and FLV, along with streaming protocols such as RTMP and WebRTC, offering a broad understanding of media transmission. It includes practical examples and experimental code that demonstrate real-world implementations such as camera capture, rendering, and encoding workflows. The repository also references various multimedia frameworks and libraries like FFmpeg, SDL2, and x264, helping users understand their integration. ... -
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audio_video_streaming
Compilation of authoritative information on audio and video streaming
...It serves as a learning hub for developers interested in multimedia systems, covering topics such as encoding, decoding, transmission protocols, and real-time communication frameworks. The repository includes example implementations like multi-user video chat systems, WebRTC demos, and cross-platform media players to provide hands-on learning opportunities. It also documents widely used technologies such as RTP, RTMP, HLS, and WebRTC, helping users understand the full lifecycle of streaming pipelines from capture to rendering. In addition to educational materials, it references industry tools, libraries, and frameworks, making it a valuable roadmap for both beginners and advanced engineers. ... -
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Metastream
Watch streaming media with friends
...New features are added on top of streaming websites such as real-time chat and timestamp markers. Synchronized playback of streaming media across various websites. Public, private, and offline sessions. Support for WebRTC peer-to-peer connections. Easily add watch party support to your website by redirecting the user to Metastream. Metastream used to be an Electron desktop application until development was stopped due to roadblocks in acquiring a Widevine license. The latest version can be found on the GitHub releases page. -
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Membrane Core
The core of Membrane Framework, multimedia processing framework
membrane_core is the foundation of the Membrane multimedia framework for Elixir, providing the abstractions and runtime needed to build real-time audio and video pipelines. It models media processing as a graph of lightweight, supervised OTP processes—elements connected by links—so work is isolated, fault-tolerant, and easy to scale or reconfigure at runtime. The core defines a clear lifecycle and callback API for elements, plus concepts like buffers, events, and capabilities/format... -
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TL WebRTC File (tl-rtc-file)
WebRTC P2P online web media streaming tool
tl-rtc-file is a lightweight JavaScript library that enables real-time, peer-to-peer file sharing using WebRTC. It allows users to transfer files directly between browsers without relying on a server, cloud storage, or backend infrastructure. The library is ideal for building web apps, chat systems, or support tools that require fast and secure file exchange with minimal setup. With optional compression and simple integration, tl-rtc-file emphasizes efficiency and privacy in web-based file sharing. -
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Peer Calls
Group peer to peer video calls for everyone written in Go
Peer Calls is a self-hosted, open-source WebRTC-based video and audio calling platform for group communication. Designed for simplicity and privacy, it allows anyone to run their own video conferencing service without relying on third-party providers. Peer Calls supports multi-user rooms, screen sharing, and chat, all delivered via a clean web interface. It’s great for small teams, communities, and educational groups seeking secure and customizable alternatives to mainstream conferencing tools. -
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Frontend Park
Frontend projects and experiments
frontend-park is a collection of engaging frontend projects and experiments, showcasing various web technologies. It includes examples ranging from pose and face recognition using TensorFlow.js to WebRTC-based video calls and Three.js 3D animations. This repository serves as a playground for exploring and learning modern frontend development techniques. -
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simple-peer
Simple WebRTC video, voice, and data channels
simple-peer is a lightweight JavaScript library that simplifies peer-to-peer WebRTC connections for real-time communication directly between browsers and devices. The project abstracts away much of the complexity of the native WebRTC API, allowing developers to establish audio, video, and data channels with minimal configuration. It supports direct peer communication without requiring heavy signaling infrastructure beyond the initial connection exchange. simple-peer is commonly used in browser-based chat applications, multiplayer games, collaborative tools, decentralized systems, and streaming platforms. ... -
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Telegram WebRTC (VoIP)
Voice chats, private incoming and outgoing calls in Telegram
Telegram WebRTC (VoIP) is a Python and C++ library that enables real-time voice and video communication features for Telegram bots and clients. It provides an interface for joining, managing, and streaming audio or video in Telegram group calls and voice chats. The library is built on top of low-level communication protocols, ensuring efficient handling of real-time media streams. -
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jeelizFaceFilter
Javascript/WebGL lightweight face tracking library
This JavaScript library detects and tracks the face in real time from the camera video feed captured with WebRTC. Then it is possible to overlay 3D content for augmented reality applications. We provide various demonstrations using main WebGL 3D engines. We have included in this repository the release versions of the 3D engines to work with a determined version (they are in /libs/<name of the engine>/). This library is lightweight and it does not include any 3D engine or third-party library. ... -
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Janus WebRTC Gateway
Perfect Docker Image for Media Streaming Expert Users
Perfect Docker image for media streaming expert users. This is a docker image for Janus Webrtc Gateway. Janus Gateway is still in the active development phase. -
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PHP HTML5 Videochat
Simple PHP setup for HTML5 Videochat web application by VideoWhisper.
Using the HTML5 videochat interface is highly recommended as main browsers have plans to discontinue Flash support in 2020. HTML5 Videochat uses WebRTC technology to allow broadcasting webcam directly from website, without need for Flash. This is a simple embedding preview edition, with simple scripts to embed app and showcase few features. Live demo for this edition is available at: https://videowhisper.com/demos/html5-videochat/ For a full implementation of advanced capabilities, see turnkey site edition available as WordPress plugin with full php source: https://paidvideochat.com/html5-videochat/ Compatible turnkey complete hosting plans with HTML5 relay streaming and installation included: https://webrtchost.com/hosting-plans/#Complete-Hosting Contact VideoWhisper anytime for clarifications, suggestions, custom development evaluation: https://videowhisper.com/tickets_submit.php -
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MCU Media Server
SIP Video Multiconference Media Server with WebRTC support.
REPOSITORY MOVED TO GITHUB!! https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. .
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