Open-source alternatives to React Native Webrtc

apprtc is a WebRTC video chat application and signaling server designed to establish peer-to-peer audio and video communication between browsers. It provides a coordination layer using a websocket-based signaling server to exchange session descriptions and network candidates. The project is delivered as a dockerized communication app, allowing for a containerized deployment of the calling service for local development or cloud hosting. It includes a network gateway that integrates STUN and TURN servers to facilitate media flow through firewalls and NATs. The implementation covers peer discov

flutter-webrtc is a real-time communication SDK and plugin for the Flutter framework. It provides a set of tools for establishing peer-to-peer media connections and low-latency data exchange across mobile, desktop, and web environments. The project enables the creation of applications with live audio and video calling, real-time media streaming, and peer-to-peer data channels for sending encrypted arbitrary data packets without a central server. It supports secure media communication through end-to-end encryption for audio, video, and data streams. The SDK covers broad capabilities including

Mumble is a low-latency voice over IP system designed for real-time communication. It provides a self-hosted, encrypted voice server that secures data transmission using public-private key authentication and digital certificates. The platform distinguishes itself through a request-response protocol for remote configuration and integration with external automation scripts. It also features a graphical overlay that renders active speaker information and chat status on top of other running applications, as well as positional audio to simulate the 3D spatial location of speakers in a virtual envi

This repository provides a collection of reference implementations and practical demonstrations for using WebRTC to establish real-time audio, video, and data communication. It contains code samples for negotiating peer-to-peer connections, managing media streams, and utilizing low-latency data channels. The project demonstrates the capture of audio and video from hardware devices, as well as the redirection of canvas element content into media streams. It includes examples of transferring arbitrary text and binary data between peers and managing the negotiation of direct connections. The sa

simple-peer is a JavaScript library that provides a wrapper for WebRTC to simplify the establishment of peer-to-peer networking in the browser. It serves as a tool for creating direct device-to-device connections for the transmission of binary data and real-time media streaming. The library manages the exchange of strings and binary buffers through a data channel implementation and provides tools for sharing real-time audio and video tracks between peers. It covers the full lifecycle of peer connectivity, including signaling coordination, session description negotiation, and the gathering of

Mediasoup is a selective forwarding unit used for real-time media routing. It enables the development of low-latency audio and video communication systems by routing streams between participants without transcoding. The project provides embedded media routing logic that can be integrated directly into an application. It supports simulcast and quality layering, allowing the system to adapt resolution and bitrate based on real-time bandwidth estimations to maintain connection stability. The capability surface includes media track management for audio, video, and screen capture, as well as bidi

Simpl is an HTML, CSS, and JavaScript example library that provides a collection of minimal functional demonstrations of core web technologies and native browser APIs. It serves as a reference implementation and pattern gallery for frontend development, offering practical examples of how to implement common web features. The project showcases a wide range of browser capabilities, including real-time web communication via WebSockets and WebRTC, responsive web design techniques for adaptive layouts, and the implementation of offline workflows using service workers. It also provides demonstratio

This library provides a complete implementation of the WebRTC protocol suite in Rust, enabling peer-to-peer audio, video, and data channels with an asynchronous, Rust-native stack. Its architecture separates protocol logic from I/O, giving developers control over threading and scheduling while the library manages the protocol state. The stack is built to be modular, supporting data channel multiplexing over a single SCTP association, an event-driven callback interface for responding to connection changes, and an interceptor pipeline for processing media packets without altering core protocol c

Mirotalksfu is a WebRTC video conferencing platform and AI-integrated meeting suite. It functions as a real-time communication system for hosting high-resolution audio and video meetings, serving as a self-hosted virtual classroom and a collaborative workspace. The platform distinguishes itself by integrating generative AI assistants, speech recognition, and digital avatars into live sessions. It also operates as an RTMP streaming gateway, allowing users to broadcast live meeting content to external audiences and platforms. The system provides a collaboration suite featuring a shared interac

Janus is a WebRTC media gateway that routes real-time audio, video, and data between web browsers and server-side application logic. It functions as a central media relay that manages session negotiation and encryption for multiple browser endpoints. The project utilizes a modular plugin architecture that decouples the core server from specific media logic, allowing developers to implement custom modules for media processing, event handling, and transport protocols. This design enables the server to act as a protocol translation bridge, converting WebRTC streams into legacy formats such as SI

flv.js is a JavaScript library and HTML5 Media Source Extension wrapper designed to play FLV video and live streams in web browsers. It enables the rendering of FLV content within a standard web video element without the need for external plugins. The project focuses on real-time transmuxing, converting FLV container data into fragmented MP4 segments to ensure browser compatibility. It includes specific implementations for low-latency live streaming and cross-origin video loading via CORS-compliant server headers. The library provides capabilities for segmented media playback, media position

PeerJS Server is a WebRTC signaling server and connection broker that coordinates handshakes between clients to establish direct peer-to-peer connections. It functions as a coordination point for discovering peer identifiers and initiating real-time media and data streams between remote browser clients. The project can be deployed as a dedicated signaling server or integrated as Node.js middleware within an existing web application to share a single network port. It manages the lifecycle of peer connections through a centralized signaling process, assigning unique identifiers to clients and m

This project is a comprehensive reference guide and directory of web browser capabilities. It serves as a technical map for accessing native operating system functions, hardware interfaces, and standard web APIs to bridge the gap between web applications and desktop or mobile environments. The resource provides detailed guidance on implementing Progressive Web App features, including offline caching, push notifications, and native installation prompts. It also catalogs methods for interacting with hardware peripherals via USB, Bluetooth, and NFC, as well as reading raw data from device sensor

aiortc is a Python implementation of the WebRTC protocol, providing an asynchronous stack for real-time audio, video, and data streaming between peers. It functions as a media engine and data channel implementation that manages peer connections through a non-blocking framework. The library enables the establishment of direct peer-to-peer networking by negotiating session descriptions and connectivity paths. It facilitates secure, bidirectional data transmission via data channels and supports low-latency media streaming using standard codecs. The system includes capabilities for network NAT t

FastRTC is a Python framework for building low-latency, bidirectional audio and video streams. It serves as a real-time communication library that provides a wrapper for WebRTC media servers, allowing users to create streaming applications with integrated media handling. The project distinguishes itself by providing a gateway for telephony integration, which maps temporary phone numbers to streaming media endpoints. It also includes built-in voice activity detection to manage automatic turn-taking and speech boundary identification in real-time conversations. The library supports mounting me

This project is a framework for developing multimodal AI agents that function as programmable participants in real-time communication rooms. It enables the construction of agents that can see, hear, and speak by integrating speech-to-text, large language models, and text-to-speech pipelines to facilitate low-latency, natural conversations. The system is distinguished by its advanced orchestration of real-time media and conversational flow, including support for full-duplex speech, preemptive response generation, and sophisticated interruption management. It further differentiates itself throu

go2rtc is a media streaming server that functions as a protocol-agnostic gateway for video and audio feeds. It ingests media from diverse sources and redistributes them across multiple streaming standards, enabling compatibility between proprietary camera hardware and web-based playback clients. The system utilizes a centralized configuration schema to manage stream routing and lifecycle orchestration based on client demand. The platform distinguishes itself through its focus on low-latency delivery, utilizing peer-to-peer connections to facilitate sub-second playback directly within web brow

Filepizza is a web-based peer-to-peer file sharing application that enables direct browser-to-browser data exchange. It utilizes WebRTC to establish connections between devices, allowing files to be sent without uploading data to a central server. The project provides a password-protected file sharing mechanism that secures transmissions via symmetric-key stream encryption. It includes a containerized signaling server that can be self-hosted to coordinate connections between peers and facilitate NAT traversal across restrictive networks and firewalls. The application supports multi-file tran

Sharedrop is a browser-based file drop application for transferring files directly between devices without requiring software installation or account registration. It uses WebRTC to establish peer-to-peer data channels, allowing files to be exchanged between different operating systems without routing traffic through a central server. The project provides mechanisms for both local and remote connectivity. It identifies active users sharing the same public IP address for immediate local device discovery and allows the creation of private sharing rooms via unique session links for users on diff

Moonlight for Android is a remote game streaming client that receives low-latency video from a remote host computer and transmits input commands back in real time. It enables remote desktop control and the management of a host computer from a mobile device. The project supports remote VR video output via 3D side-by-side video rendering for compatible monitors or headsets. It includes display output optimizations such as portrait mode, screen rotation, and view panning. The software manages input through the mapping of physical gamepads, virtual touch buttons, and mouse modes. It handles remo

This is a Rust library and peer-to-peer toolkit that implements the WebRTC protocol. It provides a network stack for establishing direct audio, video, and data streaming between peers to achieve low-latency real-time communication. The project is designed as an async-compatible network stack that maintains compatibility across different asynchronous runtimes and executors. It further functions as a deterministic protocol testing tool by separating communication logic from network I/O, allowing protocol behavior to be verified without using live network traffic.

This project is a real-time communication and media streaming server designed for broadcasting, recording, and distributing audio and video content. It functions as a live streaming server and an RTMP media server, providing the infrastructure necessary to deliver real-time media to multiple concurrent viewers over a network. The system distinguishes itself through a multi-protocol media gateway that supports RTMP, E-RTMP v2 for modern codecs like HEVC and AV1, and HTTP media tunneling to bypass restrictive firewalls. It further acts as a real-time communication platform by synchronizing shar

Mirotalk is a self-hosted video conferencing platform and peer-to-peer communication server. It utilizes WebRTC for high-resolution video calls, screen sharing, and real-time audio streaming, operating as a decentralized system that routes media streams directly between participants to ensure privacy and low latency. The platform integrates conversational language models and speech recognition to provide real-time transcription and automated meeting tasks. It also functions as a white-label collaboration suite, allowing users to apply corporate visual identities across the interface and sourc

This project is an RTMP media streaming SDK and a real-time communication framework designed for pushing and playing audio and video streams. It provides tools for interactive broadcasting, low-latency voice and video calls, and a cross-platform media player compatible with Windows, iOS, and Android. The toolkit enables interactive live broadcasting with support for multi-host interactions and the ability to push streams to distribution servers via CDN. It includes a cloud recording manager for capturing live sessions and saving them as files to cloud storage, along with a system for composit

Jitsi is an open-source communication suite providing a self-hosted platform for real-time video conferencing, audio calls, and instant messaging. It functions as a comprehensive toolkit for streaming voice and video across different devices and platforms, allowing for the deployment of private team collaboration environments. The project is distinguished by its multi-protocol communication bridging, which translates and exchanges data between diverse network standards including SIP, XMPP, and IRC. This allows users on different protocols to interact within a unified communication interface.

Neko is a virtual desktop infrastructure platform that provides containerized browser isolation and remote desktop environments. It enables users to host secure, ephemeral browser instances that can be accessed and managed through a standard web browser, ensuring consistent execution across different host systems. The platform distinguishes itself through its collaborative capabilities, allowing multiple users to view and interact with a single shared browser session in real time. It synchronizes keyboard, mouse, and gamepad inputs from multiple participants while providing integrated tools f

ShareDrop is a browser-based platform designed for direct, peer-to-peer file exchange between devices. It facilitates data transfer by establishing encrypted connections directly between browsers, ensuring that files are transmitted without being stored on central servers. The system identifies nearby devices on the same local network to enable immediate, configuration-free file sharing. For remote participants, it provides the ability to create private, temporary web rooms that use unique, ephemeral addresses to establish secure connections across different networks. The project coordinates