30 open-source projects similar to versatica/mediasoup, ranked by how many features they have in common. Compare stars, activity and what each one does to find the best Mediasoup alternative.
VDO.Ninja is a low-latency peer-to-peer media routing service and video streaming platform designed to integrate remote audio and video feeds into professional production workflows. It functions as a WebRTC broadcast integration tool and studio controller, allowing for the direct transmission of high-definition media between publishers and viewers with minimal delay. The platform distinguishes itself through extensive protocol bridging, converting between WebRTC, WHIP, WHEP, SRT, and RTMP to ensure compatibility across diverse network environments and professional studio software. It includes
Pipecat is a framework and software development kit for building real-time multimodal AI agents and speech-to-speech systems. It utilizes a frame-based data pipeline to route audio, video, and text through a modular sequence of processors, enabling the orchestration of low-latency conversational AI. The project is distinguished by its ability to coordinate complex multimodal services, including speech-to-text, language models, and text-to-speech, within a single pipeline. It features semantic voice activity detection for natural turn-taking, state-machine conversation flows for dialogue manag
This project is a scalable real-time communication platform designed for multi-party video and audio conferencing. It functions as a modular media server that orchestrates live sessions, manages virtual meeting rooms, and handles the distribution of media streams across distributed computing environments. The platform distinguishes itself through its support for protocol bridging, allowing for the translation between WebRTC, RTSP, SIP, and HLS. It incorporates hardware-accelerated transcoding to maintain performance across diverse devices and utilizes a plugin-based architecture to enable cus
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
BigBlueButton is an open-source virtual classroom software and meeting server designed for hosting real-time online teaching sessions. It functions as a WebRTC video conferencing platform and collaboration suite, providing a self-hosted environment for virtual classroom management and online course administration. The system distinguishes itself through specialized educational tools, including an interactive quiz engine, breakout room coordination, and live polling. It provides a comprehensive suite of collaborative assets such as a shared infinite whiteboard, real-time co-authored notes, and
react-native-webrtc is a mobile framework wrapper and peer-to-peer communication library used to integrate real-time audio, video, and data communication into cross-platform mobile applications. It functions as a tool for capturing device screens and streaming live media content via the WebRTC protocol. The project provides an interface for bidirectional data channels, enabling the exchange of arbitrary data packets between peers without a central relay server. It supports real-time media streaming, including peer-to-peer audio and video transmission and screen capture capabilities. The libr
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
hls.js is a JavaScript player library that enables HTTP Live Streaming playback in web browsers using the Media Source Extensions API. It functions as an adaptive bitrate streaming engine and a media streamer that manages the buffering and playback of video segments. The library features specialized capabilities for low latency streaming and digital rights management, allowing it to decrypt protected content via Encrypted Media Extensions. It includes a metadata processor for handling ID3 tags and date-range elements, as well as a pluggable loader interface to replace default network request
This project is a comprehensive computer networking textbook and instructional resource. It serves as a technical guide for the design and implementation of network layers, protocols, and hardware architecture, covering the spectrum from physical links to application-layer protocols. The content provides a detailed study of standards for congestion control, reliable data delivery, and internetwork routing. It includes specialized technical material on network security, public-key infrastructure, and the operation of modern cloud infrastructure and data centers. The material covers a broad ra
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
LiveKit is a comprehensive framework for building and orchestrating real-time, multimodal AI agents that interact with users through voice, video, and text. It provides a centralized, event-driven architecture to manage the entire lifecycle of automated participants, from initialization and session state management to graceful shutdown. By utilizing a selective forwarding unit, the platform efficiently routes media streams between participants and agents, ensuring low-latency communication and secure, token-based authentication for all connections. The platform distinguishes itself through it
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
Nettu-meet is a self-hosted video conferencing platform designed for professional and educational environments. It provides a comprehensive suite of real-time communication tools, including audio and video streaming, screen sharing, and collaborative digital whiteboards, to facilitate remote tutoring and group meetings. The platform distinguishes itself through its focus on programmatic integration and organizational customization. It exposes a RESTful API that allows external systems to automate the scheduling and management of meeting sessions, enabling the integration of live video workflo
This project is a community-curated directory of open-source software designed for deployment in private server environments and home labs. It serves as a comprehensive resource for discovering independent, self-hosted alternatives to mainstream cloud services, enabling users to maintain full data ownership and control over their digital infrastructure. The directory is structured through a hierarchical taxonomy that organizes a vast collection of applications into logical categories, ranging from media management and data analytics to private communication and team productivity tools. It dis
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.
SRS is a real-time media server designed to ingest, route, and distribute live audio and video streams across various transport protocols. It functions as a multi-protocol stream relay, including a dedicated RTMP media gateway and a WebRTC signaling server to coordinate peer-to-peer media exchanges. The system features a multi-protocol relay engine that transforms incoming media packets between different transport formats without re-encoding. This allows it to serve as a video delivery proxy that routes live media from a single source to multiple concurrent viewers using diverse delivery prot
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
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
gost is a multi-protocol proxy tunnel and secure tunneling server designed to route network traffic through encrypted connections. It functions as a traffic obfuscation gateway and a transparent proxy server capable of intercepting TCP and UDP traffic at the IP level. The project also includes a virtual network interface manager for creating TUN and TAP devices to intercept operating system packets. The system distinguishes itself through a chain-based request routing model, allowing traffic to pass through an ordered sequence of proxy nodes. It provides extensive transport-layer encapsulatio
Video.js is a customizable HTML5 video player framework and JavaScript media plugin system. It provides a foundation for rendering and controlling web video playback across different devices and screen sizes, supporting both standard HTML5 integration and adaptive bitrate streaming for live or on-demand content. The project is distinguished by a modular architecture that allows for the extension of playback functionality through a class-based plugin system and middleware-based method interception. This enables the development of tailored media interfaces and the integration of specialized beh
This project is a browser-based video conferencing platform that facilitates direct, peer-to-peer communication between participants. By utilizing browser-native media capture and direct network connections, it enables real-time audio, video, and screen sharing without relying on central media relay servers. The platform distinguishes itself through its focus on ephemeral, private communication. It generates unique, short-lived session identifiers for virtual rooms that exist only in memory, ensuring that no persistent user accounts or long-term storage are required. To maintain connection st
GameNetworkingSockets is a UDP networking library providing secure transport, peer discovery, and traffic control systems. It implements a networking layer for reliable and unreliable messaging over UDP, including tools for message fragmentation and reassembly. The project features a peer-to-peer NAT traversal tool for establishing direct host-to-host connections by punching through firewalls and network address translation layers. It secures network traffic through encrypted transport and secure key exchange. The library includes a traffic manager to organize data into prioritized lanes to
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
MediaMTX is a multi-protocol media server designed for routing, proxying, and recording real-time audio and video streams. It functions as a programmable media router and a gateway between streaming standards such as RTMP, RTSP, and WebRTC, enabling the conversion of live media between different protocols. The server distinguishes itself through on-the-fly format transmuxing and protocol-agnostic routing, which decouples input and output protocols via an internal media bus. It features a programmable automation system that executes external shell commands via event hooks triggered by client c
Moonlight-qt is a cross-platform client application designed to stream high-fidelity audio and video from a remote host computer to a local device. It functions as a remote desktop and gaming client, enabling users to interact with host environments by forwarding local keyboard, mouse, and controller inputs to the remote machine. The software distinguishes itself through the use of hardware-accelerated decoding, which processes incoming high-bitrate streams to maintain low latency and minimal CPU overhead. It supports high frame rate streaming and multi-channel surround sound, ensuring that i
This project is a media streaming server designed for the ingestion, packetization, and distribution of live audio and video content. It functions as a core infrastructure tool for managing real-time media delivery, providing the necessary logic to handle session signaling, bitstream transformation, and the multiplexing of raw data into standardized network packets. The server distinguishes itself through its support for a wide range of industry-standard streaming protocols and container formats. It includes specialized engines for adaptive bitrate delivery, enabling the conversion of raw bit
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
This is a Raft consensus library and distributed consensus engine implemented in Go. It provides the primitives necessary to build fault-tolerant distributed services by implementing a replicated state machine that ensures a group of servers agree on a shared system state through leader election and log replication. The project distinguishes itself through a pluggable architecture for storage backends and snapshot storage, decoupling the consensus logic from physical persistence. It includes specialized mechanisms for leadership transfer, protocol version management to support rolling upgrade
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
FFmpeg is a cross-platform multimedia framework designed for the recording, conversion, and streaming of audio and video content. It functions as a comprehensive toolkit that provides both a command-line utility for direct media manipulation and a collection of low-level libraries for integration into custom applications. At its core, the project utilizes a packet-based stream engine and a format-agnostic abstraction layer to handle diverse media standards, containers, and network protocols. The framework distinguishes itself through a modular, graph-based filter execution model that allows f