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versatica avatar

versatica/mediasoup

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mediasoup.org↗

Mediasoup

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 bidirectional data exchange via SCTP channels. It handles connection negotiation and secure session establishment across diverse platforms and browsers using WebRTC and RTP transport.

The system also includes observability tools for tracking real-time stream statistics and active speaker data.

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Features

  • Realtime Media Routing - Provides a library for embedding low-latency real-time media routing logic directly into applications.
  • Low-Latency Streaming - Implements a selective forwarding unit to relay audio and video with minimal latency using WebRTC.
  • Media Distribution Networks - Efficiently distributes media from a single producer to many consumers to optimize network bandwidth.
  • Media Track Management - Provides comprehensive controls for sending, receiving, and managing individual audio and video tracks between participants.
  • Simulcast Controllers - Supports simulcast controllers to send multiple quality layers of a stream for adaptive bandwidth management.
  • Adaptive Bitrate Controllers - Dynamically adjusts media resolution and bitrate based on real-time network congestion metrics.
  • Connection Handshake Protocols - Handles the complex signaling and handshake process required to establish stable media connections.
  • Selective Forwarding Units - Functions as a selective forwarding unit that optimizes bandwidth by deciding which streams to forward to specific participants.
  • Network Transport Protocols - Provides optimized network transport layers for managing multiple media tracks over secure sessions.
  • Connectivity Establishment - Facilitates the exchange of RTP and SCTP parameters to establish secure peer-to-peer media sessions.
  • Real-Time Media Transport - Implements low-latency transport of audio and video streams using the Real-time Transport Protocol (RTP).
  • Video Conferencing Systems - Supports the creation of multi-party video conferencing systems where users share audio and video tracks.
  • WebRTC Media Orchestration - Provides the orchestration layer for managing low-latency audio and video streaming using WebRTC.
  • JSON Interfaces - Uses a structured JSON interface for negotiating media parameters and connection handshakes between clients and servers.
  • Real-Time Media Servers - Acts as a real-time media server that forwards audio and video streams between multiple participants.
  • Media Stream Statistics - Tracks and displays performance metrics and active speaker data for all media participants in a session.
  • Screen Sharing - Supports sharing a user's screen as a media track to distribute visual content to other session participants.
  • Worker Process Isolation - Implements native C++ worker processes to isolate heavy media processing from the main JavaScript event loop.
  • Adaptive Quality Adjustments - Optimizes stream resolution and bitrate dynamically using quality layers and bandwidth estimation.
  • Spatial and Temporal Layering - Allows requesting specific spatial or temporal layers of a media stream to balance bandwidth and visual quality.
  • Bandwidth Management - Adjusts video resolution and bitrate based on real-time bandwidth estimations to prevent playback lag.
  • Unified Media APIs - Provides a unified API to interface with browsers and various devices for transmitting media streams.
  • Custom Data Channels - Enables bidirectional exchange of arbitrary data between clients and servers via dedicated data channels.
  • Multi-Stream Transport Management - Transports multiple audio and video streams over a single connection to improve overall network efficiency.
  • Peer-to-Peer Data Exchange - Allows the transmission of non-media data packets between peers over secure network connections.
  • SCTP Transmitters - Utilizes the Stream Control Transmission Protocol (SCTP) for reliable transmission of non-media data channels.
  • Real-Time Data Pushing - Provides bidirectional channels for sending real-time data messages to synchronize application state.
7,146 stars·1,222 forks·C++·isc·33 views

Star history

Star history chart for versatica/mediasoupStar history chart for versatica/mediasoup

How this analysis was created: This summary and feature list are AI-generated from collected project material and can contain mistakes. Stars, license and language are imported from GitHub. Inclusion does not mean that we have tested or audited this project. Check the source documentation for any feature you depend on. Learn more on our About page.

Frequently asked questions

What does versatica/mediasoup do?

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.

What are the main features of versatica/mediasoup?

The main features of versatica/mediasoup are: Realtime Media Routing, Low-Latency Streaming, Media Distribution Networks, Media Track Management, Simulcast Controllers, Adaptive Bitrate Controllers, Connection Handshake Protocols, Selective Forwarding Units.

Which projects share features with versatica/mediasoup?

Projects with overlapping indexed features include: steveseguin/vdo.ninja — VDO.Ninja is a low-latency peer-to-peer media routing service and video streaming platform designed to integrate… pipecat-ai/pipecat — Pipecat is a framework and software development kit for building real-time multimodal AI agents and speech-to-speech… open-webrtc-toolkit/owt-server — This project is a scalable real-time communication platform designed for multi-party video and audio conferencing. It… meetecho/janus-gateway — Janus is a WebRTC media gateway that routes real-time audio, video, and data between web browsers and server-side… bigbluebutton/bigbluebutton — BigBlueButton is an open-source virtual classroom software and meeting server designed for hosting real-time online… react-native-webrtc/react-native-webrtc — react-native-webrtc is a mobile framework wrapper and peer-to-peer communication library used to integrate real-time…

Projects sharing features with Mediasoup

These projects share indexed features with Mediasoup. Shared tags can include platform or build tooling; verify the primary use case before treating a result as a replacement.
  • steveseguin/vdo.ninjasteveseguin avatar

    steveseguin/vdo.ninja

    3,910View on GitHub↗

    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

    JavaScriptlivelow-latencyninja
    View on GitHub↗3,910
  • pipecat-ai/pipecatpipecat-ai avatar

    pipecat-ai/pipecat

    12,846View on GitHub↗

    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

    Pythonaichatbot-frameworkchatbots
    View on GitHub↗12,846
  • open-webrtc-toolkit/owt-serveropen-webrtc-toolkit avatar

    open-webrtc-toolkit/owt-server

    1,157View on GitHub↗

    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

    JavaScriptmcusfustreaming
    View on GitHub↗1,157
  • bigbluebutton/bigbluebuttonbigbluebutton avatar

    bigbluebutton/bigbluebutton

    9,040View on GitHub↗

    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

    JavaScriptbigbluebuttoncollaborationconferencing
    View on GitHub↗9,040
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