meetecho/janus-gateway

Features

• Real-Time Media Streaming - Broadcasts live audio and video content to web browsers with low latency.
• Real-Time Media Servers - Acts as a real-time media server managing the routing of audio, video, and data streams.
• WebRTC Media Processing Plugins - Enables the development of server-side plugins for bespoke WebRTC media processing and communication logic.
• Live Media Streaming - Delivers live audio and video streams to web browsers for real-time playback.
• RTP/RTCP Packet Processors - Processes and forwards individual RTP and RTCP packets for real-time media streaming and control.
• Media Protocol Bridges - Bridges WebRTC streams into legacy formats like SIP or RTP to enable interoperability with telephony systems.
• STUN/TURN Traversal Services - Configures STUN servers and gathering rules to facilitate peer-to-peer connectivity across NATs and firewalls.
• Video Conferencing Systems - Manages multi-party video conferencing rooms with synchronized audio, video, and data channels.
• WebRTC Media Orchestration - Routes real-time audio, video, and data between web browsers and server-side logic using WebRTC.
• ICE-Based Peer Connection Managers - Handles network discovery and maintenance using ICE to find optimal communication paths.
• Multi-Protocol Transports - Provides a transport-agnostic control plane supporting HTTP, WebSockets, MQTT, and other network protocols.
• Shared Secret Keys - Restricts API access by requiring a pre-shared secret key for every request.
• API Request Authentication - Verifies client identity using secure validation processes before granting access to the server API.
• Plugin Access Tokens - Manages the lifecycle and plugin-specific permissions of authentication tokens.
• Token-Based Authentication - Validates per-session cryptographic tokens on every API call to ensure authorized interaction.
• Media Processing Plugins - Provides a modular plugin architecture for implementing bespoke media processing and server logic.
• WebRTC Media Plugins - Loads and manages specialized WebRTC plugins such as audio bridges and video rooms.
• Media Gateways - Functions as a central WebRTC media gateway managing negotiation and encryption for multiple browser endpoints.
• Plugin Attachments - Enables users to attach to specific plugins to exchange media and data streams.
• Modular Plugin Architectures - Employs a modular plugin architecture to decouple core server logic from specific media processing modules.
• DTLS-SRTP Configurations - Configures DTLS-SRTP encryption to secure real-time media streams within WebRTC sessions.
• Plugin Message Exchanges - Implements a JSON-based request and event exchange mechanism between clients and server-side plugins.
• Plugin Media Routers - Relays RTP/RTCP and JSON messages between browsers and server-side plugins.
• WebRTC Session Lifecycles - Manages the full WebRTC session lifecycle, including media signaling and negotiation between browsers.
• Plugin Command Dispatchers - Dispatches synchronous requests to server-side plugins to query or configure their internal state.
• Lightweight Gateway Deployments - Offers a small-footprint deployment suitable for cloud gateways or embedded devices.
• Media Capture and Playback - Captures real-time media streams to disk and replays them back to users via WebRTC.
• Media Processing Plugins - Enables the creation of custom modules to handle and process media streams for delivery to browsers.
• Multi-Protocol Streaming Delivery - Distributes a single video source across multiple streaming transport protocols including WebRTC, WebSockets, and HTTP.
• Stream Recording - Captures live network streams to local files for archival and on-demand playback.
• Stream Publishing - Provides capabilities for broadcasting live media from a source to multiple concurrent viewers.
• Session Lifecycle Controls - Provides administrative controls to list, accept, and destroy active sessions for server load management.
• Websocket Connection Managers - Maintains persistent WebSocket connections for real-time control communication as an alternative to REST APIs.
• Custom Transport Protocols - Supports the implementation of custom transport protocols for flexible server-client communication.
• Protocol-Agnostic Transport Layers - Provides a protocol-agnostic transport layer for routing control messages across different network protocols.
• Audio Stream Mixers - Mixes and transcodes multiple participant audio streams into a single output for efficient conferencing.
• Session State Management - Tracks persistent client sessions and maps them to specific plugin handles for stateful communication.
• SIP Bridges - Bridges WebRTC sessions with traditional SIP-based telephony infrastructure for voice and video calls.
• SCTP - Transmits arbitrary application data over SCTP channels alongside media streams.
• Telephony and Media Bridging - Bridges modern WebRTC browser sessions with traditional SIP-based telephony infrastructure.
• Peer Connectivity - Relays audio, video, and data channels between endpoints without processing the media.
• Custom Event Handlers - Allows development of custom plugins that process and respond to specific server-side events.
• Event-Driven Hook Systems - Implements an event-driven hook system that forwards server events to external handlers via pub-sub.
• Real-Time Event Notifications - Sends real-time event notifications to external endpoints such as RabbitMQ or MQTT.
• Real-Time Server Activity Monitors - Provides an admin interface to monitor real-time server activity, active sessions, and plugin health.
• Streaming Servers - General-purpose WebRTC server supporting SFU architecture.

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