react-native-webrtc/react-native-webrtc
React Native Webrtc
react-native-webrtc ist ein mobiler Framework-Wrapper und eine Peer-to-Peer-Kommunikationsbibliothek, die verwendet wird, um Echtzeit-Audio-, Video- und Datenkommunikation in plattformübergreifende mobile Anwendungen zu integrieren. Es fungiert als Werkzeug zur Erfassung von Gerätebildschirmen und zum Streamen von Live-Medieninhalten über das WebRTC-Protokoll.
Das Projekt bietet eine Schnittstelle für bidirektionale Datenkanäle, die den Austausch beliebiger Datenpakete zwischen Peers ohne zentralen Relay-Server ermöglicht. Es unterstützt Echtzeit-Medien-Streaming, einschließlich Peer-to-Peer-Audio- und Videoübertragung sowie Funktionen zur Bildschirmaufnahme.
Die Bibliothek enthält Funktionen zur Stream-Optimierung, wie das Senden mehrerer Versionen eines Videostreams in verschiedenen Auflösungen, um sich an die Netzwerkbedingungen des Empfängers anzupassen. Sie deckt hochrangige Kommunikationsdomänen ab, einschließlich mobilem Voice-over-IP und Echtzeit-Videokommunikation.
Features
- Low-Latency Streaming - Enables high-quality, low-latency live audio and video streaming between mobile clients using WebRTC.
- Real-Time Media Streaming - Provides tools for capturing device screens and streaming live media content via the WebRTC protocol.
- Bidirectional Communication Channels - Provides low-latency bidirectional channels for transmitting arbitrary data packets alongside media streams.
- Peer-to-Peer Data Exchange - Facilitates the exchange of real-time data packets and files directly between mobile devices.
- WebRTC Data Channels - Provides an interface for secure, encrypted peer-to-peer transport of real-time text and binary data.
- STUN/TURN Traversal Services - Implements NAT traversal using STUN and TURN servers to establish direct media paths between peers.
- Peer-to-Peer Media Libraries - Integrates the native C++ WebRTC library to handle complex peer-to-peer signaling and media processing.
- Real-time Data Packet Transmission - Implements low-level dispatching of raw data packets between peers without a central relay server.
- Real-Time Voice and Video Communication - Provides the core capabilities needed to build mobile applications for live peer-to-peer video calls.
- Native C-JavaScript Bridges - Provides a bridge that serializes JavaScript calls into native C++ method invocations to control WebRTC.
- WebRTC Wrappers - Acts as a programmatic wrapper for integrating WebRTC connectivity into cross-platform mobile applications.
- Screen Streams - Captures the device screen and pushes it as a live video stream to remote peers.
- Hardware Accelerated Media Encoders - Utilizes device-specific hardware codecs to compress audio and video streams with minimal CPU overhead.
- Simulcast Controllers - Implements simulcast capabilities to transmit multiple video resolutions, optimizing playback quality based on the receiver's network conditions.
- Mobile Screen Sharing - Allows users to capture and stream their mobile device screen to remote participants in real time.
- Native View Integration - Embeds native video rendering surfaces directly into the mobile UI hierarchy for high-performance playback.
- Low-Latency Voice over IP - Supports the implementation of voice over IP calling with low-latency audio transmission.
