react-native-webrtc/react-native-webrtc
React Native Webrtc
react-native-webrtc est un wrapper de framework mobile et une bibliothèque de communication peer-to-peer utilisée pour intégrer la communication audio, vidéo et de données en temps réel dans des applications mobiles multiplateformes. Il fonctionne comme un outil pour capturer les écrans des appareils et diffuser du contenu multimédia en direct via le protocole WebRTC.
Le projet fournit une interface pour les canaux de données bidirectionnels, permettant l'échange de paquets de données arbitraires entre pairs sans serveur de relais central. Il prend en charge le streaming multimédia en temps réel, incluant la transmission audio et vidéo peer-to-peer et des capacités de capture d'écran.
La bibliothèque inclut des fonctionnalités pour l'optimisation des flux, telles que l'envoi de plusieurs versions d'un flux vidéo à différentes résolutions pour s'adapter aux conditions réseau du récepteur. Elle couvre des domaines de communication de haut niveau, incluant la voix sur IP mobile et la communication vidéo en temps réel.
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.
