react-native-webrtc/react-native-webrtc
React Native Webrtc
react-native-webrtc es un envoltorio de framework móvil y biblioteca de comunicación entre pares (P2P) utilizada para integrar audio, video y comunicación de datos en tiempo real en aplicaciones móviles multiplataforma. Funciona como una herramienta para capturar pantallas de dispositivos y transmitir contenido multimedia en vivo mediante el protocolo WebRTC.
El proyecto proporciona una interfaz para canales de datos bidireccionales, permitiendo el intercambio de paquetes de datos arbitrarios entre pares sin un servidor de retransmisión central. Admite transmisión de medios en tiempo real, incluyendo transmisión de audio y video entre pares y capacidades de captura de pantalla.
La biblioteca incluye características para la optimización de flujos, como el envío de múltiples versiones de un flujo de video a diferentes resoluciones para adaptarse a las condiciones de red del receptor. Cubre dominios de comunicación de alto nivel, incluyendo voz sobre IP móvil y comunicación de video en tiempo real.
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.
