react-native-webrtc/react-native-webrtc
React Native Webrtc
react-native-webrtc este un wrapper de framework mobil și o bibliotecă de comunicare peer-to-peer utilizată pentru a integra comunicarea în timp real prin audio, video și date în aplicații mobile cross-platform. Acesta funcționează ca un instrument pentru capturarea ecranelor dispozitivelor și streaming-ul de conținut media live prin protocolul WebRTC.
Proiectul oferă o interfață pentru canale de date bidirecționale, permițând schimbul de pachete de date arbitrare între peer-uri fără un server de releu central. Suportă streaming media în timp real, inclusiv transmisia audio și video peer-to-peer și capabilități de captură de ecran.
Biblioteca include funcționalități pentru optimizarea fluxului, cum ar fi trimiterea mai multor versiuni ale unui flux video la rezoluții diferite pentru a se adapta la condițiile rețelei receptorului. Acoperă domenii de comunicare de nivel înalt, inclusiv voce peste IP mobil și comunicare video în timp 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.
