aiortc/aiortc
Aiortc
aiortc 是 WebRTC 协议的 Python 实现，为对等点之间的实时音频、视频和数据流提供异步栈。它作为一个媒体引擎和数据通道实现，通过非阻塞框架管理对等连接。
该库通过协商会话描述和连接路径，实现直接对等网络连接的建立。它通过数据通道促进安全、双向的数据传输，并支持使用标准编解码器进行低延迟媒体流传输。
该系统包括网络 NAT 穿透和会话管理功能。安全性通过安全通信通道设置来处理，这涉及生成密钥和证书以加密所有媒体和数据流量。
Features
- Peer-to-Peer Networking - Provides a complete implementation of peer-to-peer networking for direct client-to-client data and media exchange.
- WebRTC Implementations - Building applications that require secure, encrypted exchange of media and data streams using the WebRTC protocol.
- Standard Media Codecs - Provides support for standard audio and video codecs to encode and decode media frames for transmission.
- WebRTC Protocol Stack Management - Provides a full WebRTC protocol stack for managing peer connections and media transport.
- NAT Traversal Mechanisms - Includes NAT traversal mechanisms to connect remote peers through firewalls and address translators.
- WebRTC Data Channels - Implements secure, bidirectional WebRTC data channels for exchanging arbitrary binary and text data.
- Connectivity Negotiators - Manages the negotiation of connectivity paths and session descriptions to establish peer links.
- SDP Negotiation - Implements the Session Description Protocol (SDP) for negotiating media capabilities and transport addresses.
- Real-Time Media Transport - Implements low-latency transport systems for the bidirectional exchange of real-time audio and video streams.
- Secure Communication Channels - Provides the underlying secure communication channels required for all WebRTC media and data traffic.
- WebRTC Media Orchestration - Functions as a media engine for orchestrating the encoding, decoding, and transport of real-time streams.
- ICE-Based Peer Connection Managers - Uses ICE, STUN, and TURN servers to discover network addresses and establish peer-to-peer connectivity.
- WebRTC Python Implementations - Provides a comprehensive Python implementation of the WebRTC protocol for audio, video, and data streaming.
- WebRTC Session Management - Implements the essential logic for creating and managing asynchronous WebRTC sessions for media and data exchange.
- WebRTC Streaming - Facilitates low-latency real-time streaming of audio and video between peers using the WebRTC protocol.
- Key Exchange Protocols - Implements DTLS-based key exchange protocols to secure media and data streams between peers.
- Stream Encryption and Decryption - Implements real-time encryption and decryption of audio, video, and data streams using secure handshakes.
- DTLS-SRTP Configurations - Provides DTLS-SRTP configurations to encrypt real-time media packets and prevent eavesdropping.
- Asynchronous Event Loops - Implements an asynchronous event loop based on asyncio for non-blocking network I/O and concurrency.
- SCTP-over-UDP Tunneling - Implements SCTP-over-UDP tunneling to ensure reliable data channel transmission through network firewalls.
- Real-time Data Packet Transmission - Enables bidirectional transmission of raw data packets using reliable stream control protocols.
- Streaming Protocols & Transport - Python library implementing WebRTC and ORTC for real-time media and data streaming.
- Python Libraries - Listed in the “Python Libraries” section of the Awesome Rtc awesome list.
