摘 要
本文针对基于WebRTC的实时音视频通信性能优化进行研究。首先对WebRTC技术进行了综述，介绍了WebRTC架构、通信协议和信令服务器。接着对WebRTC实时音视频通信机制进行了详细描述，包括音视频媒体协商、媒体采集和编码、媒体传输以及媒体解码和渲染。针对实时音视频传输中存在的性能瓶颈，提出了视频编码参数优化、媒体流控制算法、码率自适应算法和传输层协议优化等方法，可有效提高音视频通信质量和稳定性。研究表明，本文提出的优化方法能够有效改善WebRTC实时音视频通信的性能，提高了音视频质量和传输稳定性。最后，总结了本文的研究成果并展望了相关研究的未来发展方向。
关键词：WebRTC、实时音视频通信、性能优化
Abstract
The performance optimization of real-time audio and video communication based on WebRTC is studied in this paper. Firstly, WebRTC technology is summarized, and the architecture, communication protocol and signaling server of WebRTC are introduced. Then the real-time audio and video communication mechanism of WebRTC is described in detail, including audio and video media negotiation, media acquisition and coding, media transmission, media decoding and rendering. Aiming at the performance bottleneck in real-time audio and video transmission, some methods such as video coding parameter optimization, media stream control algorithm, bit rate adaptive algorithm and transport layer protocol optimization are proposed, which can effectively improve the quality and stability of audio and video communication. The results show that the proposed optimization method can effectively improve the performance of real-time audio and video communication of WebRTC, improve audio and video quality and transmission stability. Finally, the research results of this paper are summarized and the future development direction of related research is prospected.
Keyword: WebRTC、Real-time audio and video communication、 Performance optimization
目 录
引言 1
1WebRTC技术综述 1
1.1 WebRTC技术简介 1
1.2 WebRTC架构 2
1.3 WebRTC通信协议 2
1.4 WebRTC信令服务器 3
2 WebRTC实时音视频通信机制 4
2.1 WebRTC音视频媒体协商 4
2.2 WebRTC媒体采集和编码 4
2.3WebRTC媒体传输 5
2.4WebRTC媒体解码和渲染 6
3WebRTC实时音视频通信性能优化方法 6
3.1 视频编码参数优化 6
3.2 媒体流控制算法 7
3.3 码率自适应算法 7
3.4 传输层协议优化 8
结语 8
参考文献 10
致谢 11
本文针对基于WebRTC的实时音视频通信性能优化进行研究。首先对WebRTC技术进行了综述，介绍了WebRTC架构、通信协议和信令服务器。接着对WebRTC实时音视频通信机制进行了详细描述，包括音视频媒体协商、媒体采集和编码、媒体传输以及媒体解码和渲染。针对实时音视频传输中存在的性能瓶颈，提出了视频编码参数优化、媒体流控制算法、码率自适应算法和传输层协议优化等方法，可有效提高音视频通信质量和稳定性。研究表明，本文提出的优化方法能够有效改善WebRTC实时音视频通信的性能，提高了音视频质量和传输稳定性。最后，总结了本文的研究成果并展望了相关研究的未来发展方向。
关键词：WebRTC、实时音视频通信、性能优化
Abstract
The performance optimization of real-time audio and video communication based on WebRTC is studied in this paper. Firstly, WebRTC technology is summarized, and the architecture, communication protocol and signaling server of WebRTC are introduced. Then the real-time audio and video communication mechanism of WebRTC is described in detail, including audio and video media negotiation, media acquisition and coding, media transmission, media decoding and rendering. Aiming at the performance bottleneck in real-time audio and video transmission, some methods such as video coding parameter optimization, media stream control algorithm, bit rate adaptive algorithm and transport layer protocol optimization are proposed, which can effectively improve the quality and stability of audio and video communication. The results show that the proposed optimization method can effectively improve the performance of real-time audio and video communication of WebRTC, improve audio and video quality and transmission stability. Finally, the research results of this paper are summarized and the future development direction of related research is prospected.
Keyword: WebRTC、Real-time audio and video communication、 Performance optimization
目 录
引言 1
1WebRTC技术综述 1
1.1 WebRTC技术简介 1
1.2 WebRTC架构 2
1.3 WebRTC通信协议 2
1.4 WebRTC信令服务器 3
2 WebRTC实时音视频通信机制 4
2.1 WebRTC音视频媒体协商 4
2.2 WebRTC媒体采集和编码 4
2.3WebRTC媒体传输 5
2.4WebRTC媒体解码和渲染 6
3WebRTC实时音视频通信性能优化方法 6
3.1 视频编码参数优化 6
3.2 媒体流控制算法 7
3.3 码率自适应算法 7
3.4 传输层协议优化 8
结语 8
参考文献 10
致谢 11
