基于DSP+FreeSWITCH的IP语音会议系统研究

Research on IP Voice Conference System Based on DSP + FreeSWITCH

基于DSP+FreeSWITCH架构的语音会议系统进行研究,将SIP信令处理和媒体编解码处理独立开,将易造成性能瓶颈的媒体处理移交外部DSP编解码模块处理,通过更改外部DSP编解码模块数量灵活配置系统支持能力,以适应不同的应用场景,提升语音会议用户体验。

In this paper, the voice conference system based on DSP + FreeSWITCH architecture is studied. SIP signaling processing and media encoding and decoding processing are separated, and the media processing that causes performance bottleneck is transferred to external DSP encoding and decoding module processing. By changing the number of DSP encoding and decoding modules, the system support ability can be flexibly configured to adapt to different application scenarios and improve the user experience of voice conference.