Based On The WebRTC Audio Conferencing Systems Design And Implementation

In recent years, with the popularity of wireless communication and mobile devices,the market potential of IP media technology is gradually emerging, in which the demand of IP telephone conference system market for mobile terminal is huge. But the traditional real-time audio communications between mobile devices is difficult to spread, it needs to get the functions such as audio media collection, audio signal processing, audio codec and real-time transmission control together and needs to rely on complex and expensive hardware, and the number of supporters of conference members is very limited. To solve the problem of audio conferencing between mobile devices taking full use of IP technology, this paper designs an audio conference system server and achieves SDK encapsulation for users and developers. App developers can integrate the capabilities of audio conference system by calling the SDK interface with encapsulation.Firstly, taking the typical application scenarios for example, this paper analyzes the customer demand of current audio conference system and available IP audio technology solutions, and selects the technology program based on WebRTC, and designs architecture model of audio conference system based on WebRTC.Secondly, this paper realizes the secondary development and encapsulation of WebRTC, since that WebRTC is the processing and mixing of media stream and there is no concept and module of session management. Then this paper establishes correspondence relations between media stream marked with channelid and session marked with sessionid, and designs modules for sessionid management and business processes. The modules of media processing and session management uses SIP signaling to communicate, and realizes the encapsulation of sofia sip to meet the system requirements.Thirdly, this paper achieves various designed modules and corresponding capabilities for appropriate classes, class members and functions.Finally, the paper builds and tests the system environment, testing capabilities of creating meetings, inviting members, venue playback, dissolution of the meeting, and the clarity, delay, reduction degree of voice.