Research Of Packet Buffer Technology In WebRTC Voice Engine

The VoIP technology has become an important and widely used communicationapplication now. However, with the development of mobile internet, mobile VoIP hasbecome a new trend. Traditional VoIP technology will be confronted with difficultycaused by delay and jitter which is different from wired internet.The key factors which affect voice quality are network delay and jitter. In order tominimize the voice quality damage caused by network jitter, buffering is necessary forreceiving packets from internet which experience delay to eliminate jitter. But thebuffering technology may also lead to more delay to the whole voice communication,especially through3G network between different operator and regions, the delay willgrow very high due to the large jitter.. Thus, the selection of buffer size and buffercontrol algorithm will encounter new challenge.In this paper, the principle and mechanism of buffering in WebRTC voice enginewere studied firstly, and the issue of large voice delay within some3G networks wasalso analyzed. To solve this problem, a solution was proposed to decrease voice delayby optimize buffering mechanism of WebRTC, which extend the usage of VADtechnology from improving compression efficiency in encoder to different bufferingalgorithm in decoder by determine whether there is voice data in a packet. The solutioncan decrease the overall voice delay and improve the voice quality of VoIP.