This paper proposes a receiver-based high performance adaptive signal control for enhancing Voice over Internet Protocol (VoIP) speech quality. In the proposed method, the buffering time is minimized by way of playing out normally, expanding or compressing each packet according to adaptive network jitter estimation. And recursive linear prediction-based packet loss concealment using an adaptive muting factor delivers high voice quality by concealing consecutive packet loss. Experimental results show that the proposed algorithm delivers high voice quality by pursuing an optimal trade-off between buffering delay and packet loss rate.
展开▼
