Enhancing VoIP speech quality using combined playout control and signal reconstruction

摘要

The quality of real-time Voice over Internet Protocol (VoIP) networks is affected by network impairments such as delays, jitters, and packet loss. To solve this issue, this paper proposes a new receiver-based enhancing method of VoIP speech quality. Our approach is based on the combined playout control and signal reconstruction technique that consists of a set of algorithms that conceal packet loss, reduce buffering delay, detect spike delay, and alleviate packet delay jitter. The proposed fully receiver-based enhancing algorithm is computationally efficient, delivers high-quality voice service, and is suitable for use in any practical mobile VoIP system.