End-to-end packet delay is an important performance parameter on the Internet, because it heavily affects the quality of real-time applications. Currently, however, because the packet transmission quality (e.g., transmission delay, jitter, packet loss) may vary dynamically, it is not easy to handle real-time traffic. For UDP-based real-time applications, a smoothing buffer (playout buffer) is typically used at the client to compensate for variable delays. The issue of playout control has been studied previously, and several algorithms for controlling the playout buffer have been proposed. These studies considered the network parameters (e.g., packet loss ratio and playout delay), but not the quality perceived by end users. We first clarify the relations between mean opinion score (MOS) of played audio and the network parameters (e.g., packet loss, packet transmission delay, and transmission rate). Then, utilizing the MOS function, we propose a new playout buffer algorithm that considers the user's perceived quality for real-time applications. Our simulation and implementation tests show that the algorithm can enhance the perceived quality more effectively than existing algorithms.