Abstract
An end-to-end packet delay in the Internet is an important performance parameter, because it heavily affects the quality of real-time applications. In the current Internet, however, because the packet transmission qualities (e.g., transmission delays, jitters, packet losses) may vary dynamically, it is not easy to handle a real-time traffic. In UDP-based real-time applications, a smoothing buffer (playout buffer) is typically used at a client host to compensate for variable delays. The issue of playout control has been studied by some previous works, and several algorithms controlling the playout buffer have been proposed. These studies have controlled the network parameters (e.g., packet loss ratio and playout delay), not considered the quality perceived by users. In this paper, we first clarify the relationship between Mean Opinion Score (MOS) of played audio and network parameters (e.g., packet loss, packet transmission delay, transmission rate). Next, utilizing the MOS function, we propose a new playout buffer algorithm considering user's perceived quality of real-time applications. Our simulation and implementation tests show that it can enhance the perceived quality, compared with existing algorithms.
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Fujimoto, K., Ata, S. & Murata, M. Adaptive Playout Buffer Algorithm for Enhancing Perceived Quality of Streaming Applications. Telecommunication Systems 25, 259–271 (2004). https://doi.org/10.1023/B:TELS.0000014784.20034.74
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DOI: https://doi.org/10.1023/B:TELS.0000014784.20034.74