| Summary: | 碩士 === 臺灣大學 === 電信工程學研究所 === 96 === As wireless networks become more and more popular, VoIP has become one of the promising technologies for personal wireless communication due to its low service rate. However, there are many technical challenges in VoIP. One major challenge is the provision of Quality of Service. Due to the fact that different VoIP codecs can have different performance in the same network environment, it is possible to achieve better QoS by employing adaptive codec switching schemes. In this thesis, we propose a framework to obtain the optimal codec switching scheme for VoIP in a heterogeneous network environment. We first give a thorough introduction on the basic VoIP techniques, popular codecs (G.711 and G.726), and the ITU-T standard speech quality evaluation tool (PESQ). Then, the channel model is established for a heterogeneous network which includes both wired and wireless links. With the channel model, we propose the analytical framework to determine the optimal codec switching thresholds based on the packet loss rate observed. Numerical experiments show that the proposed framework is indeed useful in improving the audio quality
for VoIP in a heterogeneous network environment.
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