| Summary: | 碩士 === 銘傳大學 === 資訊工程學系碩士班 === 95 === Due to the prevalent access of Internet and its diversified network applications, Internet apparently becomes a part of our daily life. The VoIP (Voice over IP) is one of the best-known network applications, which provides free real-time voice communications between world-wide users. Individuals, companies and international enterprises can benefit from VoIP due the money saving from telephony calls of telecommunications.
Recently, technologies for VoIP have been well developed. As the IEEE 802.11 standards for wireless LAN are further improved. One of the improvements is that the bandwidth has been substantially increased. The mobile voice conversation provided by wireless VoIP will be more attractive for users. However, the challenge to mobile VoIP is the quality of voice communications due to packet delay and loss preserved in the wireless LAN. While the mobile VoIP user handoff to another wireless network, the voice communication will suffer the longer delay time and the larger amount of packet loss during the handoff process. The quality of voice communications will be degraded. In this paper, we will propose an effective method to reduce this kind of quality degradation of voice communications in mobile VoIP.
In this research, we will not only construct a mobile VoIP test-bed to apply a scheme to reduce the handoff latency in layer 3, but also couple with improved previous PPT (Predictive Packet Transmission) to further recover the lost packets from the handoff. This proposed scheme is called P3T (i.e. Pre-registration & Predictive Packet Transmission). We will do the best to maintain the quality of real-time voice communications for mobile VoIP. In this paper, we will not only validate our proposed scheme via a theoretic analytic model, but also to further apply the E-Model to demonstrate the objective perceptual quality of voice conversation after the improvement.
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