| Summary: | 碩士 === 國立暨南國際大學 === 資訊工程學系 === 98 === Since the Voice over Internet Protocol (VoIP) was proposed, the issue of voice quality is always a big challenge. When more and more people use VoIP applications, the quality issue becomes critical. Basically, the measurement of voice quality is to test on the client-side. However, in the real network, it is unreasonable to control clients during the communication time.
In recent years, because of the exhaustion of IP addresses, Network Address Translator (NAT) was proposed to solve the shortage of IP addresses. Nevertheless, NAT fails some peer-to-peer Internet applications, such as VoIP. So, VoIP applications need solutions for NAT traversal. There are lots of NAT traversal mechanisms suggested, such as static assignment, Virtual Private Network (VPN), and relay-based proxy server. RTP Proxy Server is a relay-based proxy server, which is the most popular one among these NAT traversal mechanisms. Now in most VoIP architectures, there exists a RTP Proxy Server for relaying RTP packets.
In this thesis, we design a monitoring system, named RTP-M which works with RTP Proxy Server, to measure the VoIP quality. Additionally, RTP-M represents the measured result as graphs which are more intuitive for human beings. We hope that our RTP-M can provide administrators of VoIP service with the troubleshooting reference.
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