| Summary: | 碩士 === 國立彰化師範大學 === 數位學習研究所 === 98 === Discovered in the study popularizing with various wireless networks such as Wi-Fi or 3G network in recent years, the development of VoIP (Voice over Internet Protocol, VoIP) business already becomes maturity stage. Especially, the convenience of mobile device provides human's communication each other in any place, exchange information in any time and business activity. This study implements VoIP using hight compatibility protocol RFC 3261 (Session Initial Protocol, SIP) that release by ITU-T as end user communication protocol, and use RFC 3550 (Real-time Transport Protocol, RTP) to carry multimedia data. Due to the limit of bandwidth in wireless network, the transfer voice data must to be compressed, so that we used G.711 voice compression algorithm that be established by ITU-T to encode/decode voice data. Most of mobile device usually locate behind the NAT/Firewall in wireless network, several problems be occured. It be hard for the user communication with other users. To deal with the problem of NAT/Firewall, solution be used as Simple Traversal of User Datagram Protocol through Network Address Translators (STUN). In order to solved the problems of packet lost and voice jitter, this application implement Packet Loss Concealment algorithm and jitter buffer to reduced phenomenon of voice jitter and made quality of voice smoother. This study provided a application program of mobile VoIP that have mobility、security、high quality of voice and high economic benefits to help human communication conveniently.
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