臺灣博碩士論文加值系統

隨著網路的普及，網路的應用也日益增加，VoIP在網路應用上更是發展迅速，且語音品質更是可以和傳統電話一較高下。此外，網路電話在使用上不需負擔高額的通話費，只要區域性網路的線路費用即可，因此有越來越多人使用網路電話。而在滿足基本的通話需求後，使用者也會要求能夠有更多的附加服務。
在本篇論文中，針對SIP系統下建立一套語音郵件的服務，讓使用者能夠以數位的方式，將漏接的電話以錄製語音方式保留下來，再透過E-mail聽取留言。同儕網路電話則是用最少的金錢與人力，不須建置伺服器方式，讓小型社群之間可以使用網路電話來互相聯絡通話。

With the network popularization, applications on network are increasing day by day. VoIP (Voice over IP) is one of exciting developments on network applications, and its voice quality has been as good as traditional phones. Furthermore, IP phone users do not afford expensive communication costs, but go to the expense of broadband service. Therefore, more and more people start to use IP phone. After meeting certain requirements of basic communication, users could ask for more additional services.
In the thesis, a voicemail service is built under SIP (Session Initiation Protocol). Users can keep missing call by recording voice with digital files, and then, one can receive recorded files from E-mail. Besides, small social networks can communicate with IP phones by using pear-to-pear telephony, which costs less money and labor due to no cost on building servers.

摘要 I
ABSTRACT II
目次 III
圖目錄 V
表目錄 VI
1. 序論 1
1.1. 研究動機與目的 1
1.2. 論文架構 1
2. SIP概論與介紹 2
2.1. 網路電話介紹 2
2.2. SIP介紹 3
2.3. SIP 元件 6
2.4. SIP訊息 9
2.4.1. SIP 請求訊息 9
2.4.2. SIP 回應訊息 10
2.5. SIP 運作流程 11
3. 系統架構 13
3.1. 系統功能 13
3.2. 硬體架構 13
3.3. 軟體架構 17
3.4. SIP UA 軟體運作流程 18
4. 研究與實做 21
4.1. 伺服器端 21
4.1.1. OpenSIPS介紹 21
4.1.2. opensips.cfg 21
4.1.3. FreeSWITCH介紹 23
4.1.4. 伺服器設定 24
4.1.5. 實際系統架構 27
4.2. 語音郵件 28
4.3. 同儕網路設定 31
4.4. 同儕網路之實作 34
5. 結論 37

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[5]FreeSWITCH Server, http://wiki.freeswitch.org/
[6]H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications”, RFC-355
[7]J. Franks, P. Hallam-Baker, J. Hostetler, S. Lawrence, P. Leach, A. Luotonen, L. Stewart, “HTTP Authentication: Basic and Digest Access Authentication”,RFC 2617, June 1999.
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