臺灣博碩士論文加值系統

在VoIP(Voice over IP)網路環境下若要與傳統PSTN(Public Switched Telephone Network)電信系統互連，則須透過閘道器(Gateway)或是類比電話適配器(Analog Telephone Adapter, ATA) 等設備作語音信令轉換，以實現PSTN到VoIP(VoIP上車)或VoIP到PSTN(VoIP下車)的轉換，進而達到節省話費的功能。而這功能可以透過具有IP PBX(IP Private Branch eXchange)搭配語音電話卡(Telephone Card)實現PSTN與VoIP之間兩種異質網路的通話服務。

本論文嘗試在有線網路環境，以SIP(Session Imitation Protocol)為基礎，實作一個低成本高效率的網路類比電話閘道兼適配器。透過此裝置得以電話號碼為基礎，可以撥打到各種網路電話終端設備。我們以卓群CM5000為系統開發平台，在上層系統功能上完成SIP-Based信令轉換機制與PSTN故有的電話服務(如通話保留/取回、呼叫前轉接、呼叫後轉接、代接呼叫、一號多機、音樂保留、群組響鈴呼叫)，而語音壓縮資料的轉換則是透過底層硬體實現。

在系統有限資源下，此裝置提供SIP與PSTN電信網路轉換，並且也提供與PSTN電信網路一樣的多功能通話服務。其優點為應用於一般家庭或是小型企業時，可降低通話成本，亦讓此裝置能提供語音和資料傳輸功能。

In VoIP(Voice over IP) network environment connecting between PSTN(Public Switched Telephone Network) and VoIP is usually done through gateway or ATA(Analog Telephone Adapter) for voice signaling conversion. This function can be achieved by having IP PBX(IP Private Branch eXchange)with telephone card.

In this paper, we attempt to implement a cost-effective analog telephone adapter based on SIP via wired network, so you can call to a variety of SIP user agents through the conventional telephone system. Our development is based on CM5000 platform to complete the signaling conversions between SIP and PSTN. Provisioning of call features, such as call hold/retrieve, call forward, call transfer, call pick-up, fine me, music on hold, ring group, and voice compression and data conversion are accomplished through the underlying hardware implementation.

Of limited resources, this adapter can accomplish signaling conversion between SIP and PSTN, and also provides most common call features as supported in PSTN. The advantage of this device is convenient to apply in general residence or small enterprise with cost reduction as well as the provision of voice and data transmission.

摘要 i
Abstract ii
目次 iii
圖目錄 v
表目錄 vii
第一章 緒論 1
1.1 研究背景與動機 1
1.2 論文貢獻 1
1.3 論文架構 2
第二章 相關研究及技術 3
2.1 PSTN與ISUP 3
2.1.1 PSTN介紹 3
2.1.2 SS7介紹 4
2.2 VoIP與SIP 7
2.2.1 VoIP介紹 7
2.2.2 SIP介紹 8
2.2.3 SDP介紹 9
2.2.4 SIP基本邏輯元件 10
2.2.5 SIP訊息 11
2.2.6 SIP基本呼叫 13
2.2.7 SIP進階呼叫 14
2.3 SIP與PSTN網路互通流程 21
2.4 相關研究討論 22
2.4.1 VoIP網路架構 22
2.4.2 VoIP與PSTN連接 23
2.4.3 VoIP商業模式與晶片市場 25
第三章 系統設計與架構 29
3.1 硬體系統架構 29
3.1.1 硬體模組設計原理 29
3.1.2 硬體系統結構 30
3.1.3 硬體周邊連結 36
3.2 軟體系統架構 37
3.2.1 軟體模組設計原理 37
3.2.2 軟體系統結構 38
3.2.3 軟體系統狀態機 39
第四章 系統建置與實驗分析 43
4.1 系統環境建置 43
4.1.1 安裝VoIP軟體 43
4.1.2 佈署VoIP網路電話環境 43
4.2 實驗結果分析 46
4.2.1 VoIP上車 46
4.2.2 VoIP下車 47
4.2.3 SIP呼叫保留/取回(Call Hold/Retrieve) 48
4.2.4 SIP呼叫前轉接(Call Forward) 50
4.2.5 SIP呼叫後轉接(Call Transfer) 51
4.2.6 SIP代接呼叫(Call Pick-up) 53
4.2.7 SIP一號多機功能(Find Me) 54
4.2.8 SIP音樂保留功能(Music on Hold) 56
4.2.9 SIP群組響鈴呼叫(Ring Group) 57
第五章 結論與未來展望 59
參考文獻 60

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