網路電話是現今世代的網路技術中重要應用之一，隨著Internet普及化及寬頻化的發展，頻寬越變越大，大部份台灣家庭中也有了網路，這時人們除了選擇傳統PSTN電信網路，也可選擇更優惠的VoIP網路電話。
多方會議是VoIP的眾多功能之一，但並非所有的Server都有支援它，以國立中山大學所使用的 OpenSIPS Server來說，就沒有支援這項功能，但FreeSWITCH Server上，卻有這樣的功能，因此我藉由整合這兩個Servers，以達到OpenSIPS也可以使用多方會議。
本論文除了著眼於兩個Servers的整合，同時也另外開發了一個網頁介面，可以讓使用者利用這個介面來達成定時啟用多方會議的功能，時間一到與會者們的電話就會響起，接起電話就可進入多方會議的通話，以簡化進入多方會議的程序。

VoIP technology is one of the important applications of the network. In addition to using traditional PSTN telephone, people can choose favorable VoIP telephone because of the Internet popularization and High speed Internet.
Conference is one of the functions of the VoIP, but not every server supported it. Such as OpenSIPS server we use in NSYSU, didn’t supported it. But FreeSWITCH Server has it. Therefore, for solving this problem, I combile two Servers. Then OpenSIPS can use the Conference function.
In this paper, we discuss how to combile OpenSIPS and FreeSWITCH. Besides, I have designed a WEB page interface to simplify the procedure to use conference. By this interface, the users can use it to set the timer to start conference. When the timer is up, all of the participants will receive a conference call. After picking up the phone, the users can enter conference.

致謝 II
中文摘要 I
ABSTRACT II
目次 III
圖目錄 V
表目錄 VII
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 3
2.1. SIP概論與元件 3
2.2. 客戶端 5
2.3. 伺服器端 5
2.3.1 代理伺服器（Proxy Server） 5
2.3.2. 註冊伺服器（Register Server） 6
2.3.3. 重新導向伺服器（Redirect Server） 7
2.4. SIP訊息 7
2.4.1. SIP請求訊息 8
2.4.2 SIP回應訊息 9
2.5. 會議描述協議(SDP) 10
2.6. SIP運作流程 12
2.7. RTP即時傳輸協定(Real-Time Transport Protocol) 14
2.8. 多方會議（SIP Conference） 16
2.8.1 CONF請求 16
2.8.2 SIP對多方會議呼叫的支援 17
2.8.3 多人語音混音方法 21
2.8.4 線性疊加法 22
2.8.5 箝位法 22
2.8.6 平均法 24
2.8.7 正規化法 24
2.8.8 正規因子法 25
2.9. FreeSWITCH 26
2.9.1 FreeSWITCH安裝步驟 26
2.9.2 FreeSWITCH 各項目錄 27
2.9.3 FreeSWITCH 與OpenSIPS的整合 28
3. 開發環境 31
3.1. 伺服器端 31
3.2. 客戶端 32
3.2.1. 網路電話硬體架構 33
3.2.2. 網路電話軟體架構 37
3.2.3. SIP UA軟體運作流程 40
4. 系統設計與實作 43
4.1. FreeSWITCH多方會議介紹以及封包流程 43
4.2. 系統設計之流程 47
4.3. 網頁介面實做 50
4.4. 多方會議的建立 56
5. 結論 60
參考文獻 61

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[7] H. Schulzrinne, S. Petrack, “RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals”, RFC-2833, May 2000.
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