在二十一世紀的今天，隨著電腦與電信的逐漸整合，VoIP(Voice over Internet Protocol)以其成本的明顯優勢以及不斷進步的語音品質，逐漸有取代傳統公眾電話交換網路(PSTN：Public Switched Telephone Network)的趨勢。VoIP利用資料封包交換(Packet Switching)的方式，藉由軟、硬體將類比的語音訊號數位化，以資料封包的形式在IP網路上傳遞做交換的動作，它的標準協定有H.323、SIP等。相對於複雜度較高、子協定過多因此在技術上受限的H.323來說，SIP就顯得較有彈性，它在網路架構下的協定標準，能透過Gateway來達成與傳統PSTN互通的目的，完成簡單的網路電話架構。SIP也正逐步成為NGN(下一代網路)和3G多媒體子系統中的重要協定。SIP的主從式架構(Client-Server)雖然簡單，但隨著使用者的增加，伺服器恐會因為長期負擔龐大的資訊流量而發生問題，而依靠伺服器所維持的VoIP網路也將會因而停止服務。
因此在本篇的論文裡，將在在有限的資源下將原本的Client-Server架構做改良，Client端可以不藉由SIP Server的幫助，達成連線通話的目的，以減低伺服器提供服務的負擔，並提升撥打網路電話的效率。

In the 21st century's today, because of the computer and the telecommunication are combined with each other, VOIP (Voice over Internet Protocol) substitute for PSTN(Public Switched Telephone Network) stage by stage according to its cost and the quality of voice. VOIP digitizes analogic voice by software and hardware, and uses Packet Switching to transmit the packets in the IP network, its standard protocol are H.323, SIP, etc. H.323 is more complex and has more sub-protocol such that it is restricted by technicality. To compare with H.323, SIP is free. It can achieve the goal communication with PSTN through Gateway, and constructs a simple structure of VOIP. SIP has become the important protocol of NGN (Next Generation Networks) and 3G mutimedia systems. The structure of Client-Server of SIP is simple, but as long as the users increase, the Server may cause some trouble because of the great deal of information, and the VOIP network which depends on the Server will stop its service.
For this reason, we will improve the Client-Server structure in the paper, we wish that clients could communication with the others without the service of SIP Server to reduce the load, and improve the perfomance of SIP signaling.

中文摘要 I
ABSTRACT II
致謝 III
目次 IV
圖目錄 VII
表目錄 IX
1. 序論 1
1.1. 研究動機 1
1.2. 論文架構 1
2. 網路電話簡介 2
2.1. VOIP通訊協定介紹 2
2.1.1. H.323介紹 2
2.1.2. SIP介紹 3
2.1.3. MGCP介紹 3
2.2. SIP協定介紹 3
2.3. SIP元件介紹 6
2.3.1. 用戶代理(User Agents) 6
2.3.2. 代理伺服器(Proxy Server) 6
2.3.3. 重定向伺服器(Redirect Server) 7
2.3.4. 登錄伺服器(Registrar Server) 8
2.4. SIP訊息介紹 9
2.4.1. SIP訊息格式 9
2.4.2. SIP請求訊息(Request) 10
2.4.3. SIP回應訊息(Response) 11
2.4.4. SIP CALL運作流程 12
2.5. SIP之P2P介紹 15
2.5.1. 集中目錄式網路架構 15
2.5.2. 純P2P網路架構 16
2.5.3. 混合式P2P網路架構 17
2.5.4. 結構化P2P網路架構 18
3. 系統架構 20
3.1. 系統功能 20
3.2. 硬體架構 20
3.2.1. 硬體架構概述 20
3.2.2. VINETIC DSP架構 24
3.3. 軟體架構 26
3.3.1. SIP UA運作流程 27
3.3.2. WEB功能及執行流程 30
3.3.3. Watchdog 32
4. 傳輸效能改善與容錯性之設計 33
4.1. 設計概念及流程 33
4.2. SERVER端架構 34
4.2.1. OpenSIPS介紹 35
4.2.2. Proxy Server之元件設計 35
4.3. CLIENT端之運作流程 38
4.3.1. 從Proxy Server取得資料成功 38
4.3.2. 從Proxy Server取得資料失敗 39
4.4. 虛擬IP與錯誤資料的處理 41
4.4.1. Virtual IP 41
4.4.2. Proxy timer與通話的重新導向 43
4.5. PROXY SERVER之效能改善及容錯 45
5. 結論 46
參考文獻 47

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