隨著網路頻寬成長與語音壓縮技術的進步，VoIP(Voice over IP)在近幾年來變的非常的熱門。雖然VoIP有極加的通話品質並且擁有很大的價格優勢，但是他在未來幾年內仍然不可能完全取代PSTN(Public Switched telephone Network)，因此我們希望能夠解決PSTN與VoIP之間連接上的問題。

在這篇論文中，我們將探討如果在英飛凌(Infineon) EASY 5120發版本上設計與實作VoIP閘道器(Gateway)，這個閘道器擁有功能強大的網路處理器ADM5120可以處理網路封包和專門處理數位和類比訊號的DSP(Digital Signal Processor)。這閘道器運作的方式是PSTN端的類比訊號從FXO(Foreign eXchange Office)送到VINETIC DSP中轉成位數訊號再透過網際網路與VoIP的客戶端進行通話。這個VoIP閘道器可以解決PSTN與VoIP兩端連接的問題。

另外為了解決SIP通訊協定穿透NAT的問題上，提出與實作整合性可以穿透任何種類NAT的解決方法，我們使用STUN(Simple Traversal of UDP through NATs)並配合伺服器做Relay的方法，不只可以穿透任何種類的NAT，而且客戶端系統可以自動偵測網路狀態而做出正確的設定，同時能保有良好的擴充性與通話品質。

With the bandwidth growing and voice coding technology enhancing, VoIP(Voice over IP)becomes a very popular application in recent years. Although VoIP has good voice quality and huge price advantage, it can't replace PSTN(Public Switched Telephone Network) completely in the next few years. We therefore hope to be able to solve the connectivity issue between PSTN and VoIP.

In this thesis, I design and implement a VoIP Gateway using Infineon EASY 5120 platform which has a powerful CPU with network processor, ADM5120, and has a DSP (Digital Signal Processor), Vinetic-2CPE PEB3332, to handle digital and analog voice signals. The operation method of the gateway is that PSTN analog signals are passed to Vinetic-2CPE from FXO (Foreign eXchange Office) port, and they are transferred into digital signals to be packetized to transit into Internet to another VoIP Client. My VoIP Gateway can solve the interconnection issues between PSTN and VoIP.

Furthermore, in order to solve NAT traversal problem of SIP protocol, we propose and implement an integration approach which uses STUN (Simple Traversal of UDP through NAT) to cooperate with RTP Relay server to traverse various types of NAT, especial symmetric NAT. My VoIP Gateway not only can traverse any type of NAT but also can perform self-configuration according to STUN test report in various network environments.

1 緒論 4
1.1 動機與目的 4
1.2 論文架構 5

2 網路電話簡介 6
2.1 SIP元件介紹 7
2.1.1 客戶端 7
2.1.2 伺服器 7
2.2 SIP請求訊息 8
2.3 SIP回應訊息 9
2.4 SIP運作流程 9

3 客戶端架構與功能 16
3.1 客戶端架構 16
3.1.1 硬體架構 16
3.1.2 軟體架構 18
3.2 客戶端Forwarding功能 21
3.2.1 VoIP to PSTN Forwarding功能 21
3.2.2 PSTN to VoIP Forwarding功能 26
3.2.3 VoIP to VoIP Forwarding功能 29
3.3 客戶端Watchdog功能 31
3.3.1 軟體Watchdog 31
3.3.2 軟體Watchdog 33

4 實現SIP通過NAT的方法 36
4.1 什麼是NAT(Network Address Translation)? 36
4.2 SIP過過NAT時所發生的問題 36
4.3 NAT型態 37
4.3.1 Full Cone NAT 38
4.3.2 Restricted Cone NAT 39
4.3.3 Restricted Port NAT 40
4.3.4 Symmetric NAT 40
4.4 目前各種穿透NAT的方法 41
4.4.1 通用隨插即用(Universal Plug and Play) 41
4.4.2 STUN(Simple Traversal of UDP through NAT) 42
4.4.3 RTP Relay 44
4.5 我們實作的方法：使用客戶端的STUN配合伺服器端的RTP Relay 44
4.5.1 利用STUN辦認不同的NAT Server的型態 45
4.5.2 針對不同的型態決定是否啟用STUN 46
4.5.3 RTP Relay伺服器的架構 49
4.5.4 配合RTP Relay 50
4.5.5 問題與解決方法 52

5 伺服器端的架構與功能 53
5.1 伺服器端架構 53
5.2 使用web page申請SIP帳號 53
5.3 call out/in service 55
5.3.1 身份確認的問題 55
5.3.2 fix_to module 55
5.3.3 call in 57
5.4 網路電話系統與校園分機整合 57
5.4.1 網路電話系統到校園分機系統 57
5.4.2 校園分機到網路電話系統 57
5.5 網路會議(conference)和語音信箱(voice mail) 58

6 結論 59

[1] M. Handley, H. Schulzrinne, E. Schooler,J. Rosenberg ”SIP: Session Initiation Protocol”, RFC 2543, March 1999

[2] J. Rosenberg, H. Schulzrinne, G. Camarillo,A. Johnston,J. Peterson,R. Sparks,M. Handley,E. Schooler ”SIP: Session Initiation Protocol”, RFC 3261, June 2002

[3] Handley, M. and V. Jacobson, ”SDP: Session Description Protocol”, RFC 2327, April 1998.

[4] J. Rosenberg, J. Weinberger, C. Huitema, R. Mahy ”STUN - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)”, RFC 3489, March 2003

[5] P. Srisuresh, M. Holdrege ”IP Network Address Translator (NAT) Terminology and Considerations”, RFC 2663, August 1999.

[6] B. Ford, P. Srisuresh, D. Kegel ”Peer-to-Peer (P2P) communication across middleboxes”, Internet-Draft, October 2003

[7] M. Arango, A. Dugan, I. Elliott, C. Huitema, S. Pickett ”Media Gateway Control Protocol (MGCP)”, RFC 2705, October 1999

[8] Jiri Kuthan, Jan Janak, Yacine Rebahi ”iptel.org SIP Express Router v0.11.0 – Admin’s Guide”, 2002

[9] ”ADM 5120 HOME GATEWAY CONTROLLER Datasheet”, AMDtek Inc. , 2003.

[10] ”EASY 5120 2-Channel VoIP Router Reference Package Hardware Description”, Infineon Technologies.

[11] ”Telephony Application Programming Interface for Infineon Voice Codec Devices”, Infineon Technologies.

[12] ”EASY 5120P-ATA Fxo System Description”, Infineon Technologies, Jul. 2004.

[13] iptel.org SIP Server:SIP Express Router, http://www.iptel.org/ser/

[14] ag-projects.com:MediaProxy, http://www.ag-projects.com/MediaProxy.html

[15] RP-PPPOE,http://www.roaringpenguin.com/penguin/open_source_rppppoe.php

[16] DHCP,http://udhcp.busybox.net/

[17] The eXtended osip library, http://savannah.nongnu.org/projects/exosip

[18] The GNU oSIP library, http://www.gnu.org/software/osip/