SIP (Session Initiation Protocol, SIP) 為一種協助多媒體會議 (multimedia session) 召開之通訊協定信號標準，因其具有簡單、彈性、具擴充性等優點，今日已被多數國際通訊大廠採用作為建構下一代以 IP 為核心的網路電話以及新一代語音服務之基礎。但當我們想要建置以SIP為基礎的網路電話服務時會碰到一個問題，就是各種廠商的SIP service logic不能互通。換句話說，因為目前並沒有標準來定義及規範SIP伺服器和service logic的溝通方式，所以每家廠商都用自己的方法來溝通，導致各家產品不能互通。以VOCAL的service logic為例，網路管理者必需調整SIP代理伺服器(proxy server)的參數，才能使其service logic可以運作。而如果有一個SIP代理伺服器並沒有提供像VOCAL的參數調整界面，那麼該SIP代理伺服器勢必不能和VOCAL的service logic合作了。這樣一來，服務建置者勢必為每一個廠商的SIP系統量身訂做其service logic，這是一件非常沒有效率的事。所以，本研究將提出一個方法，可以讓service logic輕易的移殖到任何的SIP環境和任何的SIP產品合作。本研究實際實作了一個service logic伺服器，並且以此伺服器例舉出三個實際的運作例子來証明本研究的方法可行。並且，本研究還做了一系列的效能及其它因子的評估來顯示本研究的方法不僅可以使service logic有可移殖性，且可以在很多方面運作得跟其它方法一樣好。

Session Initiation Protocol (SIP) is a kind of signaling protocol in Internet. SIP is simpler than other signaling protocols, such as H.323. Creating some telephony services in SIP is easier than in PSTN. So, we pick up SIP to build the Internet telephone network.
When we start to build the Internet telephone network environment, we meet a problem “the service logic is lack of portability” in respect of telephony services creation. Taking the VOCAL Feature Server mechanism as an example, we must configure the SIP server to tell it where the service logic is. If other SIP servers without the ability of configuration, they can not cooperate with the service logic of VOCAL. So, we propose an approach to solve this problem in this thesis. The core of this approach is that we make all SIP transactions (that is related with end users in service domain) pass through the service logic server automatically. It is unnecessary to configure the SIP servers in our approach. So, whether the SIP server with the ability to configure or not, our service logic can work with it. In short, our service logic has the portability. In this thesis, we also implement three real application examples using our approach to demonstrate our proposal.
Finally, we make some comparisons and evaluations between our approach and others. This evaluation shows that our approach can run as good as others in many aspects, and our approach has the biggest advantage- portability.

References
Journal and Conference Papers
1. Adamopoulos, D.X., Pavlou, G., Papandreou, C.A., “Advanced service creation using distributed object technology”, IEEE Communications Magazine , Volume: 40 Issue: 3 , March 2002, Page(s): 146 –154
2. Anjum, F. et al., “ChaiTime: A System for Rapid Creation of Portable NextGeneration Telephony Services Using Third-Party Software Components“, In Proc. of the 2 nd IEEE Conference on Open Architectures and Network Programming (OPENARCH), New York, USA, March 1999, access from http://citeseer.nj.nec.com/
3. Bessler, S., Nisanyan, A.V., Peterbauer, K.; "A Service Platform for Internet-Telecom Services using SIP", 6th IFIP conference on Intelligence in Networks, SMARTNET, 2000, access from http://citeseer.nj.nec.com/
4. Dalgic, I. et al., “True number portability and advanced call screening in a SIP-based IP telephony system”, IEEE Communications Magazine , Volume: 37 Issue: 7 , July 1999, Page(s): 96 –101
5. Evloguieva, E., Glitho, R.H., “Plugging in SCPs in SIP networks”, Intelligent Network Workshop, 2000. Proceedings. 2000 IEEE , 2000, Page(s): 90 –94
6. Flykt, P., Alakoski, T., “SIP services and interworking with IPv6”, 3G Mobile Communication Technologies, 2001. Second International Conference on (Conf. Publ. No. 477) , 2001 , Page(s): 186 –190
7. Glitho, R.H., Poulin, A., Khendek, F., “A high level service creation environment for parlay in a SIP environment”, Communications, 2002. ICC 2002. IEEE International Conference on , Volume: 4 , 2002, Page(s): 2008 –2013
8. Glasmann, J., Kellerer, W., Muller, H., “Service development and deployment in H.323 and SIP”, Computers and Communications, 2001. Proceedings. Sixth IEEE Symposium on , 2001, Page(s): 378 –385
9. Hong Liu, Mouchtaris, P., “Voice over IP signaling: H.323 and beyond”, IEEE Communications Magazine , Volume: 38 Issue: 10 , Oct. 2000, Page(s): 142 –148
10. Hua Zou et al., “Prototyping SIP-based VoIP services in Java”, Communication Technology Proceedings, 2000. WCC - ICCT 2000. International Conference on , Volume: 2 , 2000, Page(s): 1395 -1399 vol.2
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12. Kellerer, W., “Intelligence on top of the networks: SIP based service control layer signaling”, Intelligent Network Workshop, 2001 IEEE , 2001, Page(s): 237 –244
13. Kolberg, M.; and Magill, E.H., “Handling incompatibilities between services deployed on IP-based networks”, Intelligent Network Workshop, 2001 IEEE , 2001 , Page(s): 360 -370
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16. Rosenberg, J., Lennox, J., Schulzrinne, H., “Programming Internet telephony services”, IEEE Internet Computing , Volume: 3 Issue: 3 , May-June 1999, Page(s): 63 –72
17. Rudolf, P. and Johannes, S., “A Service Framework for Carrier Grade Multimedia Services using PARLAY APIs over a SIP System”, Wireless Mobile Internet, 2001, Page(s): 69-75
18. Wendong Wang and Shiduan Cheng, “Accessing traditional intelligent services from SIP network”, Info-tech and Info-net, 2001. Proceedings. ICII 2001 - Beijing. 2001 International Conferences on , Volume: 2 , 2001, Page(s): 772 -778
19. Stojsic, G., Radovic, R., Srbljic, S., “Formal Definition of SIP End System Behavior”, EUROCON''2001, Trends in Communications, International Conference on. , Volume: 2 , 2001, Page(s): 293-296
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Internet Drafts and Request for Comments
21. Donovan, S., “The SIP INFO Method “, Request for Comments 2976, IETF, Oct. 2000
22. Handley, M. et al., “SIP: session initiation protocol”, Request for Comments 2543, IETF, Mar. 1999.
23. Handley, M. and Jacobson, V., “SDP: Session Description Protocol “, Request for Comments 2327, IETF, Apr. 1998
24. Johnston, A., “SIP Call Flow Examples”, Internet Draft<draft-ietf-sipping-call-flows-00.txt>, IETF, 2002, Working in Progress
25. Johnston, A., “SIP Service Examples”, Internet Draft<draft-ietf-sipping-service-examples-01.txt>, IETF, 2002, Working in Progress
26. Lennox, J. et al., “Common Gateway Interface for SIP “, Request for Comments 3050, IETF, Jan. 2001.
27. Lennox, J. and Schulzrinne, H., “Call Processing Language Framework and Requirements”, Request for Comments 2824, May 2000
28. Lennox, J. and Schulzrinne, H., “CPL: A Language for User Control of Internet Telephony Services”, Internet Draft< draft-ietf-iptel-cpl-06.txt>, IETF, 2002, Working in Progress
29. Petrack, S. and Conroy, L., “The PINT Service Protocol: Extensions to SIP and SDP for IP Access to Telephone Call Services “, Request for Comments2848, IETF, Jun 2000
30. Roach, A., “SIP-Specific Event Notification”,Internet Draft<draft-ietf-sip-events-05.txt>, IETF, 2002, Working in Progress
31. Rosenberg, J. and Schulzrinne, H., “Guidelines for Authors of Extensions to the Session Initiation Protocol (SIP)”, Internet Draft<draft-ietf-sip-guidelines-05.txt>, IETF, 2002, Working in Progress
32. Rosenberg, J. and Schulzrinne, H., “SIP: Locating SIP Servers”,Internet Draft<draft-ietf-sip-srv-06.txt>, IETF, 2002, Working in Progress
33. Rosenberg, J. and Campbell, B., “Session Initiation Protocol Extension for Instant Messaging”, Internet Draft<draft-ietf-sip-message-04.txt>, IETF, 2002, Working in Progress
34. Rosenberg, J. et al., “Integration of Resource Management and SIP”, Internet Draft<draft-ietf-sip-manyfolks-resource-07.txt,.ps>, IETF, 2002, Working in Progress
35. Sparks, R., “The SIP Refer Method”, Internet Draft<draft-ietf-sip-refer-05.txt>, IETF, 2002, Working in Progress
36. Vemuri, A. and Peterson, J., “SIP for Telephones (SIP-T): Context and Architectures”, Internet Draft<draft-ietf-sipping-sipt-02.txt>, IETF, 2002, Working in Progress
White Papers and Technical Reports
37. Doumas, T., “Next Generation Telephony: A Look at Session Initiation Protocol”, White Paper of Agilent, 2001
38. Gerhard, G., “Advanced VoIP Applications”, White Paper of NUERA, 2001
39. Inmaculada Espigares del pozo, “An Implementation of the Internet Call Waiting Service using SIP”, Helsinki University of Technology, Dec.1999
40. Lamy, M., “A Description of MGCP and SIP”, White Paper of ADTRAN, 2002
41. RADVISION, “Technical Overview- RADVision SIP ToolKit”, Technical Report of RADIVISION, 2001
42. Rosenberg, J., “An Application Architecture for Communications Services”, White Paper of Dynamicsoft, 2001
43. Schulzrinne, H. and Eyers, T., “Predicting Internet Telephony Call Setup Delay”, Columbia University, 2002
44. Ubiquity, “White Paper- Application Powered Networks with SIP”, White Paper of Ubiquity, 2001
45. Ubiquity, “White Paper- SIP Service Architecture”, White Paper of Ubiquity, May. 2001
46. Venters, T., “Demystifying Protocols: A Comparison of Protocol Suitable for IP Telephony”, Sonus Networks, 2001
47. VOCAL, “VOCAL System Architecture”, http://www.vovida.org/, 2002
48. VOCAL, “VOCAL Administration Guide”, http://www.vovida.org/, 2002
49. VOCAL, “VOCAL Installation Guide”, http://www.vovida.org/, 2002
50. Wilkinson, N., “SIP Based Call Centres- A vendor independent architecture for multimedia contact centers”, Quortex Consultants, 2001
Books and Websites
51. Camarillo, G., “SIP Demystified”, McGraw Hill, 2001
52. http://java.sun.com/products/jain/
53. http://www.ietf.org/html.charters/sip-charter.html
54. http://www.parlay.org
55. http://www.sipforum.com
56. http://www.sipcenter.com
57. http://www.vovida.org
58. http://www.webpedia.com
59. Sinnreich, H. and Johnston, A.B., “Internet Communications Using SIP”, Wiley Computer Publishing, 2001