隨著網際網路服務的進步以及智慧型裝置的普及，愈來愈多人開始使用網路通話軟體取代一般電話，線上語音通訊服務顯然成為了現今熱門的討論話題。然而透過網路的語音通訊服務為了維持良好的通話品質，需要使用大量的封包流量。雖然電信業者提供了無限流量的資費專案，但並非所有民眾皆會選擇此種專案。如何在不耗費大量流量的前提下，有效提升網路通話的品質及穩定性，成為了一項重要的議題。
為了解決語音通話所帶來的流量問題，我們設計了一套雲端橋接交換機(Cloud Bridging Switch, CBS)。此交換機以雲端架構為基礎，能依需求動態分配伺服器的負載平衡。更重要的是，CBS能將IP電話與公共交換電話網路結合，僅在初始通話的階段需要使用極少的網路流量。在通話階段時，所有語音封包皆透過公共交換機傳輸，故使用者不須支付昂貴的行動網路費用。除此之外，由於語音不經行動網路而係由伺服器直接傳遞至PSTN，相較傳統IP電話通話品質及穩定性將大幅上升。
在此篇論文中，我們建立了一套雲端伺服器以提供符合SIP協定的語音通訊服務。此套伺服器展現了雲端架構高擴展性的優勢。我們並且設計了一套可運行於Android移動平台上的Application，透過此App可存取雲端服務，以極低的流量進行一對一或是多對多的網路通訊。

With the advancement of the Internet services and widespread of smart devices, there are many people starting to replace public switch telephone with VoIP software. Nowadays, voice over the Internet apparently becomes a highly discussed topic. However, in order to maintain a good conversation quality, it must cost a lot of bandwidth to use VoIP. Although Internet service providers provide billing plan with unlimited network access, not all people choose this kind of payments. How to improve the quality and stability of Internet conversation without costing a large amount of bandwidth has become an important issue.
To deal with the bandwidth problem, we introduce Cloud Bridging Switch, CBS. Cloud Bridging Switch which bases on cloud architecture is loading balanced dynamically according to demands. Moreover, CBS can bridge VoIP and PSTN, and it only costs very little bandwidth when initializing the call. All the audio packets in the call are delivered through PSTN, therefore users need not to pay the extra fee to the service provider. In addition, since the packets do not rely on the mobile network, the quality and steadiness of call are much better than of traditional VoIP software.
In this thesis, we build a series of cloud servers to provide SIP based voice services. The servers feature the high flexibility of cloud architecture. We also developed three Android application programs which can access to the cloud service and make one-to-one or many-to-many calls at low cost.

Abstract I
中文摘要 II
Table of Contents III
List of Figures V
List of Tables VI
Chapter 1 Introduction 1
Chapter 2 Related Works 4
2.1 Conference Call 4
2.2 Asterisk 5
2.3 6Talk 6
2.4 Google Cloud Message 7
2.5 Taiwan Academic Network 7
Chapter 3 Preliminary 8
3.1 Session Initiation Protocol 8
3.1.1 User Agent 9
3.1.2 SIP Message 9
3.1.3 Transactions 10
3.2 Web Service 11
3.2.1 Overview 11
3.2.2 WSDL 12
3.2.3 SOAP 13
3.2.4 UDDI 13
Chapter 4 System Architecture 14
4.1 Overview 14
4.2 User Scenario 17
4.3 System Implementation 20
4.3.1 Cloud Bridging Switch 20
4.3.2 Web Service Server 24
4.3.3 Database Server 29
4.3.4 Client 30
Chapter 5 Graphical User Interface 38
5.1 KITs 38
5.1.1 Registration and Validation 38
5.1.2 Login 39
5.1.3 Built-in Numbers 40
5.1.4 Calling 41
5.1.5 History List 42
5.2 KITs abroad 42
5.2.1 Calling 42
5.3 ConCall 43
5.3.1 Booked and Passed Conference 44
5.3.2 Booking and Starting a Conference 45
5.3.3 Conference Administration 46
Chapter 6 Conclusion and Future Works 47
References 49

[1] Microsoft, "Skype," [Online]. Available: http://www.skype.com/.
[2] Naver, "Line," [Online]. Available: http://line.me/.
[3] "民國102年個人/家戶數位機會調查報告," Research, Development and Evaluation Commission, [Online]. Available: http://archive.rdec.gov.tw/public/Attachment/3122613393871.pdf.
[4] ITU-T, "G.729," [Online]. Available: http://www.itu.int/rec/T-REC-G.729/e.
[5] Wikipedia, "Public switched telephone network," [Online]. Available: http://en.wikipedia.org/wiki/Public_switched_telephone_network.
[6] Digium, "Asterisk," [Online]. Available: http://www.asterisk.org/.
[7] "ChungHwa Telecom Conference Call," ChungHwa Telecom, [Online]. Available: http://www.cht.com.tw/enterprise/concall.html.
[8] "Taiwan Mobile Conference Call," Taiwan Mobile, [Online]. Available: http://www.twmsolution.com/product/ac/index_1.htm.
[9] "FETnet Conference Call," FETnet, [Online]. Available: http://www.fetnet.net/cs/Satellite/EBU/ebAudioCon.
[10] "Chunghwa Telecom Conference Call rate," Chunghwa Telecom, [Online]. Available: http://www.cht.com.tw/enterprise/concall-rate.html.
[11] "Taiwan Mobile Conference Call Rate," Taiwan Mobile, [Online]. Available: http://www.twmsolution.com/product/ac/index_3.htm.
[12] "FETNet Conference Call Rate," FETNet, [Online]. Available: http://www.fetnet.net/cs/Satellite/EBU/ebAudioCon_Rate.
[13] A. Imran, M. A. Qadeer and M. J. R. Khan, "Asterisk VoIP Private Branch Exchange," in International Conference on Multimedia, Signal Processing and Communication Technologies, 2009. IMPACT '09, 217 - 220, March 2009.
[14] F. Iseki, Y. Sato and M. W. Kim, "VoIP System based on Asterisk for Enterprise Network," in 2011 13th International Conference on Advanced Communication Technology (ICACT), 1284 - 1288, Feb. 2011.
[15] "6Talk," Uninum, [Online]. Available: http://www.callmesir.net/6talk/.
[16] Google, "Google Cloud Message," [Online]. Available: http://developer.android.com/google/gcm/index.html.
[17] "TANet VoIP," Ministry of Education, [Online]. Available: http://voip.tanet.edu.tw/.
[18] Wikipedia, "Session Initiation Protocol," [Online]. Available: http://en.wikipedia.org/wiki/Session_Initiation_Protocol.
[19] IETF, "RFC 3261," [Online]. Available: http://www.ietf.org/rfc/rfc3261.txt.
[20] W3C, "Web Service," 11 February 2004. [Online]. Available: http://www.w3.org/TR/2004/NOTE-ws-gloss-20040211/#webservice.
[21] W. Chou, L. Li and F. Liu, "Web Services for Service-Oriented Communication," in International Conference on Collaborative Computing: Networking, Applications and Worksharing, 2006. CollaborateCom 2006., 1 - 8, Nov. 2006.
[22] W3C, "WSDL 2.0," 26 June 2007. [Online]. Available: http://www.w3.org/TR/wsdl20/.
[23] W3C, "SOAP 1.2," 27 April 2007. [Online]. Available: http://www.w3.org/TR/2007/REC-soap12-part1-20070427/.
[24] I. MongoDB, "MongoDB," [Online]. Available: http://www.mongodb.org/.
[25] H. Mcheick and Y. Qi, "Dependency of Components in MVC Distributed Architecture," in 2011 24th Canadian Conference on Electrical and Computer Engineering (CCECE), 691 - 694, May 2011.
[26] C.-C. Cheung, M.-C. Yuen and A. C. H. Yip, "Dynamic DNS for Load Balancing," in 23rd International Conference on Distributed Computing Systems Workshops, 2003., 962 - 965, May 2003.
[27] M. L. Chin, C. E. Tan and M. I. Bandan, "Efficient Load Balancing for Bursty Demand in Web based Application Services via Domain Name Services," in 2010 8th Asia-Pacific Symposium on Information and Telecommunication Technologies (APSITT), 1 - 4, June 2010.
[28] B. Upadhyaya, Z. Ying, H. Xiao, J. Ng and A. Lau, "Migration of SOAP-based Services to RESTful Services," in 2011 13th IEEE International Symposium on Web Systems Evolution (WSE), 105 - 114, Sept. 2011.
[29] K. Y. Lai, T. K. A. Phan and Z. Tari, "Efficient SOAP Binding for Mobile Web Services," in The IEEE Conference on Local Computer Networks, 2005. 30th Anniversary., 218 - 225, Nov. 2005.
[30] L. Kapicak, P. Nevlud, J. Zdralek, P. Dubec and J. Plucar, "Remote Control of Asterisk via Web Services," in 2011 34th International Conference on Telecommunications and Signal Processing (TSP), 27 - 30, Aug. 2011.
[31] P. Montoro and E. Casilari, "A Comparative Study of VoIP Standards with Asterisk," in Fourth International Conference on Digital Telecommunications, 2009. ICDT '09., 1 - 6, July 2009.
[32] S. Yalezo and T. Mamello, "Architecting and constructing an SOA bridge for an MVC platform," in 2013 Fourth World Congress on Software Engineering (WCSE), 45 - 49, Dec. 2013.