臺灣博碩士論文加值系統

SIP 是IETF 制定主要用來初始、修改與結束多媒體會議的一個協定，這篇論
文我們提出了three-tier 的架構來實現測試SIP-base VoIP 系統的效能，利用多台的
PCs 來達到較大Loading 的產生，設計出Lead & Wait Balance algorithm 來達到
Loading Balance，而Tester 也整合了測試SIP protocol conformance、performance test for the SIP server 也可以利用multi-VoIP session generator 的方法來測試 RTP-proxy server 的效能

Session Initiation Protocol defined by IETF is the major signaling protocol for
initiating, modifying, and terminating multimedia session. Using the three-tier
technique, this paper designs a tester for SIP-based VoIP applications in functional and performance testing. Coordinated with multiple PCs, the functions of the tester include SIP protocol conformance test, performance test for the SIP server, and multiple VoIP session packet generation which can be applied to the loading test of the RTP-proxy server in VoIP application.

中文摘要 i
英文摘要 ii
誌謝 iii
目錄 iv
表目錄 vi
圖目錄 vii
一、 緒論 1
二、 相關背景 3
2.1 SIP概觀 3
2.2 SIP網路的組成 5
2.3 SIP協定的格式 7
2.3.1 SIP請求訊息 8
2.3.2 SIP回應訊息 13
2.4 SIP呼叫模組 15
2.4.1 點對點模式 16
2.4.2 代理模式 18
2.5 網路語音簡介 23
2.6 語音的壓縮編碼與解碼 25
2.7 語音壓縮格式簡介 27
2.4.1 語音網路傳輸 29

三、 系統設計與架構實現 33
3.1 系統功能 33
3.1.1 SIP server 效能測試 33
3.1.2 RTP proxy server傳遞時間測試 34
3.1.3 SIP 協定一致性測試 35
3.2 SIP-based VoIP Tester系統架構 35
3.3 效能與精確度的提升 41
3.4 Lead&Wait Balance負載平衡的設計與負載的分配 45
四、 效能分析與測試結果 48
4.1 對於SIP server 效能的比較 48
4.2 Response Time與 Session Loss Rate的量測 50
4.3 對於Lead&Balance的負載分配平衡度量測 53
五、 結論 56
參考文獻 57
附錄一 58
簡 歷 59

[1] Recommendation H.323, Packet-Based Multimedia Communications Systems, Int’l Telecommunications Union, 1999.

[2] Hohnston, J. Peterson, R. Sparks,M. Handley, and E.Schooler, “SIP:Session
Initiation Protocol”,www.ietf.org/rfc/rfc3261.txt

[3] “SIPp”,sipp.sourceforge.net

[4] M. Handley, ACIRI, H. Schulzrinne,Columbia U., E. Schooler, Cal Tech, J. Rosenberg and Bell Labs“SIP:Session Initiation,Protocol",
www.ietf.org/rfc/rfc2543.txt

[5] H. Nyquist, "Certain topics in telegraph transmission theory", Trans. AIEE,
vol. 47, pp. 617-644, Apr. 1928,Reprint as classic paper in: Proc. IEEE, Vol. 90,No. 2, Feb 2002

[6] H. Schulzrinne, GMD Fokus, S. Casner, R. Frederick, V. Jacobson,” RTP: A
Transport Protocol for Real-Time Applications”,www.faqs.org/rfcs/rfc1889.html

[7] ITU, 2003. [Online]. Available:
http://www1.cs.columbia.edu/_andreaf/new/documents/other/T-REC-G.114-200305.pdf

[8] M. Handley and V. Jacobson, “SDP: session description protocol”, Request
for Comments 2327,Internet Engineering Task Force, Apr. 1998.

[9] “SIP Express Router(SER)”,www.iptel.org

[10] “Brekeke SIP server”,www.brekeke.com

[11] 賈文康, 2006, “SIP 會談啓始協議操典 Session Initiation Protocol(SIP)
Methodology Handbook”, 松崗, March.