以IMS為基礎之及時語音影像通話引擎的實作:使用開放原始碼程式庫

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姓名

姜怡楷(Yi-kai Chiang) 查詢紙本館藏

畢業系所

資訊工程學系

論文名稱

以IMS為基礎之及時語音影像通話引擎的實作:使用開放原始碼程式庫
(IMS-based Live Voice and Video Communication Engine Implementation: open source library extension development for VoIP and Video Call)

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至系統瀏覽論文 ( 永不開放)

摘要(中)

隨著科技的發展，網路技術越趨成熟，使得網路服務的型態已逐漸改變，在使用電路交換的時代，人們只能透過簡訊及電話來傳達訊息，現在已演進為封包交換的網路，不但可以進行語音通話，更可以進行多媒體資料的傳輸,因此透過影像傳輸技術，有聲無影的時代也即將告別。
IP多媒體子系統有電路交換及封包交換系統的優勢，所以，我們將實做出IP多媒體子系統客戶端的程式, 包括基本的語音影像通話功能，及時訊息及使用者狀態顯示功能，而在本篇的論文中，將著重於及時語音及影像通話引擎的實做,並且在微軟的作業平台上，盡可能避免掉語音通話中致命的議題-回音，使得通話品質可以維持一定的水準，且能維持通話的穩定性。

摘要(英)

With the development of the network technology, it drives the internet services change intensely. In the public switched telephone network (PSTN) age, people only could use the SMS and dial a telephone to communication. Nowadays, it can offer multimedia data transmission via packet switch network. People not only could make a voice call but also make a video call. Therefore, the age of multimedia will coming.
Since the IP Multimedia Subsystem (IMS) has the advantage of the cellular system and internet system. We would like to realize the IMS client. Including basic voice and video call, instant message, and presence function. In this paper, we will focus on the voice and video communication engine implementation that using the open source library. And keep the echo-free in the Microsoft Operation System in the VoIP. As a consequence, we hope to have a good quality of voice. And keep the stable in the every communication.

關鍵字(中)

★ 網路電話
★ IP多媒體子系統
★ 公共交換電話網
★ 無回音

關鍵字(英)

★ echo-free
★ IP Multimedia Subsystem
★ PSTN
★ VoIP

論文目次

中文摘要 i
Abstract ii
致謝 iii
Table of Contents iv
List of Tables vii
List of Figures viii
1 Introduction 1
1.1 Motivation 1
1.2 Thesis Organization 3
2 Background and Related Works 5
2.1 IP Multimedia Subsystem (IMS) core overview 5
2.2 Session Initiation Protocol (SIP) 7
2.2.1 The method of the SIP message 8
2.2.2 The header of SIP message 9
2.2.3 The body of SIP Messages 10
2.3 Session Describe Protocol (SDP) 10
2.4 Real-time Transport Protocol (RTP) 11
2.5 Voice codec 12
2.6 Video codec 13
2.7 Echo problem 14
3 System Architecture 17
3.1 Goal 17
3.2 System Overview 18
3.3 System Architecture 20
3.4 System Components 21
3.4.1 Call Engine Module 21
3.4.2 Graphic User Interface 23
3.4.3 Open source library- OPAL 24
3.4.4 Open source library- PTLib 24
4 Implementation 26
4.1 Voice over Internet Protocol ( VoIP ) 26
4.2 Video over Internet Protocol 32
4.3 Silence Suppress 35
4.4 Acoustic Echo Canceller on Open Source Library 35
5 Conclusion and Future Works 41
List of References 43

參考文獻

[1] Voice over Internet Protocol.
http://en.wikipedia.org/wiki/VoIP.
[2] Overview of 3gpp release 5.
http://www.3gpp.org/fRP-030375.pdf.
[3] J. Rosenberg and H. Schulzrinne, “Reliability of provisional responses in Session Initiation Protocol (SIP). RFC 3262.” Internet Engineering Task Force, June 2002.
[4] Wireshark.
http://www.wireshark.org/.
[5] M. Handley and V. Jacobson., “SDP: Session Description Protocol. RFC 4566.”Internet Engineering Task Force, July 2006.
[6] R. F. H. Schulzrinne, S. Casner and V. Jacobson., “RTP: a transport protocol for real-time applications. RFC 3550.” Internet Engineering Task Force, July 2003.
[7] “ITU. Pulse Code Modulation (PCM) of voice frequencies. Technical Report ITU-T Rec. G.711.” International Telecommunications
Union (ITU), 1988.
[8] “ITU. Coding of speech at 8 Kbit/s using Conjugate-Structure Algebraic-Code-Excited Linear-Prediction (CS-ACELP). Technical Report ITU-T Rec. G.729.” International Telecommunications Union (ITU), March 1996.
[9] “ITU-T. Video codec for audiovisual services at p 64 kbit/s. Recommendation H.261.” International Telecommunication Union, March 1993.
[10]” ITU-T. Video coding for low bit rate communication. Recommendation
H.263.” International Telecommunication Union, May 2005.
[11] “OpalVoip - Open Source Voice, Video and Fax."
http://www.opalvoip.org/.
[12] J. Rosenberg and H. Schulzrinne, “An offer/answer model with Session Description Protocol (SDP). RFC 3264.” Internet Engineering Task Force, June 2002.
[13] E. S. J. R. M. Handley, H.Schulzrinne, “SIP: Session Initiation Protocol (SIP). RFC 2543.” March 1999.

指導教授

吳曉光(Hsiao-kuang Wu)

審核日期

2009-7-9

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