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博碩士論文 etd-0724106-220808 詳細資訊
Title page for etd-0724106-220808
論文名稱
Title
網路電話語音錄音系統之設計與實作
Design and Implementation of Voice Recorder over SIP Based VoIP System
系所名稱
Department
畢業學年期
Year, semester
語文別
Language
學位類別
Degree
頁數
Number of pages
46
研究生
Author
指導教授
Advisor
召集委員
Convenor
口試委員
Advisory Committee
口試日期
Date of Exam
2006-07-14
繳交日期
Date of Submission
2006-07-24
關鍵字
Keywords
網路電話、錄音、留言、會談啟始通訊協定
VoIP, Redirect, SER, SIP
統計
Statistics
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The thesis/dissertation has been browsed 5686 times, has been downloaded 0 times.
中文摘要
隨著網路基礎建設的落實,寬頻網路逐漸成為主流,而足夠的頻寬也讓許多運用得以成真,網路電話便是其中一例。透過網路電話溝通人們得以只付區域性的網路服務連線費,就可以以相對便宜的費用撥打長時間、長距離的電話。在網路電話趨勢慢慢浮現, 基礎的通話功能能夠滿足之後,網路電話的附加建設就更形重要,除了基本的能通話以外,使用者也會要求提供更多的附加服務功能。
本文提出並實作一個網路電話錄音的機制,讓使用者能夠以數位的方式將雙方通話中的對話錄製下來,之後利用此錄音機制實作一語音留言伺服器,提供使用者當受話方未接聽電話時可以轉入語音信箱錄製留言,受話方可於稍後撥打電話至留言伺服器收聽自己的留言訊息。
Abstract
As the network fundamental infrastructures become mature, broadband network turns into the main stream. Sufficient bandwidth makes many applications, for example, voice over IP (VoIP), become possible. Through IP phone, people only need to pay local Internet service fee, which is relatively more inexpensive, to be able to make long-distance call with remote people. After the basic calling facility is ready, additional VoIP services become more and more important. User will demand for more additional service functions.
In this thesis, I propose and implement a voice recording facility based on SIP-based VoIP system. Users can record both caller and callee's voice together in digital way. Furthermore, we use this facility to provide a voice message recording service. When callee does not pickup his/her phone, caller's phone will be redirected to voice message recording server. Caller can record his/her voice message into callee's directory on the voice recording server, and callee can listen to his/her own voice message later.
目次 Table of Contents
1. 導論 6
1.1. 簡介 6
1.2. 論文架構 7
2. 相關研究 8
2.1. 會談啟始通訊協定(Session Initial Protocol, SIP ) 8
2.1.1. SIP的發展歷史 8
2.1.2. SIP的基本元件 9
2.1.3. SIP 訊息(SIP Message) 11
2.1.4. SIP的交易與對話 14
2.1.5. SIP 的基本操作 15
2.1.6. SIP 的優點 18
2.2. 會談描述協議(SDP, SIP Description Protocol) 19
2.3. 及時傳輸協定(RTP, Real-time Transport Protocol) 20
2.3.1. RTP Header 21
2.4. SER (SIP Express Router) 22
2.4.1. SER的特色 22
2.4.2. 條件式判斷 23
2.4.3. 重寫URI 24
3. 實作平台 26
3.1. 錄音伺服器 26
3.2. Infineon Easy 5120平台 26
4. 架構設計與實作 29
4.1. 電話錄音機制 29
4.2. 錄音伺服器接聽動作 30
4.3. 錄音伺服器錄音機制 30
4.4. 錄音伺服器放音機制 32
4.5. 錄音伺服器留言機制 35
4.6. SER 伺服器修改及設定 36
5. 結論與未來發展方向 38
5.1. 結論 38
5.2. 未來發展 38
Reference: 40
Appendix A. eXosip 42
Appendix B. oSIP 43
參考文獻 References
[1] IETF J. Rosenberg, et al, “RFC3261, SIP: Session Initiation Protocol”, June 2002
http://www.faqs.org/rfcs/rfc3261.html
[2] IETF M Handley, V. Jacobson, et al, “RFC2327, SDP: Session Description Protocol ”,April 1998
http://www.faqs.org/rfcs/rfc3264.html
[3] IETF H. Schulzrinne, et al, “RFC3550, RTP: A Transport Protocol for Real-Time Applications”, July 2003
http://www.faqs.org/rfcs/rfc3551.html
[4] “RTP, Real-Time Transport Protocol”
http://www.networksorcery.com/enp/protocol/rtp.htm
[5] “SIP Express Router”
http://www.iptel.org/ser/
[6] Jan Janak, “SIP Introduction”
http://www.iptel.org/ser/doc/sip_intro/sip_introduction.html
[7] Alan B. Johnston, “SIP Understanding the Session Initiation Protocol”, Artech House, 2004
[8] Jiri Kuthan, Jan Janak, Yacine Rebahi, “iptel.org SIP Express Router Admin’s Guide”
http://www.iptel.org/ser/admin.html
[9] oSIP library documentation
http://www.gnu.org/software/osip/
[10] eXosip library documentation
http://savannah.nongnu.org/projects/exosip/
[11] “Voice over IP - Per Call Bandwidth Consumption”
http://www.cisco.com/warp/public/788/pkt-voice-general/bwidth_consume.html
[12] MySQL 3.23, 4.0, 4.1 Reference Manual
http://dev.mysql.com/doc/refman/4.1/en/index.html
[13] phpMyAdmin documentations
http://www.phpmyadmin.net/home_page/docs.php
[14] W. Richard Stevens, “Advanced Programming in the UNIX(R) Environment”, Addison-Wesley Professional; 2 edition, June, 2005
[15] “Telephony Application Programming Interface for Infineon Voice Codec Devices”, Infineon Technologies
[16] “EASY 5120 Hardware Description”, Infineon Technologies
[17] Apache HTTP Server Version 2.2 Documentation
http://httpd.apache.org/docs/2.2/
[18] SIP: Session Initiation Protocol
http://www.cs.columbia.edu/sip/
[19] Xlite User Manual, CounterPath Solutions ,Inc.
http://www.xten.com/index.php?menu=Products&smenu=UserGuides
[20] H. Schulzrinne and J. Rosenberg, "A Comparison of SIP and H.323 for Internet Telephony," The 8th International Workshop on Network and Operating Systems Support for Digital Audio and Video (NOSSDAV 98), Cambridge, England, July 1998.
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