Please use this identifier to cite or link to this item: http://hdl.handle.net/11455/8670
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dc.contributor郭耀文zh_TW
dc.contributorYaw-Wen Kuoen_US
dc.contributor廖俊睿zh_TW
dc.contributorJan-Ray Liaoen_US
dc.contributor.advisor溫志煜zh_TW
dc.contributor.advisorChih-Yu Wenen_US
dc.contributor.author溫嘉政zh_TW
dc.contributor.authorWen, Jia-Jangen_US
dc.contributor.other中興大學zh_TW
dc.date2010zh_TW
dc.date.accessioned2014-06-06T06:41:58Z-
dc.date.available2014-06-06T06:41:58Z-
dc.identifierU0005-2707200914450200zh_TW
dc.identifier.citation[1] ETSI TS 101 329-5 QoS Measurement for VoIP [2] ITU-T G.107 The E Model: A computational model for use in planning [3] ITU-T SG12 Contribution D105, VQmon Description, January 2003 [4] ITU-T Recommendation P.862 Perceptual Estimation of Speech Quality (PESQ) [5] The NIST Net network emulator,http://snad.ncsl.nist.gov/itg/nistnet/ [6] SIP,http://voip.about.com/od/sipandh323/p/whatisSIP.htm [7] 影響企業級IP電話性能因素分析及解決方案, http://www.eettaiwan.com/ART_8800455932_622964_TA_95262775.HTM [8] 良好整合軟硬體滿足VoIP設備開發需求, http://www.eettaiwan.com/ART_8800444433_675327_TA_0d475f45.HTM [9] Throughput,http://en.wikipedia.org/wiki/Throughput [10] Voice over Internet Protocol,http://en.wikipedia.org/wiki/Voip [11] Session Initiation Protocol, http://en.wikipedia.org/wiki/Session_Initiation_Protocol#SIP_Messages [12] RFC 1889,http://freesoft.org/CIE/RFC/1889/index.htm [13] SIP: Session Initiation Protocol,http://www.ietf.org/rfc/rfc3261.txt [14] http://www.ietf.org/html.charters/sip-charter.html [15] Codecs,http://www.voip-info.org/wiki/view/Codecs [16] IEEE 802.11g-2003,http://en.wikipedia.org/wiki/802.11g [17] IxChariot - Network Assessment Tool,http://www.ixiacom.com/products/ixchariot/zh_TW
dc.identifier.urihttp://hdl.handle.net/11455/8670-
dc.description.abstract以SIP為基礎的VoIP平台已漸漸成為市場主流,在提供SIP網路電話服務給企業用戶使用者時,除了要兼具效能、節省成本外,其便利性和創新是企業成長的動力及競爭力。因此在本論文中,我們採用Mobile SIP的概念來設計測試環境,讓使用者小範圍的移動並搭配自訂的環境以延伸出不同的實驗劇本來滿足用戶端的實際需求。用透過支援SIP協定的工具軟體搭配無線網路以多量的連線建立來模擬企業端的網路電話負載量,找出寬頻CODEC及窄頻CODEC在外加干擾因素如抖動、延遲、封包遺失等影響下網路品質參數分析曲線並完成語音效能相關參數的分析及語音編解碼器的選擇。zh_TW
dc.description.abstractVoIP platform based on SIP has already become the market mainstream gradually, while offering SIP network telephone service to the user of enterprise, besides having efficiency concurrently, saving the cost, its convenience and innovation are motive force and competitiveness that enterprises grow up. So in this thesis, we adopt the concept of Mobile SIP to design the environment of testing, let user movement among a small circle and collocate environment that book by oneself with is it happen different experiment drama is it satisfy actual demand, user of end to come to extend. With through support tool software, SIP of protocol collocate wireless network is it imitate load amount of phone of network, enterprise of end to come to set up with line of volume, find out wide-band CODEC and narrow-band CODEC in interfering if factor Jitter, Delay, package loss is it down network quality parameter analyse curve and finish voice efficiency relevant analysis and choice of arranging CODEC of voice of parameter to influence to lose.en_US
dc.description.tableofcontents誌謝…………………………………………………………………………………v 摘要…………………………………………………………………………………vi Abstract…………………………………………………………………………vii 目錄……………………………………………………………………………viii 圖目錄……………………………………………………………………………x 表目錄……………………………………………………………………………xii 第一章 序論……………………………………………………………………1 1.1簡介……………………………………………………………………1 1.2研究動機………………………………………………………………2 1.3論文章節架構…………………………………………………………2 第二章 相關知識………………………………………………………………3 2.1 VoIP介紹 ……………………………………………………………3 2.2 SIP通訊協定…………………………………………………………4 2.3 VoIP語音壓縮技術 …………………………………………………9 2.3.1 G.711u語音壓縮技術簡介 ……………………………………9 2.3.2 G.723.1語音壓縮技術簡介 ……………………………………9 2.3.3 G.726語音壓縮技術簡介 ……………………………………10 2.3.4 G.729語音壓縮技術簡介 ……………………………………10 第三章 VoIP語音品質量測影響參數 ………………………………………12 3.1 Throughput對VoIP語音品質的影響………………………………12 3.2 RFC1889 Jitter對VoIP語音品質的影響 …………………………12 3.3 MOS 、P.861(PSQM) 、E模型對VoIP語音品質的影響………12 3.4單向延遲對VoIP語音品質的影響…………………………………18 第四章 系統設計 ……………………………………………………………19 4.1 NIST Net 網路模擬器簡介…………………………………………19 4.2 IxChariot工具介紹 ………………………………………………20 4.3 無線網路IEEE 802.11g ……………………………………………22 4.4 系統架構圖、實驗設備表 …………………………………………23 4.5 模擬環境 …………………………………………………………25 第五章 系統實作 ……………………………………………………………26 5.1 實驗一:網路品質變化對CODEC的Throughput表現……………26 5.2 實驗二:網路品質變化對CODEC的RFC1889 Jitter表現………30 5.3 實驗三:網路品質變化對CODEC的MOS表現……………………33 5.4 實驗四:網路品質變化對CODEC的OneWayDelay表現…………37 5.5 實驗五:限制網路頻寬512kbps對CODEC的MOS比較…………40 第六章 結論與未來發展 …………………………………………………………43 參考文獻………………………………………………………………………44zh_TW
dc.language.isoen_USzh_TW
dc.publisher電機工程學系所zh_TW
dc.relation.urihttp://www.airitilibrary.com/Publication/alDetailedMesh1?DocID=U0005-2707200914450200en_US
dc.subjectVoIPen_US
dc.subjectSIPen_US
dc.subjectCODECen_US
dc.title在無線區域網路下VoIP語音品質分析與編解碼器的選擇zh_TW
dc.titleExperimental Analysis of the Effect of Codec Choice on VoIP Voice Quality over WLANsen_US
dc.typeThesis and Dissertationzh_TW
item.openairecristypehttp://purl.org/coar/resource_type/c_18cf-
item.languageiso639-1en_US-
item.openairetypeThesis and Dissertation-
item.grantfulltextnone-
item.fulltextno fulltext-
item.cerifentitytypePublications-
Appears in Collections:電機工程學系所
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