Please use this identifier to cite or link to this item: http://hdl.handle.net/11455/8732
DC FieldValueLanguage
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謝景安zh_TW
dc.contributor.authorHsieh, Ching Anen_US
dc.contributor.other中興大學zh_TW
dc.date2010zh_TW
dc.date.accessioned2014-06-06T06:42:03Z-
dc.date.available2014-06-06T06:42:03Z-
dc.identifierU0005-3007200908440800zh_TW
dc.identifier.citation[1] Li Zheng、 Liren Zhang、Dong Xu, ”Characteristics of Network Delay and Delay Jitter and its Effect on Voice over IP(VoIP)” [2] RFC 3261,” SIP: Session Initiation Protocol” [3] RFC 3550,” RTP: A Transport Protocol for Real-Time Applications” [4] Thomas Hug,”Trace-Based Network Emulation” [5] Abderrahmane Lakas、Mohammed Boulmalf, ”Experimental Analysis of VoIP over Wireless Local Area Networks” [6] Michael F. Finneran,”VoIP quality issues: Jitter, delay, and echo” [7] VoIP的語音品質測量方法 http://www.eettaiwan.com/ART_8800378916_675327_TA_5c4ce775.HTM [8] ITU-T Recommendation P.861,”Objective quality measurement of telephone-base speech codecs” [9] ITU T Recommendation G.107, ”The E-model, a computational model for use in transmission planning” [10] ITU-T Recommendation G.113,” Appendix I: Provisional planning values for the equipment impairment factor Ie ” [11] NIST,http://snad.ncsl.nist.gov/nistnet/ [12] Mark Carson, Darrin Santay,”NIST Net – A Linux-Based Network Emulation Tool” [13] Riri Fitri Sari, Pedro Libratu Putu Wirya,”Performance Analysis of Session Initiation Protocol on Emulation Network Using NIST NET” [14] ZHANG Ze-ming, HU Lei, REN Jiu-chun, GAO Chuan-shan,” Research on Speech Quality of VoIP Based on E-Model” [15] Chin-Lin Liu, “Performance Evaluation of VoIP Flows With Different Service Models” [16] 柯守全, “Performance Measurement and Path Selection for Internet Telephony” [17] jdugan, mitchkutzko ,”Iperf” http://sourceforge.net/projects/iperf/ [18] Ethereal,http://www.ethereal.com/zh_TW
dc.identifier.urihttp://hdl.handle.net/11455/8732-
dc.description.abstractVoice Over IP(VoIP)在網路技術中,是一個很熱門的主題,它利用網際網路的協定,提供了即時的語音封包傳遞。由於網路頻寬是由大家所共用分享的,語音封包將與一般網路流量競爭使用的情況下,在網路不佳時,語音品質也跟著不佳。語音品質會受到延遲、抖動與不可靠的封包傳遞所影響,尤其這些參數在語音作QoS的量測時,更形重要。因此,VoIP面臨最主要的問題就是很難保證語音的QoS。 在本文中,討論了VoIP相關的品質因素,包括語音編解碼器的選擇、封裝、封包遺失、延遲、延遲變異(jitter)以及網路服務的模式,透過這些參數,我們提出一個VoIP通話品質的評估方法,檢測在不同的背景網路流量與Jitter之間的關係,我們藉由NIST(美國國家標準和技術研究院) Net與Chariot工具來模擬VoIP流量,並使用六種常用的語音編解碼器,以及瞬時流量與固定流量的兩種背景網路流量。根據模擬的結果,我們將提出不同的語音編解碼器在上述之背景網路流量下,延遲和延遲變異的效能分析。zh_TW
dc.description.abstractVoice Over IP is a hot topic for network technology. It refers to real-time delivery of packet voice across networks using the Internet protocol. Due to the bandwidth is shared by Internet users, the voice packages will compete with the traffic load, which make the voice quality deteriorate rapidly with heavily traffic load. Voice quality is affected by unreliable packet delivery. Therefore, one of the main problems in the implementation of packet voice over IP is to maintain the desired QoS . In this thesis, we discuss some factors of voice quality, including the choice of speech codec, packetization, packet loss, delay, delay variation (jitter) and network service model, Based on these factors, we propose a method to evaluate the communication quality for VoIP networks and to examine the relationship between the jitter and the characteristics of the background traffic. We simulate the VoIP traffic by NIST(National Institute of Standards and Technology) Net and Chariot. Given six speech codecs and two transmission models, regarding the results of our simulations, the performance analysis for the delay and delay jitter of different speech codecs based on two background traffic is presented.en_US
dc.description.tableofcontents誌謝 i 摘要 ii Abstract iii 目錄 iv 圖目錄 vi 表目錄 viii 第一章 緒論 1 1.1研究背景 1 1.2 研究動機 2 1.3論文章節架構 3 第二章 VoIP相關協定介紹 4 2.1 H.323通訊協定 4 2.2 SIP通訊協定 6 2.3 網路傳輸協定 9 2.3.1 RTP傳輸協定 9 2.3.2 RTCP傳輸協定 10 2.4 VoIP語音壓縮技術 11 2.4.1 語音壓縮編碼的原理 11 2.4.2 語音壓縮編碼技術 12 第三章 VoIP語音品質探討 14 3.1 影響VoIP通話品質因數 14 3.1.1 編碼機制 (CODEC) 14 3.1.2 頻寬 (Bandwidth) 15 3.1.3 封包延遲 (Packet Delay) 15 3.1.4 封包遺失 (Packet Loss) 16 3.1.5 抖動 (Jitter) 16 3.2 VoIP的語音品質標準 17 3.2.1 MOS[7] 17 3.2.2 PSQM / PAMS / PESQ 18 3.2.3 E-Model[9] 18 3.3 VoIP的語音測量方法 21 3.3.1 Iperf測試網路效能 21 3.3.2 Ethereal語音效能評估分析 23 第四章 網路語音測試環境模擬 25 4.1 系統架構 25 4.2 測試工具 26 4.2.1 NIST Net network emulator介紹 26 4.2.2 Chariot工具介紹 27 4.3 語音效能監測 29 4.3.1 多用戶流量競爭下之語音傳輸效能 29 4.3.2 單一用戶在不同傳輸週期下之語音傳輸效能 32 第五章 系統實作 35 5.1 各Codecs在多用戶流量競爭下之Delay Jitter表現 35 5.1.1 Traffic Load對G.711u(64Kbps)的影響 35 5.1.2 Traffic Load對G.711a(64Kbps)的影響 37 5.1.3 Traffic Load對G.723_ACELP(5.3Kbps)的影響 38 5.1.4 Traffic Load對G.723_MPMLQ(6.3Kbps)的影響 40 5.1.5 Traffic Load對G.729(8Kbps)的影響 42 5.1.6 Traffic Load對G.726(32Kbps)的影響 43 5.1.7 Codec之間在不同Traffic Load傳輸之比較 44 5.2 各Codecs在不同傳輸週期下之Delay Jitter表現 46 5.2.1 Burst對G.711u(64Kbps)的影響 46 5.2.2 Burst對G.711a(64Kbps)的影響 47 5.2.3 Burst對G.723_ACELP(5.3Kbps)的影響 48 5.2.4 Burst對G.723_MPMLQ(6.3Kbps)的影響 49 5.2.5 Burst對G.729(8Kbps)的影響 50 5.2.6 Burst對G.726(32Kbps)的影響 51 5.2.7 Codec之間在不同burst傳輸之比較 52 第六章 結論與未來發展 54 參考文獻 55zh_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=U0005-3007200908440800en_US
dc.subjectVoIPen_US
dc.subjectRTPen_US
dc.subjectJitteren_US
dc.subjectNIST Neten_US
dc.subjectCharioten_US
dc.titleVoIP Codec於不同傳輸模式下之效能量測zh_TW
dc.titlePerformance Measurement of VoIP Codecs in Different Transmission Modelsen_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
Show simple item record
 

Google ScholarTM

Check


Items in DSpace are protected by copyright, with all rights reserved, unless otherwise indicated.