Please use this identifier to cite or link to this item: http://hdl.handle.net/11455/8670
標題: 在無線區域網路下VoIP語音品質分析與編解碼器的選擇
Experimental Analysis of the Effect of Codec Choice on VoIP Voice Quality over WLANs
作者: 溫嘉政
Wen, Jia-Jang
關鍵字: VoIP;SIP;CODEC
出版社: 電機工程學系所
引用: [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/
摘要: 
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。

VoIP 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.
URI: http://hdl.handle.net/11455/8670
其他識別: U0005-2707200914450200
Appears in Collections:電機工程學系所

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