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題名 基於群組模式下之大型網路語音會談系統
A cluster-based transmission scheme for large scale VoIP conferencing
作者 邵育晟
貢獻者 連耀南
邵育晟
關鍵詞 語音通話
點對點
日期 2009
上傳時間 9-Apr-2010 14:49:27 (UTC+8)
摘要 利用VoIP進行網路會談(VoIP Conference)在現今的社會已有越來越流行的趨勢,尤其是遠距離語音會議的應用方面,不但能節省費用,還能同時允許多人上線會談,但隨著上線人數的增加,網路的頻寬需求與連線數量相對倍增,通話延遲時間也因頻道擁塞而難以控制,此時會議的品質(QoS)往往大打折扣,網路的負擔也會大幅增加。現行的主要解決方案為選擇網路會議中處理能力較強的使用者設備,將來自個別說話者的聲音經過混音疊加後再轉送給其他使用者,再利用不同架構的multicasting tree廣播語音封包至所有成員,藉此減少網路頻寬與連線數量的負擔,達到增加同時說話人數的目的。但這些作法在使用者所在位置過於分散、遙遠時,仍然會造成許多節點無法達到所需的通話品質。
     本研究針對在大型網路語音會談中,與會人數過多造成通話品質不佳之問題,提出了解決方法,我們假設會議中大部份時間僅有一個說話者,首先為每位使用者加上靜音消除機制,只有發話者的封包直接送給其他所有與會人員,如此不但降低網路負擔,更可以刪除混音的需求,大幅降低傳輸時間,此外並以分群組(Cluster)的方式將位於同一地區的使用者分為同一群組,群組內以Full Mesh機制相連,如此便能建立一複雜度較小的MLDST (Minimum Loss Diameter Spanning Tree) 樹狀結構的multicasting tree用以廣播語音封包至所有成員,可以進一步控制傳輸時間及封包遺失率。我們根據使用者之間彼此的連線速度與實際距離來分群組,從群組中選出連線能力較佳的節點作為群組轉送點(Cluster Head),透過此轉送點收到聲音後做群組內的發送,並藉由一個multicasting tree將封包在嚴格控制延遲時間及封包遺失率之下,將語音封包廣播至其他所有Cluster Head,以確保VoIP網路會議之品質。
     我們在PlanetLab這種實際網路的實驗平台上進行實驗,評估本方法的效能,實驗結果顯示我們的方法可以在與會人數達到20人時,仍能有效控制延遲時間在350ms左右,若利用Google雲端運算系統的協助,平均延遲時間將可減少50ms至70ms。由於PlanetLab上的電腦性能普遍不佳,我們預期本技術應用於實際網路時,可以增加與會人數。
參考文獻 [1] S. Banerjee, B. Bhattacharjee, and C. Kommareddy, "Scalable application layer multicast", in Proc. of ACM SIGCOMM, Pittsburgh, PA, August 2002.
[2] A. Bharambe, V. Padmanabhan, and S. Seshan, "Supporting Spectators in Online Multiplayer Games", in Proc. of HOTNETSIII, San Diego, November 2004.
[3] Uyless Black, "VOICE OVER IP", New Jersey: Prentice Hall, 2000.
[4] M. Castro, P. Druschel, A.-M. Kermarrec, A. Nandi, A. Rowstron, and A. Singh, "SplitStream: High-Bandwdith Multicast in Cooperative Environments", in Proc. of ACM SOSP 2003, New York, October 2003.
[5] M. Castro, P. Druschel, A.-M. Kermarrec, and A. Rowstron, "SCRIBE: A large-scale and decentralized application-level multicast infrastructure", in Proc. of ACM SIGCOMM, San Diego, CA, August 2001.
[6] Li-Chen Chi, "Echo Cancellation in Large-Scale VoIP conferencing", 2009.
[7] Wei-Chung Chiu, "Quality Assurance of Streaming Data Dissemination over P2P Network", 2009.
[8] Y. H. Chu, S. G. Rao, S. Seshan, and H. Zhang, "Enabling Conferencing Applications on the Internet using an Overlay Multicast Architecture", in Proc. of ACM SIGCOMM, San Diego, CA, August 2001.
[9] B. Goode, "Voice over Internet Protocol (VoIP)", in Proc. OF THE IEEE, VOL.90, NO.9, Sep. 2002.
[10] Xiaohui Gu, Z. Wen, Philip S. Yu, and Zon-Yin ShaeZhen, "peerTalk: A Peer-to-Peer Multi-Party Voice-Over-IP System", in Parallel and Distributed Systems, IEEE Transactions, April 2008.
[11] Xiaohui Gu, Z. Wen, Philip S. Yu, and Zon-Yin ShaeZhen, "Supporting Multi-Party Voice-Over-IP Services with Peer-to-Peer Stream Processing", in Proc. of ACM Multimedia, Singapore, November 2005.
[12] D. Kostic, A. Rodriguez, J. Albrecht, and A. Vahdat, "Bullet: High Bandwidth Data Dissemination Using an Overlay Mesh", in Proc. of ACM SOSP 2003, New York, October 2003.
[13] J. Lennox and H. Schulzrinne, "A Protocol for Reliable Decentralized Conferencing", in NOSSDAV`03, Monterey, California, USA, June. 2003.
[14] B. Li and H. Yin, "Peer-to-Peer Live Video Streaming on the Internet: Issues, Existing Approaches, and Challenges", in IEEE Communications Magazine, Toronto, Ont., Canada, June 2007.
[15] M. Radenkosvic and C. GreenHalgh, "Multi-party Distributed Audio Service with TCP Fairness", in Proc. of ACM Multimedia, 2002, Juan-les-Pins, France, December 2002.
[16] Google Talk, http://www.google.com/talk/intl/zh-TW/.
[17] G.723, http://www.itu.int/rec/T-REC-G.723/e.
[18] G.729, http://www.itu.int/rec/T-REC-G.729/e.
[19] Mean Opinion Score, http://en.wikipedia.org/wiki/Mean_opinion_score.
[20] Skype, http://www.skype.com.
[21] The E-Model, http://www.itu.int/rec/T-REC-G.107.
描述 碩士
國立政治大學
資訊科學學系
96753001
98
資料來源 http://thesis.lib.nccu.edu.tw/record/#G0096753001
資料類型 thesis
dc.contributor.advisor 連耀南zh_TW
dc.contributor.author (Authors) 邵育晟zh_TW
dc.creator (作者) 邵育晟zh_TW
dc.date (日期) 2009en_US
dc.date.accessioned 9-Apr-2010 14:49:27 (UTC+8)-
dc.date.available 9-Apr-2010 14:49:27 (UTC+8)-
dc.date.issued (上傳時間) 9-Apr-2010 14:49:27 (UTC+8)-
dc.identifier (Other Identifiers) G0096753001en_US
dc.identifier.uri (URI) http://nccur.lib.nccu.edu.tw/handle/140.119/38541-
dc.description (描述) 碩士zh_TW
dc.description (描述) 國立政治大學zh_TW
dc.description (描述) 資訊科學學系zh_TW
dc.description (描述) 96753001zh_TW
dc.description (描述) 98zh_TW
dc.description.abstract (摘要) 利用VoIP進行網路會談(VoIP Conference)在現今的社會已有越來越流行的趨勢,尤其是遠距離語音會議的應用方面,不但能節省費用,還能同時允許多人上線會談,但隨著上線人數的增加,網路的頻寬需求與連線數量相對倍增,通話延遲時間也因頻道擁塞而難以控制,此時會議的品質(QoS)往往大打折扣,網路的負擔也會大幅增加。現行的主要解決方案為選擇網路會議中處理能力較強的使用者設備,將來自個別說話者的聲音經過混音疊加後再轉送給其他使用者,再利用不同架構的multicasting tree廣播語音封包至所有成員,藉此減少網路頻寬與連線數量的負擔,達到增加同時說話人數的目的。但這些作法在使用者所在位置過於分散、遙遠時,仍然會造成許多節點無法達到所需的通話品質。
     本研究針對在大型網路語音會談中,與會人數過多造成通話品質不佳之問題,提出了解決方法,我們假設會議中大部份時間僅有一個說話者,首先為每位使用者加上靜音消除機制,只有發話者的封包直接送給其他所有與會人員,如此不但降低網路負擔,更可以刪除混音的需求,大幅降低傳輸時間,此外並以分群組(Cluster)的方式將位於同一地區的使用者分為同一群組,群組內以Full Mesh機制相連,如此便能建立一複雜度較小的MLDST (Minimum Loss Diameter Spanning Tree) 樹狀結構的multicasting tree用以廣播語音封包至所有成員,可以進一步控制傳輸時間及封包遺失率。我們根據使用者之間彼此的連線速度與實際距離來分群組,從群組中選出連線能力較佳的節點作為群組轉送點(Cluster Head),透過此轉送點收到聲音後做群組內的發送,並藉由一個multicasting tree將封包在嚴格控制延遲時間及封包遺失率之下,將語音封包廣播至其他所有Cluster Head,以確保VoIP網路會議之品質。
     我們在PlanetLab這種實際網路的實驗平台上進行實驗,評估本方法的效能,實驗結果顯示我們的方法可以在與會人數達到20人時,仍能有效控制延遲時間在350ms左右,若利用Google雲端運算系統的協助,平均延遲時間將可減少50ms至70ms。由於PlanetLab上的電腦性能普遍不佳,我們預期本技術應用於實際網路時,可以增加與會人數。
zh_TW
dc.description.tableofcontents 中文摘要 i
     Abstract iii
     誌謝辭 v
     目錄 vi
     圖目錄 viii
     表目錄 x
     第一章 緒論 1
     1.1 大型網路語音會談 1
     1.2 研究動機及方法 1
     1.3 VoIP原理 2
     1.3.1 取樣及編碼 2
     1.4 混音 3
     1.5 影響VoIP通話品質之參數 5
     1.6 語音會談常見問題 6
     1.7 語音會談品質評量指標 11
     第二章 背景與相關研究 14
     2.1 語音會談網路架構 15
     2.1.1 Centralized Server 15
     2.1.2 Full Mesh Broadcasting 16
     2.1.3 Coupled Distributed Processing 17
     2.1.4 P2P Multicasting 18
     2.1.5 peerTalk: A Peer-to-Peer Multi-Party Voice-Over-IP System 19
     2.2 現有傳輸機制比較 21
     第三章 Cluster-Based Multicasting Scheme 23
     3.1 設計理念 23
     3.2 CMT設計流程 28
     3.2.1. 分群機制 28
     3.2.2. 群首篩選 30
     3.2.3. 建立Multicasting Tree 31
     3.3 雲端計算 34
     3.3.1 系統架構 35
     第四章 效能評估 38
     4.1 實驗評估指標 38
     4.2 實驗環境 38
     4.3 實驗一 39
     4.3.1 實驗目標 39
     4.3.2 實驗結果與分析 39
     4.4 實驗二:雲端計算 48
     4.4.1 實驗目標 48
     4.4.2 實驗結果與分析 49
     第五章 結論 54
     參考文獻 55
zh_TW
dc.language.iso en_US-
dc.source.uri (資料來源) http://thesis.lib.nccu.edu.tw/record/#G0096753001en_US
dc.subject (關鍵詞) 語音通話zh_TW
dc.subject (關鍵詞) 點對點zh_TW
dc.title (題名) 基於群組模式下之大型網路語音會談系統zh_TW
dc.title (題名) A cluster-based transmission scheme for large scale VoIP conferencingen_US
dc.type (資料類型) thesisen
dc.relation.reference (參考文獻) [1] S. Banerjee, B. Bhattacharjee, and C. Kommareddy, "Scalable application layer multicast", in Proc. of ACM SIGCOMM, Pittsburgh, PA, August 2002.zh_TW
dc.relation.reference (參考文獻) [2] A. Bharambe, V. Padmanabhan, and S. Seshan, "Supporting Spectators in Online Multiplayer Games", in Proc. of HOTNETSIII, San Diego, November 2004.zh_TW
dc.relation.reference (參考文獻) [3] Uyless Black, "VOICE OVER IP", New Jersey: Prentice Hall, 2000.zh_TW
dc.relation.reference (參考文獻) [4] M. Castro, P. Druschel, A.-M. Kermarrec, A. Nandi, A. Rowstron, and A. Singh, "SplitStream: High-Bandwdith Multicast in Cooperative Environments", in Proc. of ACM SOSP 2003, New York, October 2003.zh_TW
dc.relation.reference (參考文獻) [5] M. Castro, P. Druschel, A.-M. Kermarrec, and A. Rowstron, "SCRIBE: A large-scale and decentralized application-level multicast infrastructure", in Proc. of ACM SIGCOMM, San Diego, CA, August 2001.zh_TW
dc.relation.reference (參考文獻) [6] Li-Chen Chi, "Echo Cancellation in Large-Scale VoIP conferencing", 2009.zh_TW
dc.relation.reference (參考文獻) [7] Wei-Chung Chiu, "Quality Assurance of Streaming Data Dissemination over P2P Network", 2009.zh_TW
dc.relation.reference (參考文獻) [8] Y. H. Chu, S. G. Rao, S. Seshan, and H. Zhang, "Enabling Conferencing Applications on the Internet using an Overlay Multicast Architecture", in Proc. of ACM SIGCOMM, San Diego, CA, August 2001.zh_TW
dc.relation.reference (參考文獻) [9] B. Goode, "Voice over Internet Protocol (VoIP)", in Proc. OF THE IEEE, VOL.90, NO.9, Sep. 2002.zh_TW
dc.relation.reference (參考文獻) [10] Xiaohui Gu, Z. Wen, Philip S. Yu, and Zon-Yin ShaeZhen, "peerTalk: A Peer-to-Peer Multi-Party Voice-Over-IP System", in Parallel and Distributed Systems, IEEE Transactions, April 2008.zh_TW
dc.relation.reference (參考文獻) [11] Xiaohui Gu, Z. Wen, Philip S. Yu, and Zon-Yin ShaeZhen, "Supporting Multi-Party Voice-Over-IP Services with Peer-to-Peer Stream Processing", in Proc. of ACM Multimedia, Singapore, November 2005.zh_TW
dc.relation.reference (參考文獻) [12] D. Kostic, A. Rodriguez, J. Albrecht, and A. Vahdat, "Bullet: High Bandwidth Data Dissemination Using an Overlay Mesh", in Proc. of ACM SOSP 2003, New York, October 2003.zh_TW
dc.relation.reference (參考文獻) [13] J. Lennox and H. Schulzrinne, "A Protocol for Reliable Decentralized Conferencing", in NOSSDAV`03, Monterey, California, USA, June. 2003.zh_TW
dc.relation.reference (參考文獻) [14] B. Li and H. Yin, "Peer-to-Peer Live Video Streaming on the Internet: Issues, Existing Approaches, and Challenges", in IEEE Communications Magazine, Toronto, Ont., Canada, June 2007.zh_TW
dc.relation.reference (參考文獻) [15] M. Radenkosvic and C. GreenHalgh, "Multi-party Distributed Audio Service with TCP Fairness", in Proc. of ACM Multimedia, 2002, Juan-les-Pins, France, December 2002.zh_TW
dc.relation.reference (參考文獻) [16] Google Talk, http://www.google.com/talk/intl/zh-TW/.zh_TW
dc.relation.reference (參考文獻) [17] G.723, http://www.itu.int/rec/T-REC-G.723/e.zh_TW
dc.relation.reference (參考文獻) [18] G.729, http://www.itu.int/rec/T-REC-G.729/e.zh_TW
dc.relation.reference (參考文獻) [19] Mean Opinion Score, http://en.wikipedia.org/wiki/Mean_opinion_score.zh_TW
dc.relation.reference (參考文獻) [20] Skype, http://www.skype.com.zh_TW
dc.relation.reference (參考文獻) [21] The E-Model, http://www.itu.int/rec/T-REC-G.107.zh_TW