基於行動裝置之分散式多人會議系統

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姓名

陳學儀(Chen, Hsueh-Yi) 查詢紙本館藏

畢業系所

通訊工程學系

論文名稱

基於行動裝置之分散式多人會議系統
(Mobile-based Decentralized Multiparty Conference System)

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摘要(中)

現今，市面上有許多企業已推出多人會議即時對談應用，例如Skype、LINE、Web Real-Time Communication ( WebRTC ) 等。在這些應用中，架構可分為集中式與去中心化兩種。
目前主流是雲端技術，為集中式架構，使用多台伺服器進行影像、語音的轉送。在這樣的架構下，雖然可以降低每一位會議的參與者使用之頻寬與運算需求，且藉由伺服器強大的運算能力，能夠支援大量的會議參與者，但伺服器的建置與維護需大量開銷。
反之，在去中心化架構下，每一位會議參與者之間直接交換影像、語音資料，無需透過額外的伺服器。雖然避免了維護伺服器的成本，但隨著與會者的增加，頻寬也逐漸不敷使用，因此無法同時有大量的參與者。若每一與會者的裝置運算能力不高時，與會者的數量將變得更稀少。
在這篇論文中，我們比較、分析主流的集中式與去中心化架構，將集中式架構套用於去中心化架構之中，並結合了聲音的混合 ( Mix ) 與解混合 ( Demix )，提出新的網路拓樸應用於普及的行動裝置之上，設計與實作出純行動裝置之間直接進行多人會議系統。

摘要(英)

Nowadays, many enterprises have already published the multiparty conference real-time applications such as Skype, LINE, ZOOM, and Web Real-Time Communication (WebRTC). Among these multiparty conference applications, there are two kinds of architectures: centralized and decentralized architectures.
Currently, the popular technology is cloud computing, which is centralized architecture. It uses servers to relay the video and audio data among devices. Such architecture can reduce the bandwidth and CPU loading of each participant in the conference, and support large amount of participants in a conference by powerful servers. As the number of participants increases, the server also requires more computing power.
In contrast, with decentralized architecture, each participant in conference directly exchanges the data of video and audio with each other without the assistance from server. Although it does not need the cost of maintaining server, the bandwidth and computation power required increases as the number of participants increases. Consequently, the number of participants is limited by the resource limitation.
In this thesis, we compare and analyze the centralized and decentralized architectures, integrate the advantages of centralized and decentralized architectures, and utilize the mixing and demixing technologies for audio traffic. More specifically, this thesis proposes a new network topology for mobile devices to support multiparty conference.

關鍵字(中)

★ 多人會談
★ 語音
★ 會談

關鍵字(英)

★ Multiparty conference
★ voip
★ conference

論文目次

中文摘要 i
ABSTRACT ii
CONTENTS iii
LIST OF FIGURES iv
LIST OF TABLES v
1. INTRODUCTION 1
2. RELATED WORKS 4
2-1 Single Chatting 4
2-2 Multiparty Conference 4
2-3 Open Source Libraries 6
2-3-1 WebRTC 6
2-3-2 Opus Codec 7
2-3-3 WavPack 10
2-3-4 Mixing and Demixing 10
3. DESIGN AND MECHANISM 13
3-1 Buffer Reordering Mechanism 15
3-2 Lossless Codec And Lossy Codec 18
3-3 Server-based Topology 20
3-4 Host-based Topology 21
3-5 Mesh-based Topology 22
3-6 Cooperation-based Topology 23
3-7 Comparison of Topologies 25
4. PERFORMANCE AND EVALUATION 27
4-1 Latency 27
4-2 Evaluation 28
5. CONCLUSIONS AND FUTURE WORK 30
6. REFERENCES 31

參考文獻

[1] Voice over IP. https://en.wikipedia.org/wiki/Voice_over_IP.
[2] Skype home page. http://www.skype.com/zh-Hant/.
[3] LINE home page. http://line.me/zh-hant/.
[4] ZOOM home page. https://zoomnow.tw/.
[5] WebRTC home page. https://webrtc.org/.
[6] Opus Codec home page. https://www.opus-codec.org/.
[7] Valin, J., et al. "RFC 6716: Definition of the Opus Audio Codec." Internet engineering task force (IETF) standard (2012).
[8] WavPack home page. http://www.wavpack.com/
[9] J. Bolot, “End-to-End Packet Delay and Loss Behavior in the Internet,” Proc. ACM SIGCOMM, 1993, pp. 289-298.
[10] Kuo, Chia-Chen, Ming-Syan Chen, and Jeng-Chun Chen. "An adaptive transmission scheme for audio and video synchronization based on real-time transport protocol." IEEE, 2001.
[11] Audio mixing. https://en.wikipedia.org/wiki/Audio_mixing
[12] Chang, Jen-Chun, and Wanjiun Liao. "Application-Layer Conference Trees for Multimedia Multipoint Conferences Using Megaco/H. 248." IEEE, 2001.
[13] Li, Jin. "Mutualcast: A serverless peer-to-peer multiparty real-time audio conferencing system." Multimedia and Expo, 2005. ICME 2005. IEEE International Conference on. IEEE, 2005.
[14] Ben Khedher, D. "A Peer-to-Peer self-organizing scheme for multiparty session." Communications (ICC), 2012 IEEE International Conference on. IEEE, 2012.
[15] Pulse-code modulation. https://en.wikipedia.org/wiki/Pulse-code_modulation
[16] Viktor T. Toth. Mixing digital audio https://www.vttoth.com/CMS/index.php/technical-notes/68
[17] Ford, Bryan, Pyda Srisuresh, and Dan Kegel. "Peer-to-Peer Communication Across Network Address Translators." USENIX Annual Technical Conference, General Track. 2005.
[18] Rosenberg, J., J. Weinberger, and C. Huitema. "RFC3489, Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)[S/OL]." (2008): 0-15.
[19] Rosenberg, J., et al. "Rfc 5389: Session traversal utilities for nat (stun)."Internet Engineering Task Force (2008).
[20] Suli, Wang, and Liu Ganlai. "STUNT Technology in P2P Network Application."2012 Fourth International Conference on Computational and Information Sciences. IEEE, 2012.
[21] Wang, Yong, Zhao Lu, and Junzhong Gu. "Research on Symmetric NAT Traversal in P2P applications." Computing in the Global Information Technology, 2006. ICCGI′06. International Multi-Conference on. IEEE, 2006.
[22] Mahy, Rohan, Philippe Matthews, and Jonathan Rosenberg. "RFC 5766: Traversal using relays around NAT (TURN): relay extensions to session traversal utilities for NAT (STUN)." Internet Engineering Task Force (2010).

指導教授

許獻聰(Shiann-tsong Sheu)

審核日期

2016-7-28

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