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. |
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). |