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  1. NTU Theses and Dissertations Repository
  2. 電機資訊學院
  3. 電信工程學研究所
請用此 Handle URI 來引用此文件: http://tdr.lib.ntu.edu.tw/jspui/handle/123456789/31503
完整後設資料紀錄
DC 欄位值語言
dc.contributor.advisor謝宏昀(Hung-Yun Hsieh)
dc.contributor.authorChung-Wei Lien
dc.contributor.author李中崴zh_TW
dc.date.accessioned2021-06-13T03:13:54Z-
dc.date.available2007-08-10
dc.date.copyright2006-08-10
dc.date.issued2006
dc.date.submitted2006-08-07
dc.identifier.citationREFERENCES
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[24] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, “SIP: Session initiation protocol,” RFC 3261, IETF, June 2002.
[25] H. Schulzrinne and J. Rosenberg, “Signaling for internet telephony,” Columbia Internet telephony research group, Jaunary 1998.
[26] J. Rosenberg and H. Schulzrinne, “Models for multiparty conferencing in SIP,”Internet Draft, Internet Engineering Task Force, November 2000, work In progress.
[27] K. Singh, G. Nair, and H. Schulzrinne, “Centralized conferencing using SIP,”Internet Telephony Workshop, pp. 57–63, April 2001, columbia University, New York. 77
[28] A. Johnston and O. Levin, “Session initiation protocol call control- conferencing for user agents,” Internet Draft, SIPPING Working Group, Internet Engineering
Task Force, February 2004, work In progress.
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[31] V. W. and M. Roelands, “An overlap-add technique based on waveform similarity (WSOLA) for high quality time-scale modification of speech,” Acoustic, Speech, and Signal Processing, 1993. ICASSP-93., 1993 IEEE International Conference on, pp. 554–557, 27-30 April 1993.
[32] H. Schulzrinne and et al., “RTP: A transport protocol for real-time applications,”RFC 1889, IETF, Jaunary 1996.
[33] R. Sparks, “The session initiation protocol (SIP) refer method,” RFC 3515, IETF, April 2003.
[34] R. Mahy and D. Petire, “The session initiation protocol (SIP),” 2004.
[35] W. Jiang, J. N. Lennox, S. Schulzrinne, H. S. K., and X. Wu, “Integrating internet telephony services,” Internet Computing, IEEE, no. 3, pp. 64–72, May-June 2002.
[36] S. Hawwa, “Audio mixing for centralized conferences in a SIP environment,”Multimedia and Expo. 2002. ICME ’02. Proceedings. 2002 IEEE International Conference on, pp. 26–29, August 2002.
[37] P. Koskelainen, H. Schulzrinne, and X. Wu, “A SIP-based conference control framework,” Proceedings of the 12th international workshop on Network and operating sysems support for digital audio and video (NOSSDAV02 ACM), May 12-14 2002, miami Beach, Florida, USA.
[38] W. Wu, B. N., and B. K., “SIP-based vertical handoff between WWANs and WLANs,” Wireless Communications, IEEE [see also IEEE Personal Communications], no. 3, pp. 66–72, June 2005.
[39] S. Tao, K. Xu, A. Estepa, T. Fei, L. Gao, and R. Guerin, “Improving VoIP quality through path switching,” INFOCOM 2005. 24th Annual Joint Conference of the IEEE Computer and Communications Societies. Proceedings IEEE, pp. 2268–2278, March 13-17 2005.
[40] J. suk Kang, Y. do Kang, I. ho Na, and Y. sung Choi, “A study of subjective speech quality measurement over VoIP network,” Info-tech and Info-net, 2001. Proceedings IC—— 2001-Beijing. 2001 International Conferences on, pp. 311–316, 29 October - 1 November 2001. 78
[41] Y. J. Liang, N. Farber, and B. Girod, “Adaptive playout scheduling and loss concealment for voice communication over IP network,” Multimedia, IEEE Transactions on, no. 4, pp. 532–543, December 2003.
[42] WiFly, http://www.wifly.com.tw/.
[43] D. Bigorgne, O. Boeffard, B. Cherbonnel, F. Emerard, D. Larreur, L. S.-M. J.L., and I. Metayer, “Multilingual PSOLA text-to-speech system,” Acoustic, Speech, and Signal Processing, 1993. ICASSP-93., 1993 IEEE International Conference on, pp. 187–190, 27-30 April 1993.
[44] Wireless Research API (WRAPI), http://sysnet.ucsd.edu/pawn/wrapi/.
[45] iptel.org legend, SIP Express Router, http://www.iptel.org/ser/.
[46] A. Rix, J. Beerends, M. Hollier, and A. Hekstra, “Perceptual evaluation of speech quality (PESQ)- a new method for speech quality assessment of telephone networks
and codecs,” Acoustic, Speech, and Signal Processing 2001. Proceedings. (ICASSP’ 01) IEEE, ITU-T Recommendation P.862, pp. 749–752, 7-11 May 2001.
[47] D. A. Joseph, B. S. Manoj, and C. S. R. Murthy, “Interoperability of Wi-Fi hotspots and cellular networks,” Proceedings of the 1st ACM international
workshop on Wireless mobile applications and services on WLAN hotspots (WMASH’04), October 1 2004.
[48] P. Rodriguez, R. Chakravorty, J. Chesterfield, I. Pratt, and S. Banerjee, “MAR: A commuter router infrastructure for the mobile internet,” Proceedings of the 2nd international conference on Mobile systems, applications, and services (ACM MobiSys ’04), June 2004.
[49] H. Schulzrinne and E. Wedlund, “Application-layer mobility using SIP,” Service Portability and Virtual Customer Environments, 2000 IEEE, pp. 29–36, 1 December 2000.
[50] J. Lennox, K. Murakami, and M. Karaul, “Interworking internet telephony and wireless telecommunications networks,” Computer Communication Review, ACM SIGCOMM, no. 5, October 2001.
[51] O. Haase, Kazutaka, Murakami, and T. F. Laporta, “Unified mobility managerenabling efficient SIP/UMTS mobile network control,” Wireless Communications,
IEEE [see also IEEE Personal Communications], no. 4, pp. 66–75, August 2003.
[52] S. R. Ahuja and J. R. Ensor, “VoIP what’s it good for?” Queue, no. 6, Semptember 2004. 79
[53] J. Rosenberg, “A framework for conferencing with the session initiation protocol,” Internet Draft, Internet Engineering Task Force, October 2003, work In progress.
[54] I. Miladinovic and J. Stadler, “Multiparty conferencing signaling using the session initiation protocol (SIP),” Internet Draft, Internet Engineering Task Force, October 2001, vienna University.
[55] S. Rein, F. H., P. Fitzek, and M. Reisslein, “Voice quality evalution in wireless packet communication systems: A tutorial and performance results for
rohc,” Wireless Communication, IEEE [see also IEEE Personal Communications], no. 1, pp. 60–67, February 2005, arizona State University, USA. 80
dc.identifier.urihttp://tdr.lib.ntu.edu.tw/jspui/handle/123456789/31503-
dc.description.abstractThe objective of the thesis is to design and implement algorithms for seamless vertical handoff between GSM and VoWLAN on dual-mode mobile devices. Due to the low cost of WLAN chips, in the future everyone is expected to own one or more dual-mode handsets equipped with the GSM and WLAN modules. Although the users can access GSM and WLAN simultaneously, they still won't feel convenient without the interoperability between GSM and WLAN. For example, people wish to use VoIP through WLAN access to save money but use GSM audio service to maintain session continuity. Currently, there are several research endeavors working toward solving the vertical handoff problems between GSM and VoWLAN access, including proprietary and standardized solutions. However, current proprietary solutions are designed for specific dual-mode clients and the users cannot enjoy the vertical handoff services without the specific handsets. Standardized solutions are designed for GSM telephony operators and hence vertical handoff services will fail if the users are not within the coverage of corresponding WLAN service. Therefore, we need to design a flexible vertical handoff solution to satisfy all situations, including all kinds of dual-mode clients, different WLAN operators, and different GSM operators.
To design our intelligent vertical handoff algorithm, we conduct several experiments to figure out potential problems during the vertical handoff procedure. By way of the experimental results we understand that the problems of deciding handoff timing and
synchronizing two different audio streams are the most critical issues. To perform seamless vertical handoff between VoWLAN and GSM accesses, we design the handoff decision mechanism to ensure that the GSM and VoWLAN coverage will overlap. By utilizing the audio time-scaling method, we design an audio mixing mechanism to ensure seamless audio switching between GSM and VoWLAN. To implement the algorithm on a dual-mode client, we make several modifications on related entities. At the remote client side, we let it respond to the handoff querying by using winsock programming. At the dual-node client side, we do the handoff triggering and modify the SIP softphone to perform audio waveform tapping for controlling and buffering the instant audio streams of GSM and VoWLAN. The results show that without audio mixing algorithm, there is audio block for more than three hundred milliseconds. With audio mixing mechanism, on the other hand, there is no audio gap during the vertical handoff procedure and the audio streams can be seamlessly migrated. Hence, the proposed vertical handoff algorithms with audio mixing can help the dual-mode client seamlessly switch the audio path and achieve seamless vertical handoffs between GSM and VoWLAN.
en
dc.description.provenanceMade available in DSpace on 2021-06-13T03:13:54Z (GMT). No. of bitstreams: 1
ntu-95-R93942109-1.pdf: 4684713 bytes, checksum: 8ba3ef3e3e21a58af81d374086a261c2 (MD5)
Previous issue date: 2006
en
dc.description.tableofcontentsLIST OF TABLES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . iv
LIST OF FIGURES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . v
ABSTRACT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . vii
1 INTRODUCTION . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
2 BACKGROUND . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
2.1 Voice over IP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
2.1.1 Basic SIP Architecture . . . . . . . . . . . . . . . . . . . . . 6
2.1.2 SIP Extensions . . . . . . . . . . . . . . . . . . . . . . . . . . 12
2.2 GSM Infrastructures and Interoperability with IP Networks . . . . . 13
2.2.1 GSM Infrastructures . . . . . . . . . . . . . . . . . . . . . . . 13
2.2.2 SIP Gateway . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
2.2.3 Interoperability between GSM and SIP . . . . . . . . . . . . 16
2.3 Current Solutions for Vertical Handoff . . . . . . . . . . . . . . . . . 18
2.3.1 Proprietary Solutions . . . . . . . . . . . . . . . . . . . . . . 18
2.3.2 Unlicensed Mobile Access (UMA) . . . . . . . . . . . . . . . 19
2.3.3 Solutions of Researched Communities . . . . . . . . . . . . . 27
3 CHALLENGES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
3.1 SIP for Vertical Handoffs . . . . . . . . . . . . . . . . . . . . . . . . 32
3.2 Handoff Decision Problem . . . . . . . . . . . . . . . . . . . . . . . . 33
3.2.1 Call Setup Time Measurement . . . . . . . . . . . . . . . . . 34
3.2.2 Voice Quality Measurement . . . . . . . . . . . . . . . . . . . 36
3.3 Audio Synchronization Problem . . . . . . . . . . . . . . . . . . . . 38
4 ALGORITHM DESIGN . . . . . . . . . . . . . . . . . . . . . . . . . 42
4.1 Vertical Handoff Procedure . . . . . . . . . . . . . . . . . . . . . . . 43
4.2 Handoff Decision . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46
4.3 Audio Mixing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46
4.3.1 Audio Time-Scaling Algorithm . . . . . . . . . . . . . . . . . 49
4.4 Seamless Vertical Handoff Algorithm . . . . . . . . . . . . . . . . . . 53
4.4.1 Vertical Handoff Procedure from GSM to VoWLAN . . . . . 53
4.4.2 Vertical Handoff Procedure from VoWLAN to GSM . . . . . 55
5 IMPLEMENTATION . . . . . . . . . . . . . . . . . . . . . . . . . . . 57
5.1 Dual-Mode Client Modification . . . . . . . . . . . . . . . . . . . . . 57
5.1.1 Handoff Trigger . . . . . . . . . . . . . . . . . . . . . . . . . 57
5.1.2 Auto-Answer Mode . . . . . . . . . . . . . . . . . . . . . . . 58
5.1.3 Waveform Tapping . . . . . . . . . . . . . . . . . . . . . . . 60
5.2 Remote Client Modification . . . . . . . . . . . . . . . . . . . . . . . 63
6 TESTBED EVALUATION . . . . . . . . . . . . . . . . . . . . . . . . 64
6.1 Testbed Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64
6.2 Vertical Handoff Operations . . . . . . . . . . . . . . . . . . . . . . 65
6.3 Results and Evaluation . . . . . . . . . . . . . . . . . . . . . . . . . 66
6.3.1 Audio Waveforms . . . . . . . . . . . . . . . . . . . . . . . . 67
6.3.2 Metric Evaluation . . . . . . . . . . . . . . . . . . . . . . . . 70
7 CONCLUSION AND FUTURE WORK . . . . . . . . . . . . . . . 75
REFERENCES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76
dc.language.isoen
dc.title支援雙模行動裝置語音通訊之無縫垂直換手機制設計與實作zh_TW
dc.titleDesign and Implementation of Seamless Vertical Handoff Algorithms for Voice Communications on Dual-Mode Mobile Devicesen
dc.typeThesis
dc.date.schoolyear94-2
dc.description.degree碩士
dc.contributor.oralexamcommittee謝宏昀(Hung-Yun Hsieh),廖婉君(Wan-jiun Liao),逄愛君(Ai-Chun Pang)
dc.subject.keyword垂直換手,無線網路,行動電話網路,zh_TW
dc.subject.keywordvertical handoff,GSM,VoWLAN,en
dc.relation.page80
dc.rights.note有償授權
dc.date.accepted2006-08-08
dc.contributor.author-college電機資訊學院zh_TW
dc.contributor.author-dept電信工程學研究所zh_TW
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