DOI QR코드

DOI QR Code

Enhanced Timing Recovery Using Active Jitter Estimation for Voice-Over IP Networks

  • Kim, Hyoung-Gook (Department of Electronics Convergence Engineering, Kwangwoon University)
  • 투고 : 2011.10.08
  • 심사 : 2012.03.19
  • 발행 : 2012.04.30

초록

Improving the quality of service in IP networks is a major challenge for real-time voice communications. In particular, packet arrival-delay variation, so-called "jitter," is one of the main factors that degrade the quality of voice in mobile devices with the voice-over Internet protocol (VoIP). To resolve this issue, a receiver-based enhanced timing recovery algorithm combined with active jitter estimation is proposed. The proposed algorithm copes with the effect of transmission jitter by expanding or compressing each packet according to the predicted network delay and variations. Additionally, the active network jitter estimation incorporates rapid detection of delay spikes and reacts to changes in network conditions. Extensive simulations have shown that the proposed algorithm delivers high voice quality by pursuing an optimal trade-off between average buffering delay and packet loss rate.

키워드

참고문헌

  1. B. Sat and B. W. Wah, "Analyzing voice quality in popular VoIP applications," 2009 IEEE Computer Society, IEEE Multimedia, vol.16, pp.46-59, Jan.2009. https://doi.org/10.1109/MMUL.2009.2
  2. A. Shallwani, "An adaptive playout algorithm with delay spike detection for real-time VoIP," Department of Electrical and Computer Engineering McCill University Master Thesis, vol. 2, pp. 997-1000, Sep.2003.
  3. Z. Becvar, L. Novak, J. Zelenka, M. Brada and P. Slepicka, "Impact of additional noise on subjective and objective quality assessment in VoIP," 2007 International Workshop on Multimedia Signal Processing, pp.39-42, Oct.2007.
  4. Z. Becvar, J. Zelenka and M. Brada, "Impact of saturation on speech quality in VoIP," 15th International Conference on Systems, Signals and Image Processing, pp.381-384, Jun.2008.
  5. D. Florencio and L.-W. He, "Enhanced adaptive playout scheduling and loss concealment techniques for Voice over IP networks," 2011 IEEE International Symposium on Circuits and Systems, pp.129-132, May.2011.
  6. R. Ramjee, J. Kurose, D. Towsley and H. Schulzrinne, "Adaptive playout mechanisms for packetized audio applications in wide area networks," IEEE Infocom Conference on Computer Communication, vol.2, pp.680-688, Jun.1994.
  7. M. Narbutt and L. Murphy, "VoIP playout buffer adjustment using adaptive estimation of network delays," 18th International Teletraffic Congress (ITC-18), pp.1171-1180, Sep.2003.
  8. A. Kansal and A. Karandikar, "Adaptive delay estimation for low jitter audio over Internet," IEEE Global Telecommunication Conference, vol.4, pp.2591-2595, Nov.2001.
  9. J. Pinto and K. J. Christensen, "An algorithm for playout of packet voice based on adaptive adjustment of talkspurt silence periods," 24th Conference on Local computer Networks, vol.5, pp.224-231, Oct.1999.
  10. Y. Liang, N. Farber and B. Girod, "Adaptive playout scheduling and loss concealment for voice communication over IP networks," IEEE Transactions on Multimedia, vol.5, no.4, pp.532-543, Dec.2003.
  11. C. J. Sreenan, J.-C. Chen, P. Agrawal and B. Narendran, "Delay reduction techniques for playout buffering," IEEE Transactions on Multimedia, vol.2, pp.88-100, Jun.2000. https://doi.org/10.1109/6046.845013
  12. S. Chi and B. F. Womack, "QoS-based optimal adaptive playout buffer scheduling using the packet arrival distribution," IEEE MILCOM, pp.15-19, Oct.2009.
  13. H. Li, G. Zhang and W. Kleijn, "Adaptive playout scheduling for VoIP using the K-Erlang distribution," The 2010 European Signal Processing Conference, pp.1494-1498, Aug.2010.
  14. J. Aragao Jr. and G. Barreto, "Novel approaches for online playout delay prediction in VoIP applications using time series models," Computers & Electrical Engineering, vol.36, pp.536-544, May.2010. https://doi.org/10.1016/j.compeleceng.2009.12.006
  15. P. DeLeon and C. Sreenan, "An adaptive predictor for media playout buffering," IEEE International Conference on Acoustics, Speech, Signal Processing, vol.6, pp.3097-3100, March, 1999.
  16. K. Fujimoto, S. Ata and M. Murata, "Adaptive playout buffer algorithm for enhancing perceived quality of streaming applications," IEEE Global Telecommunication Conference, vol.3, pp. 451-2457, Nov.2002.
  17. N. Aoki, "A VoIP packet loss concealment technique taking account of pitch variation in pitch waveform replication," Electronics and Communications in Japan, vol.89, pp.1-9, Mar.2006.
  18. V. P. Bhute and U. N. Sharawankar, "Speech packet concealment techniques based on time-scale modification for VoIP," ICCSIT 2008 International Conference on Computer Science and Information Technology, pp.825-828, Aug.2008.
  19. J.-H. Chen, "Packet loss concealment for predictive speech coding based on extrapolation of speech waveform," ACSSC 2007 Conference Record of the Forty-First Asilomar Conference on Signal, Systems and Computers, pp.2088-2092, Nov.2007.
  20. K. Kondo and K. Nakagawa, "A speech packet loss concealment method using linear prediction," IEICE Transactions on Information and System, vol.E89-D, no.2, pp.806-813, Feb.2006. https://doi.org/10.1093/ietisy/e89-d.2.806
  21. S. V. Andrsen, W. B. Kleijn, and P. Sorqvist, "Method and arrangement in a communication system," U.S. Patent 7 321 851, Sep.2008.
  22. D. Dorran, "Audio time-scale modification," Dublin Institute of Technology Doctoral Thesis, May.2005.
  23. S. L. Ng, S. Hoh and D. Singh, "Effectiveness of adaptive codec switching VoIP application over heterogeneous networks," 2nd Conference on Mobile Technology, Applications and Systems, Nov.2005.
  24. ITU-T Recommendation P.862(2001) Amendment 2(11/05), http://www.itu.int/rec/T-REC-P.862-200511-I!Amd2

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