The KIPS Transactions:PartC (정보처리학회논문지C)

Korea Information Processing Society (한국정보처리학회)
권호 표지
DOI QR코드

DOI QR Code

An Adaptive FEC based Error Control Algorithm for VoIP

VoIP를 위한 적응적 FEC 기반 에러 제어 알고리즘

  • Choe, Tae-Uk (Dept.of Computer Science, Graduate School of Busan National University) ;
  • Jeong, Gi-Dong (Dept.of Computer Science, Busan National University)
  • 최태욱 (부산대학교 대학원 전자계산학과) ;
  • 정기동 (부산대학교 전자계산학과)
  • Published : 2002.06.01
Download PDF
⟨ Previous Next ⟩

Abstract

In the current Internet, the QoS of interactive applications is hardly guaranteed because of variable bandwidth, packet loss and delay. Moreover, VoIP which is becoming an important part of the information infra-structure in these days, is susceptible to network packet loss and end-to-end delay. Therefore, it needs error control mechanisms in network level or application level. The FEC-based error control mechanisms are used for interactive audio application such as VoIP. The FEC sends a main information along with redundant information to recover the lost packets and adjusts redundant information depending on network conditions to reduce the bandwidth overhead. However, because most of the error control mechanisms do not consider end-to-end delay but packet loss rate, their performances are poor. In this paper, we propose a new error control algorithm, SCCRP, considering packet loss rate as well as end-to-end delay. Through experiments, we confirm that the SCCRP has a lower packet loss rate and a lower end-to-end delay after reconstruction.

현재의 인터넷은 가변적인 대역폭과 패킷손실 그리고 지연으로 인하여 대화식 응용의 QoS 보장이 어렵다. 특히 최근에 정보의 기반구조로 중요성이 강조되고 있는 VOIP는 패킷손실률과 종점간지연이 클 때 통화품질이 크게 떨어지므로 네트웍 수준에서나 응용 수준에서 에러제어 기법이 요구된다. 인터넷 전화와 같은 대화식 응용을 위한 응용 수준의 에러 제어 기법으로 FEC(Forward Error Correction)가 가장 많이 사용되고 있는데, 이 기법은 주정보와 더불어 부가정보를 전송함으로서 패킷손실을 복구하는 방법으로 네트웍의 상태에 따라 적응적으로 부가정보의 양을 조절한다. 그러나 기존의 알고리즘들은 패킷손실률만을 고려하여 부가정보를 조절하였으며 부가 정보를 증가시킬 때 수반되는 종점간지연을 간과함으로써 통화품질을 떨어뜨리는 단점이 있다. 본 논문에서는 패킷손실률뿐만 아니라 종점간지연을 고려하는 FEC기반 에러제어 기법인 SCCRP (Selecting a Codec Combination using Reward and Penalty)를 제안한다. 실험 결과, SCCRP는 다른 알고리즘들에 비해 복구 후 패킷손실률은 물론 복구 후 종점간지연을 낮게 유지하였다.

Keywords

References

  1. S. Blake, D. Blake, M. Carlson, et al, 'An architecture for differentiated service,' RFC 2475, IETF, Dec., 1998
  2. J. Wroclawski, 'Specification of the controlled-load network element service,' RFC 2211, IETF, Sept., 1997
  3. S.Shenker, C. Partridge, and R. Guerin, 'Specification of guaranteed quality of service,' RFC 2212, IETF, Sept. 1997
  4. Ramachandran Ramjee, Jim Kurose, Don Towsley, and Henning Schulz, 'Adaptive playout mechanisms for packetized audio applications in wide-area networks,' in Proc. of IEEE Inforcom, Toronto, Canada, June, 1994 https://doi.org/10.1109/INFCOM.1994.337672
  5. Sue B. Moon, Jim Kurose, and Don Towsley, 'Packet audio playout delay adjustment : performance bounds and algorithms,' ACM/Springer Multimedia Systems, Vol.5, No.1, Jan., 1998 https://doi.org/10.1007/s005300050073
  6. D. Sisalem and H. Schulzrinne, 'The loss-delay baed adjustment algorithm : A TCP-friendly adaptation scheme,' Proc. of NOSSDAV, Cambrideg, England, July, 1998
  7. R. Rejaie, M. Handley, and D. Estrin, 'RAP : An end-to-end rate-based congestion control mechanism for realtime streams in the Internet,' Proc. of IEEE Infocom, Mar. 1999 https://doi.org/10.1109/INFCOM.1999.752152
  8. J. L. Ramsey, 'Realization of optimum interleavers,' IEEE Transactions on Information theory, IT-16, May, 1970
  9. D. J. Goodman, G. B. Lockhart, O. J. Wasem, and W. C. Wong, 'Speeches;and Signal Processing,Waveform substitution techniques for recovering missing speech segments in packet voice communications,' IEEE Transactions on Acoustics, Speeches, and Signal Processing, ASSP-34(6), Dec., 1986
  10. S. Floyd, V. Jacobson, S. McCanne, et al. 'A reliable multicast framework for light-weight sessions and applications level framing,' IEEE/ACM Transactions on Networking, Dec., 1997 https://doi.org/10.1109/90.650139
  11. Nachum Shacham and Paul McKenney, 'Packet recovery in high-speed networks using coding and buffer management,' in Proc. of IEEE Inforcom, San Francisco, California, June, 1990 https://doi.org/10.1109/INFCOM.1990.91241
  12. L. Rizzo, 'Effective erasure codes for reliable computer communication protocols,' ACM Computer Communication Review, Vol.27, April, 1997 https://doi.org/10.1145/263876.263881
  13. J. C. Bolot, and A. B. Garcia, 'Control mechanisms for packet audio in the internet,' in Proc. of IEEE Infom, San Fransisco, California, Mar., 1996 https://doi.org/10.1109/INFCOM.1996.497898
  14. C. Padhye, K. J. Christensen and W. Moreno, 'A new adaptive FEC loss control algorithm for voice over IP applications,' in Proc. of IEEE IPCC00, 2000 https://doi.org/10.1109/PCCC.2000.830332
  15. C. Perkins, O. Hodson, and Hardman, 'A Survey of Packet Loss Recovery Techniques for Streaming Audio,' IEEE Network, Vol.12, No.5, pp.40-48, September, 1998 https://doi.org/10.1109/65.730750
  16. J. Rosenberg and H. Schulzrinne, 'An RTP payload format for generic forward error correction,' IETF Audio/Video Transport Working Group, RTC2198, July, 1998
  17. V. Hardman, M. A. Sasse, M. Handley and A. Watson, 'Relialbe audio for use over the Internet,' in Proceedings of INET'95, 1995
  18. J.-C. Bolot, A. Vega-Garcia, 'The case for FEC based error control for packet audio in the Internet,' to appear in ACM/Springer Multimedia Systems, 1999
  19. ns-Network Simulator. Seehttp://www-mash.cs.berkeley.edu/ns/ns.html
  20. Myoung-Kyoung Ji, Tae-Uk Choi, et al, 'Selecting an audio redundancy codec combination for error control in Internet telephony,' in Proc. of ICPADS2001, KyongJu, Korea, June, 2001 https://doi.org/10.1109/ICPADS.2001.934879
  21. Myoung-Kyoung Ji, Tae-Uk Choi, et al, 'Adeptive FEC error control algorithm using loss and delay for Internet telephony,' in Proc. of ICME2001, Tokyo, Japan, August, 2001