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Adaptive VoIP Transmission over Heterogeneous Wired/Wireless Networks

  • Abdelbasset Trad
  • Qiang Ni
  • Hossam Afifi
Part of the Lecture Notes in Computer Science book series (LNCS, volume 3311)

Abstract

In this paper, we present an adaptive architecture for the transport of VoIP traffic over heterogeneous wired/wireless Internet environments. This architecture uses a VoIP gateway associated with an 802.11e QoS enhanced access point (QAP) to transcode voice flows before their transmissions over the wireless channel. The instantaneous bit rate is determined by a control mechanism based on the estimation of channel congestion state. Our mechanism dynamically adapts audio codec bit rate using a congestion avoidance technique so as to preserve acceptable levels of quality. A case study presenting the results relative to an adaptive system transmitting at bit rates typical of G.711 PCM (64 kbit/s) and G.726 ADPCM (40, 32, 24 and 16 kbit/s) speech coding standards illustrates the performance of the proposed framework. We perform extensive simulations to compare the performance between our adaptive audio rate control and TFRC mechanism. The results show that the proposed mechanism achieves better voice transmission performance, especially when the number of stations is fairly large.

Keywords

Packet Loss Medium Access Control Forward Error Correction Packet Loss Rate Fairness Index 
These keywords were added by machine and not by the authors. This process is experimental and the keywords may be updated as the learning algorithm improves.

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References

  1. 1.
    IEEE Std 802.11-1999, Part 11: Wireless LAN MAC and Physical Layer specifications. Reference number ISO/IEC 8802-11:1999(E) Google Scholar
  2. 2.
    Veeraraghavan, M., Cocker, N., Moors, T.: Support of Voice Services in IEEE 802.11 Wireless LANs. In: Proceedings of INFOCOM 2001, Alaska (April 2001)Google Scholar
  3. 3.
    Chen, D.-Y., Garg, S., Kappes, M., Trivedi, K.S.: Supporting VBR VoIP traffic with IEEE 802.11 WLAN in PCF mode. In: Proceedings of OPNETWork 2002, Washington D.C. (August 2002)Google Scholar
  4. 4.
    IEEE 802.11e/D4.0. Wireless MAC and Physical Layer specifications: MAC Enhancements for QoS (November 2002)Google Scholar
  5. 5.
    ITU-T. One-Way Transmission Time. Recommendation G.114 (February 1996)Google Scholar
  6. 6.
    ITU-T. Pulse code modulation (PCM) of voice frequencies. Recommendation G.711 (November 1988)Google Scholar
  7. 7.
    ITU-T. 40, 32, 24, 16 kbit/s Adaptive Differential Pulse Code Modulation (ADPCM). Recommendation G.726 (December 1990) Google Scholar
  8. 8.
    Schulzrinne, H., Casner, S., Frederick, R., Jacobson, V.: RTP: A Transport Protocol for Real-Time Applications. Request for Comments 1889 (January 1996)Google Scholar
  9. 9.
    Floyd, S., Fall, K.: Promoting the Use of End-to-End Congestion Control in the Internet. IEEE/ACM Transactions on Networking (August 1999)Google Scholar
  10. 10.
    Padhye, J., Firoiu, V., Towsley, D., Kurose, J.: Modeling TCP Throughput: A Simple Model and its Empirical Validation. In: Proceedings of ACM SIGCOMM 1998, Vancouver, Canada (August/September 1998)Google Scholar
  11. 11.
    Floyd, S., Handley, M., Padhye, J., Widmer, J.: Equation-Based Congestion Control for Unicast Applications. In: Proceedings of ACM SIGCOMM 2000, Stockholm, Sweden, pp. 43–56 (August/September 2000)Google Scholar
  12. 12.
    Handley, M., Floyd, S., Padhye, J., Widmer, J.: TCP Friendly Rate Control (TFRC): Protocol Specification. Request for Comments 3448 (January 2003)Google Scholar
  13. 13.
    Cen, S., Cosman, P.C., Voelker, G.M.: End-to-end Differentiation of Congestion and Wireless Losses. In: Proceedings of Multimedia Computing and Networking (MMCN) conf. 2002, San Jose, CA, pp. 1–15 (January 2002)Google Scholar
  14. 14.
    Chen, M., Zakhor, A.: Rate Control for Streaming Video over Wireless. In: Proceedings of INFOCOM 2004, Hong Kong (March 2004)Google Scholar
  15. 15.
    Bolot, J.-C., Fosse-Parisis, S., Towsley, D.: Adaptive FEC-based Error Control for Internet Telephony. In: Proceedings of INFOCOM 1999, New York (March 1999)Google Scholar
  16. 16.
    Boutremans, C., Le Boudec, J.Y.: Adaptive Delay Aware Error Control for Internet Telephony. In: Proceedings of 2nd IP-Telephony workshop, Columbia University, New York (April 2001)Google Scholar
  17. 17.
    Matta, J., Pépin, C., Lashkari, K., Jain, R.: A Source and Channel Rate Adaptation Algorithm for AMR in VoIP Using the Emodel. In: Proceedings of NOSSDAV 2003, San-Francisco, California (June 2003)Google Scholar
  18. 18.
    Rejaie, R., Handley, M., Estrin, D.: RAP: An End-to-end Rate-based Congestion Control Mechanism for Realtime Streams in the Internet. In: Proceedings of IEEE INFOCOM 1999, New York (March 1999)Google Scholar
  19. 19.
    Sisalem, D., Schulzrinne, H.: The Loss-Delay based Adjustment Algorithm: a TCP-friendly Adaptation Scheme. In: Proceedings of NOSSDAV 1998, Cambridge, UK (July 1998)Google Scholar
  20. 20.
    Rhee, I., Ozdemir, V., Fi, Y.: TEAR: TCP Emulation at Receivers – Flow Control for Multimedia Streaming. NCSU Technical Report (April 2000)Google Scholar
  21. 21.
    Karam, M., Tobagi, F.: Analysis of the Delay and Jitter of Voice Traffic over the Internet. In: Proceedings of IEEE INFOCOM 2001, Anchorage, AL (April 2001)Google Scholar

Copyright information

© Springer-Verlag Berlin Heidelberg 2004

Authors and Affiliations

  • Abdelbasset Trad
    • 1
  • Qiang Ni
    • 1
  • Hossam Afifi
    • 2
  1. 1.INRIA, Planete ProjectSophia-AntipolisFrance
  2. 2.INT-INRIAEvryFrance

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