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.
This is a preview of subscription content, log in via an institution to check access.
Access this chapter
Tax calculation will be finalised at checkout
Purchases are for personal use only
Preview
Unable to display preview. Download preview PDF.
References
IEEE Std 802.11-1999, Part 11: Wireless LAN MAC and Physical Layer specifications. Reference number ISO/IEC 8802-11:1999(E)
Veeraraghavan, M., Cocker, N., Moors, T.: Support of Voice Services in IEEE 802.11 Wireless LANs. In: Proceedings of INFOCOM 2001, Alaska (April 2001)
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)
IEEE 802.11e/D4.0. Wireless MAC and Physical Layer specifications: MAC Enhancements for QoS (November 2002)
ITU-T. One-Way Transmission Time. Recommendation G.114 (February 1996)
ITU-T. Pulse code modulation (PCM) of voice frequencies. Recommendation G.711 (November 1988)
ITU-T. 40, 32, 24, 16 kbit/s Adaptive Differential Pulse Code Modulation (ADPCM). Recommendation G.726 (December 1990)
Schulzrinne, H., Casner, S., Frederick, R., Jacobson, V.: RTP: A Transport Protocol for Real-Time Applications. Request for Comments 1889 (January 1996)
Floyd, S., Fall, K.: Promoting the Use of End-to-End Congestion Control in the Internet. IEEE/ACM Transactions on Networking (August 1999)
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)
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)
Handley, M., Floyd, S., Padhye, J., Widmer, J.: TCP Friendly Rate Control (TFRC): Protocol Specification. Request for Comments 3448 (January 2003)
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)
Chen, M., Zakhor, A.: Rate Control for Streaming Video over Wireless. In: Proceedings of INFOCOM 2004, Hong Kong (March 2004)
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)
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)
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)
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)
Sisalem, D., Schulzrinne, H.: The Loss-Delay based Adjustment Algorithm: a TCP-friendly Adaptation Scheme. In: Proceedings of NOSSDAV 1998, Cambridge, UK (July 1998)
Rhee, I., Ozdemir, V., Fi, Y.: TEAR: TCP Emulation at Receivers – Flow Control for Multimedia Streaming. NCSU Technical Report (April 2000)
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)
Author information
Authors and Affiliations
Editor information
Editors and Affiliations
Rights and permissions
Copyright information
© 2004 Springer-Verlag Berlin Heidelberg
About this paper
Cite this paper
Trad, A., Ni, Q., Afifi, H. (2004). Adaptive VoIP Transmission over Heterogeneous Wired/Wireless Networks. In: Roca, V., Rousseau, F. (eds) Interactive Multimedia and Next Generation Networks. MIPS 2004. Lecture Notes in Computer Science, vol 3311. Springer, Berlin, Heidelberg. https://doi.org/10.1007/978-3-540-30493-7_3
Download citation
DOI: https://doi.org/10.1007/978-3-540-30493-7_3
Publisher Name: Springer, Berlin, Heidelberg
Print ISBN: 978-3-540-23928-4
Online ISBN: 978-3-540-30493-7
eBook Packages: Springer Book Archive