ABSTRACT
We present a dynamic joint source and channel coding adaptation algorithm for the AMR speech codec based on the ITU-T Emodel. This model takes both delay and packet loss into consideration. We address the problem of finding the optimal choice of source and channel bit rates given QoS information about the wired and wireless IP network and subject to constraints on maximum packet loss, maximum delay and maximum allowed transmission rate. Our results show that an adaptation is necessary to preserve acceptable levels of quality while making optimal use of the allowed bandwidth. Our technique requires a small number of computations that allows real time operation in parallel to voice streams.
- Jain, M., and Dovrolis, C. End-to-end available bandwidth: Measurement methodology, dynamics, and relation with TCP throughput. In Proceedings of ACM Sigcomm 2002 (Pittsburgh, PA, August 2002). Google ScholarDigital Library
- ITU-T Recommendation G.107. The Emodel, a computational model for use in transmission planning. July 2002.Google Scholar
- Kitawaki, N., and Itoh, K. Pure delay effects on speech quality in telecommunications. IEEE Journal on Selected Areas in Communications, 9, 4 (May 1991).Google ScholarDigital Library
- Markopoulou, A., Tobagi, F., and Karam, M. Assessment of VoIP quality over Internet Backbones. In Proceedings of IEEE Infocom 2002 (New York, NY, June 2002).Google ScholarCross Ref
- ITU-T Recommendation G.108. Application of the Emodel: A planning guide. September 1999.Google Scholar
- Rosenberg, J., Qiu, L., and Schulzrinne, H. Integrating packet FEC into adaptive voice playout buffer algorithms on the Internet. In Proceedings of IEEE Infocom 2000 (Tel-Aviv, Israel, March 2000).Google ScholarCross Ref
- Seo, J.W., Woo, S.J., and Bae, K.S. A study on the application of an AMR speech codec to VoIP. In Proceedings of IEEE ICASSP 2001 (Salt Lake City, UT, May 2001). Google ScholarDigital Library
- 3rd Generation Partnership Project; Technical Specification Group Services and System Aspects; Mandatory Speech Codec speech processing functions AMR speech codec; Transcoding functions (3G TS 26.090 version 3.1.0).Google Scholar
- Bolot, J.C., Fosse-Parisis, S., and Towsley, D. Adaptive FEC-based error control for Internet telephony. In Proceedings of IEEE Infocom 1999 (New York, NY, March 1999).Google ScholarCross Ref
- Perkins, C., et al. RTP payload for redundant audio data. RFC 2198, IETF, September 1997. Google ScholarDigital Library
- Lin, S., and Costello D., Error Control Coding: Fundamentals and Applications, Prentice-Hall, Englewood Cliffs, NJ, 1983.Google ScholarDigital Library
- Sjoberg, J., et al. Real-time transport protocol (RTP) payload format and file storage format for the adaptive multi-rate (AMR) and adaptive multi-rate wideband (AMR-WB) audio codecs. RFC 3267, IETF, June 2002. Google ScholarDigital Library
- Karam, M., and 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 ScholarCross Ref
- Genista Corporation. 3G Voice Service Quality, Objective Characterization of WCDMA Voice Quality. 2001.Google Scholar
- ITU-T P.861. Objective quality measurement of telephone-band (300--3400 Hz) speech codecs. February 1998.Google Scholar
- ITU-T P.862. Perceptual evaluation of speech quality (PESQ), an objective method for end-to-end speech quality assessment of narrowband telephone networks and speech codecs. February 2001.Google Scholar
- Jiang, W., and Schulzrinne, H. Comparison and optimization of packet loss repair methods on VoIP perceived quality under bursty loss. In Proceedings of NOSSDAV 2002 (Miami Beach, FL, May 2002). Google ScholarDigital Library
- Kaindl, M., and Görtz, N. AMR voice transmission over mobile Internet. In Proceedings of IEEE ICASSP 2002 (Orlando, FL, May 2002).Google Scholar
- Rosenberg, J., and Schulzrinne, H. An RTP payload format for generic forward error correction. RFC 2733, IETF, December 1999. Google ScholarDigital Library
- Yoshimura, T., Ohya, T., Kawahara, T., and Etoh, M. Rate and robustness control with RTP monitoring agent for mobile multimedia streaming. In Proceedings of IEEE ICC 2002 (New York, NY, April--May 2002).Google ScholarCross Ref
- Boutremans, C., and Le Boudec, J.Y. Adaptive delay aware error control for Internet telephony. In Proceedings of the 2nd IP-Telephony Workshop (New York, NY, April 2001).Google Scholar
- Jiang, W., and Schulzrinne, H. Comparisons of FEC and codec robustness on VoIP quality and bandwidth efficiency. In Proceedings of ICN 2002 (Atlanta, GA, August 2002).Google ScholarCross Ref
- Jiang, W., and Schulzrinne, H. Comparisons of FEC and codec robustness on VoIP quality and bandwidth efficiency. In Proceedings of ICN 2002 (Atlanta, GA, August 2002).Google ScholarCross Ref
- Matta, J., and Takeshita, A. End-to-end voice over IP quality of service estimation through router queuing delay monitoring. In Proceedings of IEEE Globecom 2002 (Taipei, Taiwan, November 2002).Google ScholarCross Ref
- Matta, J., and Jain, R. Extended CAT Probe, US Patent Application, filed March 2003.Google Scholar
- Kohler, E. et al. Datagram congestion control protocol. Internet Draft (work in progress), IETF. March 2003.Google Scholar
- Proceedings of the Second Internet Measurement Workshop (ACM IMW 2002), (Marseille, France, November 2002).Google Scholar
- Cooperative Association for Internet Data Analysis: http://www.caida.org/Google Scholar
Index Terms
- A source and channel rate adaptation algorithm for AMR in VoIP using the Emodel
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