Source and channel rate adaptation for VoIP

Multiplex communications – Pathfinding or routing – Combined circuit switching and packet switching

Reexamination Certificate

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C370S356000

Reexamination Certificate

active

10366944

ABSTRACT:
A coding system and method for a terminal including a multi-rate codec is disclosed. The terminal includes a multi-rate adaptive coder that is capable of transmitting a continuous voice stream transmission at a source code bit rate and a channel code bit rate. A quality of service probing module probes an end-to-end network path of the continuous voice stream transmission to obtain at least one quality of service parameter. A quality of service management module determines at least one constraint associated with the continuous voice stream transmission. An adaptive bit rate algorithm module dynamically adjusts the source code bit rate and the channel code bit rate as a function of the quality of service parameter and the constraint to obtain a maximum value of perceived user performance during the continuous voice stream transmission.

REFERENCES:
patent: 6349197 (2002-02-01), Oestreich
patent: 6577648 (2003-06-01), Raisanen et al.
patent: 6646995 (2003-11-01), Le Strat et al.
patent: 2005/0254508 (2005-11-01), Aksu et al.
patent: 2006/0069553 (2006-03-01), Hakansson et al.
J. Matta, “CATprobe: A tool to estimate congestion and available bandwidth in IP networks”, DoCoMo USA Labs Internal Technical Report, Jun. 2002.
ITU-T Recommendation G.107, “The Emodel, a computational model for use in transmission planning”, Jul. 2002.
N. Kitawaki and K. Itoh, “Pure delay effects on speech quality in telecommunications”, IEEE Journal on Selected Areas in Communications, vol. 9, No. 4, May 1991.
A. Markopoulou, F. Tobagi, M. Karam, “Assessment of VoIP quality over Internet Backbones”, IEEE Infocom, Aug. 2002.
ITU-T Recommendation G.108, “Application of the Emodel: A planning guide”, Sep. 1998.
Y. Liang, N. Farber, B. Girod, “Adaptive playout scheduling using time-scale modification in packet voice communications”, Proc. Of ICASSP 2001.
J. Rosenberg, L. Qiu, H. Schulzrinne, “Integrating packet FEC into adaptive voice playout buffer algorithms on the Internet”, Proc. of Infocom 2000.
J. Matta, “Relating packet losses on the wired and wireless hops to end-to-end path packet loss in IP networks”, Draft, Oct. 2002.
R.G. Cole, J.H. Rosenbluth, “Voice over IP performance monitoring”, In Proceedings of ACM SIGCOMM 2001.
J. W. Seo, S. J. Woo, K.S.Bae, “A study on the application of an AMR speech codec to VoIP”.
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 1.1.0).
J.-C. Bolot , A. Vega Garcia, “Control mechanisms for packet audio in the Internet”, Proc. IEEE Infocom 1996.
J.-C. Bolot, S. Fosse-Parisis, D. Towsley, “Adaptive FEC-based error control for Internet telephony”, Proc. IEEE Infocom 1999, Mar. 1999, New York, NY.
T. Yoshimura, T. Ohya, T. Kawahara, M. Etoh, “Rate and robustness control with RTP monitoring agent for mobile multimedia streaming”, Proc. of ICC 2002, Apr. 28-May 2, 2002, New York, NY.
C. Perkins et al. “RTP payload for redundant audio data”, RFC 2198, Sep. 1997.
S. Lin, D. Costello, Error Control Coding: Fundamentals and Applications, Prentice-Hall, Inc., Englewood Cliffs, NJ 07632, 1983, pp. 170-177.
J. Sjoberg 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, Jun. 2002.
M. Karam, F. Tobaji, “Analysis of the delay and jitter of voice traffic over the Internet”, Jul. 2001.
Genista Corporation, “3G Voice Service Quality, Objective Characterization of WCDMA Voice Quality”, 2001.
ITU-T P.862 “Perceptual evaluation fo speech quality (PESQ), an objective method for end-to-end speech quality assessment of narrowband telephone networks and speech codecs,” Feb. 2001.
W. Jiang and H. Schulzrinne, “Comparison and optimization of packet loss repair methods on VoIP perceived quality under bursty loss,” Proc. of NOSSDAV 2002, May 12-14, 2002, Miami Beach, FL, USA.
M. Kaindl and N. Goortz, “AMR voice transmission over mobile Internet,” Proc. of IEEE ISCASSP, 2002, Orlando, FL.
Boutremans, C., LeBoudec, J.Y., “Adaptive Delay Aware Error Control for Internet Telephony,” EPFL, Lausanne, Switzerland, 2nd IP Telephony Workshop, Columbia University, New York, New York, Apr. 2001.
Boutremans, C., LeBoudec, J.Y., “Adaptive Joint Playout Buffer and FEC Adjustment for Internet Telephony,” EPFL, Lausanne, Switzerland, Technical Report IC/2002/035, May, 2002.

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