英文摘要
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In this paper, we assume that the sender supports a two-mode codec with two audio sample files (PCM and GSM). The sender tracks the packet loss status according to the algorithm-loss rate estimation based on variable frame size and switches two audio sample files in accordance with packet loss status. The proposed algorithm smoothes short term variations in loss rates, while responds quickly to real changes in the loss rate of the traffic. The ability of the algorithm is qualified by the agility, stability and avoidance of frequent switching for two sample files.
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参考文献
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-
A. Takahashi,H. Yoshino,N. Kitawaki(2004).Perceptual QoS assessment technologies for VoIP.IEEE communications magazine.
-
D. Awduche,A. Chin,A. Elwalid,I. Widjaja,X. Xiao(2002).Overview and principles of internet traffic engineering.RFC 3272, Internet Engineering Task Force.
-
D. D. Clark,W. Fang(1998).Explicit allocation of best-effort packet delivery service.IEEE/ACM Trans. Networking.
-
F. Agharebparast,V. C. M. Leung(2003).A new traffic rate estimation and monitoring algorithm for the QoS-enabled internet.GLOBECOM 2002-IEEE Global Telecommunications Conference.
-
G. Huston(2000).Next Steps for the IP QoS architecture.RFC 2990, Internet Engineering Task Force.
-
H. Schulzrinne,S. Casner,R. Frederick,V. Jacobson(2003).A Transport Protocol for Real-Time Applications.RFC 3550, Internet Engineering Task Force.
-
I. Busse,B. Deffner,H. Schulzrinne(1996).Dynamic QoS control of multimedia applications based on RTP.Computer Communications.
-
ITU-T draft,P.VTQ.The assessment of voice transmission quality from protocol analysis information in IP networks.
-
ITU-T Recommendation G.107(2003).The E-Model, a computational model for use in transmission planning.
-
ITU-T Recommendation P.862(2001).Perceptual evaluation of speech quality (PESQ), an objective method for end-to-end speech quality assessment of narrowband telephone networks and speech codes.
-
J. Light ,A. Bhuvaneshwari(2004).Performance analysis of audio codecs over Real-Time Transmission Protocol (RTP) for voice services over internet protocol.Second Annual Conference on Communication Networks and Services Research (CNSR`04).
-
J. Padhye,J. Kurose,D. Towsley,R. Koodli(1999).A model based TCP-friendly rate control protocol.In Proceedings of NOSSDAV.
-
L. Roychoudhuri,E. Al-Shaer,H. Hamed,G. B. Brewster(2003).Audio transmission over the internet: Experiments and observations.ICC 2003-IEEE International Conference on Communications.
-
M. Handley,J. Padhye,J. Widmer(2001).TCP Friendly Rate Control(TFRC): Protocol Specification.Internet-Draft Draft-ietf-tsvwg-tfrc-01.txt, Internet Engineering Task Force.
-
N. Cranley,L. Murphy(2001).Adaptive quality of service for streamed MPEG-4 over the internet.ICC 2001-IEEE International conference on communications.
-
P. Young(1984).Recursive estimation and time-series analysis.Berlin, Springer-Verlag.
-
S. Floyd,M. Handley,J. Padhye,J. Widmer(2000).Equation-based congestion control for unicast application.In Proceedings of SIGCOMM.
-
S. Palacharla,A. Karmouch,S. A. Mahmoud(1997).Design and Implementation of a Real-time Multimedia Presentation System using RTP.COMPSAC`97-21st International Computer Software and Applications Conference.
-
T. Yoshimura,T. Ohya,T. Kawahara,M. Etoh(2002).Rate and Robustness Control with RTP monitoring agent for mobile multimedia streaming.ICC 2002-IEEE International Conference on Communications.
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