References

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CONACyT

``Consejo Nacional de Ciencia y Tecnología'', México.
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CoolTalk

InSoft, ``CoolTalk'',
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Degener 94

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Demers 89

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Emling 63

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Floyd 93a

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Free Phone

A. Vega García, ``Free Phone - Telephony over the Internet'',
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Garrett 93

M. W. Garrett, M. Vetterli, ``Joint source/channel coding of statistically multiplexed real time services on packet networks'', ACM/IEEE Trans. Networking, vol. 1, no. 1, pp. 71-80, Feb. 1993.

G.I.T

``Networking and Telecommunications Group, College of Computing, Georgia Institute of Technology'',
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Golestani 94

S. Jamaloddin Golestani, ``A Self-Clocked Queueing Scheme for Broadband Applications'', Proc. INFOCOM'94, Toronto, Ontario, Canada, June 1994, pp. 636-646.

GSM

J. Degener, C. Bormann, ``GSM 06.10 lossy speech compression'',
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GSM Code

J. Degener, C. Bormann, ``GSM Implementation'',
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GSM Intro

J. Scourias, ``Overview of the Global System for Mobile Communications'', University of Waterloo.
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Huggins 76

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IETF

``IETF - Internet Engineering Task Force'',
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Internet PHONE

Vocaltec,``Internet PHONE'',
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INRIA

``INRIA - Institut National de Recherche en Informatique et en Automatique'', http://www.inria.fr/

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IVS

T. Turletti, ``IVS - INRIA Videoconferencing System'',
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Jacobson 88

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Jayant 80

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Kroon 92

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Kurose 93

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LPC-10

T. E. Tremain, ``The Government Standard Linear Predictive Coding Algorithm: LPC-10'', Speech Technology, vol. 1, pp. 40-49, Apr. 1982.

MBone

``MBone - Virtual Internet Backbone for Multicast IP'',
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McKenney 91

P. E. McKenney, ``Stochastic Fairness Queueing'', Internetworking: Research and Experience, Vol. 2, pp. 113-131 (1991).

MERCI

``MERCI - Multimedia European Research Conferencing Integration'',
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MICE

``MICE - Multimedia Integrated Conferencing for Europe'',
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Microsoft DVI

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Montgomery 93

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Nagle 87

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NetSpeak

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NeVoT

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Norros 91

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NVAT

``NVAT - Network Video Audio Tool'',
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Parekh 93

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Qualcomm

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RAT

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RealAudio

``RealAudio - Audio on Demand for the Internet'',
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Renard

J. P. Renard, ``High Fidelity Audio Coding'', Lernout & Hauspie Speech Products.

RFC 791

DARPA Internet Program, ``Internet Protocol (IP) specification'', RFC 791, September 1981.

RFC 889

D. Mills, ``Internet Delay experiments'', ARPANET Working Group Request for Comment, December 1983, RFC 889.

Roberts 94

J. W. Roberts, ``Virtual Spacing for Flexible Traffic Control'', Rapport Technique CNET.

RODEO

``Rodeo High-Speed Networks, Open Networks'',
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RSVP

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RTP

H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson, ``RTP: A Transport Protocol for Real-Time Applications'', RFC 1889, January 1996.

RTP Profile

H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson, ``RTP Profile for Audio and Video Conferences with Minimal Control'', RFC 1890, January 1996.

Sanneck 96

H. Sanneck, A. Stenger, K. B. Younes, B. Girod, ``A new technique for audio packet loss concealment'', Proceedings of Global Internet (Jon Crowcroft and Henning Schulzrinne, eds.), (London, England), IEEE, Nov. 1996.
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SCIP

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Shacham 90

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Sherif 93

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Sieckmeyer 95

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SOLIDOR

``Construction of Distributed Systems & Applications''
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TCP

J. Postel, ``Transmission Control Protocol (TCP) specification'', RFC 793, September 1981.

ToolVox

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UDP

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VAT

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VIC

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VocalTec

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Wang 92

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WebPhone

``WebPhone'', http://www.netspeak.com/

Yajnik 95

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Zhang 90

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Résumé

La diffusion des ordinateurs, ajouté à la disponibilité de matériel informatique audio/vidéo bon marché, ainsi qu'à la disponibilité de liaisons à plus haut débit, ont fait surgir l'intérêt d'utiliser le réseau Internet pour envoyer de l'audio et de la vidéo, types de données qui traditionnellement étaient réservés aux réseaux spécialisés à cet effet, et depuis déjà quelques années l'audio et la vidéoconférence sont devenus une pratique courante. Mais la nature même de l'Internet, fait que ce réseau ne soit pas adapté pour la transmission des données temps réel, ceci a comme conséquence que la qualité de l'audio envoyé à travers l'Internet a en moyen une qualité médiocre. Cette thèse s'adresse précisément à l'analyse et solution de ces problèmes pour permettre à une application d'audioconférence ou téléphone sur Internet, d'adapter son comportement pour maintenir une qualité auditive acceptable même dans des cas où le réseau est assez congestionné. Ces solutions, sous la forme de mécanismes de contrôle, ont été implémentés et testés sur le logiciel d'audioconférence et téléphone sur Internet Free Phone que nous avons développé. Un étude sur le comportement qui auraient ces mécanismes dans un Internet qui évoluait pour intégrer la discipline de service Fair Queueing a montré que ces mécanismes, qui seraient encore nécessaires, auraient même un meilleur performance dans ce type de réseau.