Optimal control of variable rate coding in integrated voice/data packet networks
Performance Evaluation
ACM SIGCOMM Computer Communication Review
Performance evaluation of Forward Error Correction in ATM networks
SIGCOMM '92 Conference proceedings on Communications architectures & protocols
End-to-end packet delay and loss behavior in the internet
SIGCOMM '93 Conference proceedings on Communications architectures, protocols and applications
Joint source/channel coding of statistically multiplexed real-time services on packet networks
IEEE/ACM Transactions on Networking (TON)
What video can and cannot do for collaboration: a case study
Multimedia Systems
A reliable multicast framework for light-weight sessions and application level framing
SIGCOMM '95 Proceedings of the conference on Applications, technologies, architectures, and protocols for computer communication
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NOSSDAV '95 Proceedings of the 5th International Workshop on Network and Operating System Support for Digital Audio and Video
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NOSSDAV '01 Proceedings of the 11th international workshop on Network and operating systems support for digital audio and video
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WOWMOM '02 Proceedings of the 5th ACM international workshop on Wireless mobile multimedia
Queuing analysis of simple FEC schemes for voice over IP
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Queuing analysis of simple FEC Schemes for voice over IP
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adaMOS: MOS-adaptive VoIP sources
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ACM Transactions on Embedded Computing Systems (TECS)
On securing real-time speech transmission over the internet: an experimental study
EURASIP Journal on Applied Signal Processing
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International Journal of Computer Applications in Technology
Analysis of FEC Function for Real-Time DV Streaming
AINTEC '07 Proceedings of the 3rd Asian conference on Internet Engineering: Sustainable Internet
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Packet Loss Concealment Algorithm for VoIP Transmission in Unreliable Networks
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On the impact of loss and delay variation on Internet packet audio transmission
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REDUP: a packet loss recovery scheme for real-time audio streaming over wireless IP networks
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Towards more adaptive voice applications
ISoLA'10 Proceedings of the 4th international conference on Leveraging applications of formal methods, verification, and validation - Volume Part I
Real-Time audio quality evaluation for adaptive multimedia protocols
MMNS'05 Proceedings of the 8th international conference on Management of Multimedia Networks and Services
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QoS-sensitive transport of real-time MPEG video using adaptive redundancy control
Computer Communications
Autonomic QoS Optimization of Real-Time Internet Audio Using Loss Prediction and Stochastic Control
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PROTEUS: network performance forecast for real-time, interactive mobile applications
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Survey on application-layer mechanisms for speech quality adaptation in VoIP
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Multimedia Tools and Applications
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The current Internet provides a single class best effort service. From an application's point of view, this service amounts in practice to providing channels with time-varying characteristics such as delay and loss distributions. One way to support real time applications such as interactive audio given this service is to use control mechanisms that adapt the audio coding and decoding processes based on the characteristics of the channels, the goal begin to maximize the quality of the audio delivered to the destinations. In this paper, we describe and analyze a set of such control mechanisms. They include a jitter control mechanism and a combined error and rate control mechanism. These mechanisms have been implemented and evaluated over the Internet and the MBone. Experiments indicate that they make it possible to establish and maintain reasonable quality audioconferences even across fairly congested connections.