End-to-end packet delay and loss behavior in the internet
SIGCOMM '93 Conference proceedings on Communications architectures, protocols and applications
Packet audio playout delay adjustment: performance bounds and algorithms
Multimedia Systems
TCP in presence of bursty losses
Proceedings of the 2000 ACM SIGMETRICS international conference on Measurement and modeling of computer systems
Real-time voice communication over the internet using packet path diversity
MULTIMEDIA '01 Proceedings of the ninth ACM international conference on Multimedia
NOSSDAV '02 Proceedings of the 12th international workshop on Network and operating systems support for digital audio and video
Error Spreading: Reducing Bursty Errors in Continuous Media Streaming
ICMCS '99 Proceedings of the IEEE International Conference on Multimedia Computing and Systems - Volume 2
Adaptive playout scheduling and loss concealment for voice communication over IP networks
IEEE Transactions on Multimedia
A literature survey on traffic dispersion
IEEE Network: The Magazine of Global Internetworking
WiCOM'09 Proceedings of the 5th International Conference on Wireless communications, networking and mobile computing
Reward mechanisms for P2P VoIP networks
Information Technology and Management
Enhanced packet loss recovery in voice multiplex-multicast based VoIP networks
Proceedings of the 1st Amrita ACM-W Celebration on Women in Computing in India
A DTN mode for reliable internet telephony
Proceedings of the 21st international workshop on Network and operating systems support for digital audio and video
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Delivery of real time streaming applications, such as voice and video over IP, in packet switched networks is based on dividing the stream into packets and shipping each of the packets on an individual basis to the destination through the network. The basic implicit assumption on these applications is that shipping all the packets of an application is done, most of the time, over a single path along the network. In this work, we present a model in which packets of a certain session are dispersed over multiple paths, in contrast to the traditional approach. The dispersion may be performed by network nodes for various reasons such as load-balancing, or implemented as a mechanism to improve quality, as will be presented in this work. To study the effect of packet dispersion on the quality of voice over IP (VoIP) applications, we focus on the effect of the network loss on the applications, where we propose to use the Noticeable Loss Rate (NLR) as a measure (negatively) correlated with the voice quality. We analyze the NLR for various packet dispersion strategies over paths experiencing memoryless (Bernoulli) or bursty (Gilbert model) losses, and compare them to each other. Our analysis reveals that in many situations the use of packet dispersion reduces the NLR and thus improves session quality. The results suggest that the use of packet dispersion can be quite beneficial for these applications.