High performance TCP in ANSNET
ACM SIGCOMM Computer Communication Review
Voice over IP performance monitoring
ACM SIGCOMM Computer Communication Review
Proceedings of the 2004 conference on Applications, technologies, architectures, and protocols for computer communications
A traffic characterization of popular on-line games
IEEE/ACM Transactions on Networking (TON)
Open issues in router buffer sizing
ACM SIGCOMM Computer Communication Review
Call Admission Control and Traffic Engineering of VoIP
ICDT '07 Proceedings of the Second International Conference on Digital Telecommunications
Empirical evaluation of VoIP aggregation over a fixed WiMAX testbed
Proceedings of the 4th International Conference on Testbeds and research infrastructures for the development of networks & communities
Rtp: audio and video for the internet
Rtp: audio and video for the internet
Perspectives on router buffer sizing: recent results and open problems
ACM SIGCOMM Computer Communication Review
Modelling the Internet Delay Space Based on Geographical Locations
PDP '09 Proceedings of the 2009 17th Euromicro International Conference on Parallel, Distributed and Network-based Processing
ICDT '09 Proceedings of the 2009 Fourth International Conference on Digital Telecommunications
Hybrid testbed for network scenarios
Proceedings of the 3rd International ICST Conference on Simulation Tools and Techniques
A multiplexing scheme for H.323 voice-over-IP applications
IEEE Journal on Selected Areas in Communications
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This work presents a study of RTP multiplexing schemes, which are compared with the normal use of RTP, in terms of experienced quality. Bandwidth saving, latency and packet loss for different options are studied, and some tests of Voice over IP (VoIP) traffic are carried out in order to compare the quality obtained using different implementations of the router buffer. Voice quality is calculated using ITU R-factor, which is a widely accepted quality estimator. The tests show the bandwidth savings of multiplexing, and also the importance of packet size for certain buffers, as latency and packet loss may be affected. The customer's experience improvement is measured, showing that the use of multiplexing can be interesting in some scenarios, like an enterprise with different offices connected via the Internet. The system is also tested using different numbers of samples per packet, and the distribution of the flows into different tunnels is found to be an important factor in order to achieve an optimal perceived quality for each kind of buffer. Grouping all the flows into a single tunnel will not always be the best solution, as the increase of the number of flows does not improve bandwidth efficiency indefinitely. If the buffer penalizes big packets, it will be better to group the flows into a number of tunnels. The router processing capacity has to be taken into account too, as the limit of packets per second it can manage must not be exceeded. The obtained results show that multiplexing is a good way to improve customer's experience of VoIP in scenarios where many RTP flows share the same path.