Performance analysis of VoIP data over IP networks

The paper presents the results of research and analysis of voice data transmission quality in IP packet networks. It analyses mechanisms allowing for the assessment of packet telephony data transmission quality. Possible transmission quality levels and adequate quality metrics, applicable in the recommen- dations of standardisation organisations, as well as suggested limit values conditioning acceptable voice data transmission quality were indicated and discussed. A packet network model was designed and tested, taking into account VoIP architecture supporting various audio codecs used for voice compression. Transmission mechanisms based on audio codecs G.711, G.723, G.726, G.728 and G.729 were investigated. It was shown that    for delay-sensitive traffic which fluctuates beyond its nominal rate, selected codecs have an advantage over others and allow   for better transmission quality of VoIP traffic with guaranteed bandwidth and delay

Modeling Quality of Service Techniques for Packet-Switched Networks

Quality of service is the ability to provide different priorities to different applications, users or dataflows, or to guarantee a certain level of performance to a dataflow. The chapter uses timed Petri nets to model techniques that provide the quality of service in packet-switched networks and illustrates the behavior of developed models by performance characteristics of simple examples. These performance characteristics are obtained by discrete-event simulation of analyzed models

Performance analysis of VoIP data over IP networks

The paper presents the results of research and analysis of voice data transmission quality in IP packet networks. It analyses mechanisms allowing for the assessment of packet telephony data transmission quality. Possible transmission quality levels and adequate quality metrics, applicable in the recommen- dations of standardisation organisations, as well as suggested limit values conditioning acceptable voice data transmission quality were indicated and discussed. A packet network model was designed and tested, taking into account VoIP architecture supporting various audio codecs used for voice compression. Transmission mechanisms based on audio codecs G.711, G.723, G.726, G.728 and G.729 were investigated. It was shown that    for delay-sensitive traffic which fluctuates beyond its nominal rate, selected codecs have an advantage over others and allow   for better transmission quality of VoIP traffic with guaranteed bandwidth and delay