2002
DOI: 10.1002/nem.443
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Study of delay patterns of weighted voice traffic of end‐to‐end users on the VoIP network

Abstract: In this paper we study delay patterns of weighted voice traffic of end-to-end users on the Voice over Internet Protocol (VoIP) network. We compare the delay performance of voice traffic which varies with queue management techniques such as First-In First-Out (FIFO) and Weighted Fair Queuing (WFQ) and voice codec algorithms such as G.723 and G.729 and select an optimal algorithm.

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Cited by 7 publications
(7 citation statements)
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References 8 publications
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“…However, with little compromise to quality, it is possible to implement different ITU-T codecs that yield much less required bandwidth per call and relatively a bit higher, but acceptable, end-to-end delay. This can be accomplished by applying compression, silence suppression, packet loss concealment, queue management techniques, and encapsulating more than one voice packet into a single Ethernet frame [3,9,[16][17][18][19][20][21].…”
Section: Voip Traffic Characteristics Requirements and Assumptionsmentioning
confidence: 99%
See 1 more Smart Citation
“…However, with little compromise to quality, it is possible to implement different ITU-T codecs that yield much less required bandwidth per call and relatively a bit higher, but acceptable, end-to-end delay. This can be accomplished by applying compression, silence suppression, packet loss concealment, queue management techniques, and encapsulating more than one voice packet into a single Ethernet frame [3,9,[16][17][18][19][20][21].…”
Section: Voip Traffic Characteristics Requirements and Assumptionsmentioning
confidence: 99%
“…VoIP requires almost no packet loss. In literature, 0.1-5% packet loss was generally asserted [6,[21][22][23]. However, in [24] the required VoIP packet loss was conservatively suggested to be less than 10 K5 .…”
Section: Define Performance Thresholds and Growth Capacitymentioning
confidence: 99%
“…However, with little compromise to quality, it is possible to implement different ITU-T codecs that yield much less required bandwidth per call and relatively a bit higher, but acceptable, end-to-end delay. This can be accomplished by applying compression, silence suppression, packet loss concealment, queue management techniques, and encapsulating more than one voice packet into a single Ethernet frame [5,11,[18][19][20][21][22][23]. Figure 2 illustrates the sources of delay for a typical voice packet.…”
Section: Figure 2 End-to-end Components Of Voipmentioning
confidence: 99%
“…VoIP requires almost no packet loss. In literature 0.1% to 5% packet loss was generally asserted [8,[23][24][25]. However, in [26] the required VoIP packet loss was conservatively suggested to be less than 10 -5 .…”
Section: Define Performance Thresholds and Growth Capacitymentioning
confidence: 99%
“…QoS can be defined as the ability of a network to provide consistent and predictable service to certain traffic type in an IP network in order to give a guaranteed end-to-end service. According to [2,3], important QoS parameters for VoIP include: * Delay: The mouth-to-ear (M2E) delay consists of delays accumulated during packetization, serialization, propagation, dejittering, and queuing. ITU G. 114 specification recommends one-way end-to-end delay to be less than 150 ms for high quality VoIP calls.…”
Section: Qos Provisioning Techniquesmentioning
confidence: 99%