Aggregate Traffic Models for VoIP Applications

Hassan, Hoda; Garcia, Jean Marie; Bockstal, C.

doi:10.1109/icdt.2006.15

Cited by 24 publications

(11 citation statements)

References 4 publications

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“…The maximum tolerable packet delay for VoIP is set to 30ms. Acceptable VoIP service is maintained for a packet timeout ratio less than 4% [10]. Users exceeding this ratio are in QoS outage.…”

Section: A Channel and Simulation Parametersmentioning

confidence: 99%

Scheduling Techniques for Improving Call Capacity for VoIP Traffic in MIMO-OFDMA Networks

Nicolaou

Doufexi

Armour

et al. 2009

2009 IEEE 70th Vehicular Technology Conference Fall

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Abstract-Fourth Generation Networks will almost invariably adopt OFDMA (Orthogonal Frequency Division Multiple Access) and MIMO (Multiple Input-Multiple Output technologies, in order to meet high data rate and Quality of Service (QoS) requirements. The Worldwide Interoperability for Microwave Access (WiMAX) MAC Layer, which is based on the IEEE 802.16 standard, is designed to support a variety of applications, including voice and multimedia services. The problem of providing QoS in broadband wireless systems is one of managing radio resources effectively. Efficient scheduling algorithms that balance the QoS requirements of each application and user with the available radio resources need to be developed. This paper considers a number of scheduling policies concentrating on real-time Voice over IP (VoIP) traffic. Via numerical results we show that the conventional notion of fairness fails to guarantee service for low latency applications such as VoIP for an increasing traffic load. We show better QoS is achieved via a greedy scheduling approach that manages to serve packets faster.

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Section: A Channel and Simulation Parametersmentioning

confidence: 99%

Scheduling Techniques for Improving Call Capacity for VoIP Traffic in MIMO-OFDMA Networks

Nicolaou

Doufexi

Armour

et al. 2009

2009 IEEE 70th Vehicular Technology Conference Fall

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“…Each VoIP session can be either in the active or inactive state. During the active state, fixed sized packets of 32 bytes are generated at a constant interval of 20ms [10]. Table 1 a summary of the traffic distribution parameters for VoIP is presented.…”

Section: A Traffic Characteristicsmentioning

confidence: 99%

“…Packets that exceed this delay constraint are assumed to time-out and hence are removed from the buffer queue. QoS outage for VoIP is registered for users whose average packet timeout ratio over a session exceeds 4% [10]. The aggregate system's QoS performance is measured in terms of the satisfied customer ratio (i.e.…”

Section: A Traffic Characteristicsmentioning

confidence: 99%

“…Therefore, excessive packet timeouts not only result in a degradation of the perceived QoS but also give rise to poor utilisation of the available radio resources. An upper bound for tolerable packet timeout ration is set at 4% [10]. Users exceeding this ratio are assumed to be in QoS outage.…”

Section: A Traffic Characteristicsmentioning

confidence: 99%

“…A two state Markov process is used to model VoIP, representing the active voice and silent periods respectively [10]. The alternating periods of activity and silence are exponentially distributed with average durations β 1 (β being the parameter of the exponential law of the active period) and α 1 respectively (α being the parameter of the exponential law of the inactive period).…”

Section: A Traffic Characteristicsmentioning

confidence: 99%

See 2 more Smart Citations

Unitary beamforming techniques enhancing VoIP call admittance capacity

Nicolaou

Doufexi

Armour

et al. 2010

21st Annual IEEE International Symposium on Personal, Indoor and Mobile Radio Communications

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Abstract:-Fourth Generation Networks will invariably adopt MIMO-OFDMA techniques, in order to cope with increased data rate demands and improved Quality of Service (QoS) requirements. OFDMA is a popular multiple access candidate, since it facilitates multi-user diversity by enabling two dimensional multiple access, in time and frequency domains. Random Beamforming enables exploitation of spatial multi-user diversity and spatial capacity multiplexing gain. Layered Random Beamforming is able to achieve further multiple access benefits by enabling different users to be scheduled on different spatial layers of the same time-frequency resource. This paper examines the QoS performance of SU-MIMO and MU-MIMO precoding schemes for a VoIP system under a realistic traffic and channel environment. A novel Link Adaptation algorithm operating under PER constraints for delay sensitive VoIP traffic that promises increased data rates, flexibility and improved QoS is proposed and examined in this paper.

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Admission control mechanism for quality of service and security in H.323 voice gateway

EL-Ashri

Badran

Zaki

et al. 2021

Concurrency and Computation

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The rapid demand on the VOIP communication lead to the increase of the development of information security and the offered quality. Therefore, information technology has increased the deepening and developing of call admission control (CAC). In which, admission decisions and resource reservations are taken at the edge voice gateway, rather than resources or individual routers within the network. This means that, CAC decision is made before a voice call is established and it depends on the availability of the required call resources to provide suitable quality of service (QoS) for the new call. In this article, a new model for voice network, in order to achieve QoS along with security, is proposed. In the proposed model, the admission control decision is taken place at H.323 voice gateways using resource reservation protocol. It is shown that the admission control decision should be taken at edge H.323 voice gateways. This will lead to provide the accepted security for both, source and destination nodes which were previously authenticated through the identity services engine. It is concluded from the simulation results that the packet loss rate of call flows after implementing the mechanism is almost 0%. In addition, there is no jitter although there is some delay for packet delivery "Latency." The latency observed by the proposed model is still accepted according to the ITU G.114 standard which recommends a maximum of a 150 ms one-way latency. All results are tested in GNS3 based simulation to evaluate the performance of voice gateway based upon CAC.

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Aggregate Traffic Models for VoIP Applications

Cited by 24 publications

References 4 publications

Scheduling Techniques for Improving Call Capacity for VoIP Traffic in MIMO-OFDMA Networks

Scheduling Techniques for Improving Call Capacity for VoIP Traffic in MIMO-OFDMA Networks

Unitary beamforming techniques enhancing VoIP call admittance capacity

Admission control mechanism for quality of service and security in H.323 voice gateway

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