On Modeling VoIP Traffic in Broadband Networks (original) (raw)

Performance of VoIP applications in a simple differentiated services network architecture

2001

IP networks were designed to support non-real time applications, such as file transfer or e-mail. These applications are characterised by their bursty traffic and high bandwidth demands at burst times, but they are not highly sensitive to delay or delay variation (jitter). On the other hand VoIP application requires timely packet delivery with low delay, jitter and packet loss values. Integration of voice and data onto a single network is becoming a priority for many network operators. To achieve that goal IP networks must be enhanced with mechanisms that ensure the quality of service required to carry real-time traffic such as voice. Three parameters emerge as the primary factors affecting voice quality within networks that offer VoIP technologies: clarity, end-to-end delay and echo. To support interactive voice application on an IP network we must be able to control four QoS categories: bandwidth, delay (latency), jitter and packet loss.

A New Model for VoIP Traffic Generation

Anais de VII International Telecommunications Symposium, 2010

In this work a new model for VoIP traffic generation is proposed. The innovation of this model consists in modeling the user behavior instead of the aggregated traffic. We have analyzed the call holding time and the time interval between calls to characterize the user behavior. In order to provide an accurate packet generation, the data nature was modeled by identifying the time for packet transmission and the time interval between packets. Those variables of the proposed model were characterized with probability distributions. The parameters of the distributions were obtained with the analysis of real data collected from two major Brazilian telecommunications carriers. A VoIP traffic simulator was implemented and its results were compared with real data to validate the model. The similarity between synthetic and real data indicates that our model works properly and can be used for VoIP networks modeling and workload generation.

Capacity Allocation for Voice over IP Networks Using Maximum Waiting Time Models

Lecture Notes in Computer Science, 2004

As voice services impose stringent quality of service (QoS) guarantees to perform well over IP networks, large network resources should be allocated to their traffic class. It gets unaffordable when hard guarantees are required as in deterministic-based mechanisms such as the guaranteed services model of the integrated services (IntServ) architecture. However, the amount of network resources could be drastically decreased if only a small number of all voice connections are allowed to be negatively affected. In this work, a new capacity allocation method based on the maximum waiting time model is explored. It is established from the following concept: by providing statistical quality guarantees to those packets that experience the maximum waiting time among all packets of the active voice connections, all other packets are implicitly protected from excess delay and, thus, from service degradation. This method is investigated and mathematically analyzed for the voice service class in converged IP networks.

VoIP Traffic Shaping Analyses in Metropolitan Area Networks

2007

This paper represents VoIP shaping analyses in devices that apply the three Quality of Service techniques -IntServ, DiffServ and RSVP. The results show queue management and packet stream shaping based on simulation of the three mostly demanded services -VoIP, LAN emulation and transaction exchange. Special attention is paid to the VoIP as the most demanding service for real time communication.

Traffic Behaviour of VoIP in a Simulated Access Network

2006

Insufficient Quality of Service (QoS) of Voice over Internet Protocol (VoIP) is a growing concern that has lead the need for research and study. In this paper we investigate the performance of VoIP and the impact of resource limitations on the performance of Access Networks. The impact of VoIP performance in Access Networks is particularly important in regions where Internet

On traffic characteristics and bandwidth requirements of voice over IP applications

2003

Voice over IP (VoIP) services will play an important role in future IP networks, promising cost savings and new revenue sources to operators and service providers. To provide quality of service guarantees, certain mechanisms need to be implemented which support predictable packet handling, bandwidth allocation, and call admission control. In order to configure these mechanisms, a solid understanding of VoIP traffic characteristics and its respective bandwidth requirements is necessary. In this paper, we characterize traffic traces generated by various VoIP applications. According to the H.323 standard, the characteristics are described by means of token bucket parameters which are then used to derive the required service rates for individual traffic flows. While in most cases sources send out fairly steady packet streams, there are situations where software-based clients emit rather bursty traffic resulting in unreasonably high bandwidth needs. On basis of these traffic flows, we investigate the effects that token bucket parameters have on the bandwidth demand and discuss tradeoff possibilities in order to reduce it.

Capacity management and routing policies for voice over IP traffic

IEEE Network, 2000

here has been substantial interest in the recent past in migrating telephony service away from circuit-switched networks onto an IP-based packet-switched network infrastructure. One critical issue to be resolved in achieving this migration is how to support the quality of service (QoS) requirements of a high-quality telephony service over an IP network. A telephony service requires stringent bounds on end-to-end packet delay, jitter, and loss. Ensuring that these requirements are met requires the use of resource management mechanisms, such as scheduling and admission control, in the network. A telephony service also requires that the probability of blocking offered calls be fairly small (≤1 percent). This requires the use of capacity planning and provisioning mechanisms to ensure that the network has adequate capacity to handle the expected traffic volume, and routing mechanisms to ensure that the offered traffic is routed over the network in a manner that makes the most efficient use of available network capacity.

Improving Quality of Service Through Buffer Optimization in VoIP Network

Voice over Internet Protocol (VoIP) is developed for voice communications system based on voice packets transmitted over IP network with real-time communications of voice across networks using the Internet protocols. Quality of Service (QoS) mechanism is applied to guarantee successful voice packets transmitted over IP network with reduced delay or drop according to assigned priority of voice packets. In this paper, the goal of simulation models is present to investigate the performance of VoIP buffer size for improving quality of service (QoS) through buffer optimization with the simulation results by using OPNET modeler version 14.5. The performance of the proposed algorithm is analyzed and compared the quality of service for VoIP. The final simulated result shows that the VoIP service performance best buffer size 128 Kb over WiMAX network.

Statistical analysis and modeling of Internet VoIP traffic for network engineering

Electronic Journal of Statistics, 2010

In this paper we show empirical data from Internet traffic measurements. Collected measurements are analyzed for different protocols, such as TCP and UDP. We perform statistical analysis through the correlation coefficients, covariance, and self-similarity degree i.e. Hurst parameter. Our experimental studies captured traffic with Hurst parameter around 0.7-0.75, which is near half way between values of 0.5 (it is not a self-similar) and 1 (strong self-similar properties). We use Maximum Likelihood approach to fit the obtained time series to existing distributions, such as Pareto and exponential distribution, where the first one is a self-similar process and the second is not. The analysis pointed out that Internet traffic with such values for the Hurst parameter could be modeled with similar accuracy using either distribution, Pareto and exponential.

Optimization Model for Achieving Efficient VoIP Networks

In this paper, we propose an optimization method based-on E-Model for designing an efficient VoIP network. Our method is based on selection of some VoIP network parameters such as voice coder, communication protocol, packet loss level, network utilization and resource allocation. We draw analytic approach for achieving rating value (R) that represent level of quality of service. In this approach, we make some simplification and focus on delay and packet loss calculation to find R-value.