VoIP Packet Delay Techniques: A Survey

The continuous development in the field of communication have paved the way for Voice over Internet Protocol (VoIP). VoIP is a group of hardware and software that facilitates people to utilize the Internet as the transmission medium for telephone calls by transmitting voice data in packets using IP instead of using conventional circuit transmissions of the Public Switched Telephone Network (PSTN). At present, VoIP is becoming an important tool for quick communication across the world. There are several Internet telephony applications existing at present. The major disadvantage in VoIP is that the packet delay. In VoIP, the terminology jitter is used to refer the type of packet delay where the delay has a huge setback in the quality of the voice conversation. Several packet delay techniques were proposed in recent years. Some of the important packet delay techniques are discussed in the literature. This survey would definitely help the researchers to carry out their research for providing better communication in VoIP without any delay

The EuQoS system: A solution for QoS routing in heterogeneous networks

EuQoS is the acronym for “end-to-end quality of service support over heterogeneous networks,”which is a European research project aimed at building an entire QoS framework,addressing all the relevant network layers, protocols, and technologies. This framework, which includes the most common access networks (xDSL, UMTS, WiFi, and LAN) is being prototyped and tested in a multidomain scenario throughout Europe, composing what we call the EuQoS system. In this article we present the novel QoS routing mechanisms that are being developed and evaluated in the framework of this project. The preliminary performance results validate the design choices of the EuQoS system, and confirm the potential impact this project is likely to have in the near future.Postprint (published version

Quality of service over ATM networks

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Quality of Service optimisation framework for Next Generation Networks

Within recent years, the concept of Next Generation Networks (NGN) has become widely accepted within the telecommunication area, in parallel with the migration of telecommunication networks from traditional circuit-switched technologies such as ISDN (Integrated Services Digital Network) towards packet-switched NGN. In this context, SIP (Session Initiation Protocol), originally developed for Internet use only, has emerged as the major signalling protocol for multimedia sessions in IP (Internet Protocol) based NGN. One of the traditional limitations of IP when faced with the challenges of real-time communications is the lack of quality support at the network layer. In line with NGN specification work, international standardisation bodies have defined a sophisticated QoS (Quality of Service) architecture for NGN, controlling IP transport resources and conventional IP QoS mechanisms through centralised higher layer network elements via cross-layer signalling. Being able to centrally control QoS conditions for any media session in NGN without the imperative of a cross-layer approach would result in a feasible and less complex NGN architecture. Especially the demand for additional network elements would be decreased, resulting in the reduction of system and operational costs in both, service and transport infrastructure. This thesis proposes a novel framework for QoS optimisation for media sessions in SIP-based NGN without the need for cross-layer signalling. One key contribution of the framework is the approach to identify and logically group media sessions that encounter similar QoS conditions, which is performed by applying pattern recognition and clustering techniques. Based on this novel methodology, the framework provides functions and mechanisms for comprehensive resource-saving QoS estimation, adaptation of QoS conditions, and support of Call Admission Control. The framework can be integrated with any arbitrary SIP-IP-based real-time communication infrastructure, since it does not require access to any particular QoS control or monitoring functionalities provided within the IP transport network. The proposed framework concept has been deployed and validated in a prototypical simulation environment. Simulation results show MOS (Mean Opinion Score) improvement rates between 53 and 66 percent without any active control of transport network resources. Overall, the proposed framework comes as an effective concept for central controlled QoS optimisation in NGN without the need for cross-layer signalling. As such, by either being run stand-alone or combined with conventional QoS control mechanisms, the framework provides a comprehensive basis for both the reduction of complexity and mitigation of issues coming along with QoS provision in NGN

STOCHASTIC MODELING AND TIME-TO-EVENT ANALYSIS OF VOIP TRAFFIC

Voice over IP (VoIP) systems are gaining increased popularity due to the cost effectiveness, ease of management, and enhanced features and capabilities. Both enterprises and carriers are deploying VoIP systems to replace their TDM-based legacy voice networks. However, the lack of engineering models for VoIP systems has been realized by many researchers, especially for large-scale networks. The purpose of traffic engineering is to minimize call blocking probability and maximize resource utilization. The current traffic engineering models are inherited from the legacy PSTN world, and these models fall short from capturing the characteristics of new traffic patterns. The objective of this research is to develop a traffic engineering model for modern VoIP networks. We studied the traffic on a large-scale VoIP network and collected several billions of call information. Our analysis shows that the traditional traffic engineering approach based on the Poisson call arrival process and exponential holding time fails to capture the modern telecommunication systems accurately. We developed a new framework for modeling call arrivals as a non-homogeneous Poisson process, and we further enhanced the model by providing a Gaussian approximation for the cases of heavy traffic condition on large-scale networks. In the second phase of the research, we followed a new time-to-event survival analysis approach to model call holding time as a generalized gamma distribution and we introduced a Call Cease Rate function to model the call durations. The modeling and statistical work of the Call Arrival model and the Call Holding Time model is constructed, verified and validated using hundreds of millions of real call information collected from an operational VoIP carrier network. The traffic data is a mixture of residential, business, and wireless traffic. Therefore, our proposed models can be applied to any modern telecommunication system. We also conducted sensitivity analysis of model parameters and performed statistical tests on the robustness of the models’ assumptions. We implemented the models in a new simulation-based traffic engineering system called VoIP Traffic Engineering Simulator (VSIM). Advanced statistical and stochastic techniques were used in building VSIM system. The core of VSIM is a simulation system that consists of two different simulation engines: the NHPP parametric simulation engine and the non-parametric simulation engine. In addition, VSIM provides several subsystems for traffic data collection, processing, statistical modeling, model parameter estimation, graph generation, and traffic prediction. VSIM is capable of extracting traffic data from a live VoIP network, processing and storing the extracted information, and then feeding it into one of the simulation engines which in turn provides resource optimization and quality of service reports

On local CAC schemes for scalability of high-speed networks

This article has been extended from the original: “On local CAC schemes for scalability of high-speed networks” by the same authors, which was presented on the International Conference on Transparent Optical Networks (ICTON) held in Athens, Greece, on June 22-26, 2008Next generation networks are required to provide bandwidth on-demand for sessions with fine-time granularity. In this sense, centralized CAC (Connection Admission Control) approaches could suffer from scalability problems if the number of requests for connections is excessive. In this paper we investigate local CAC schemes where the admission decisions are performed at the network edges, based on precalculated admission quotas.The authors would like to acknowledge the BONE Network of Excellence, partially funded by the European Union Seventh Framework Programme, the MUSE project, partially funded by the EU Sixth FP, and the RUBENS project as part of the EUREKA CELTIC initiativ