The Contributory Effect of Latency on the Quality of Voice Transmitted over the Internet (original) (raw)
Related papers
2017
Identifying those causes and parameters that affect the Quality of Service (QoS) of Voice-over-Internet Protocol (VoIP) through heterogeneous networks such as WiFi, WiMAX and between them are carried out using the OPNET simulation tool. Optimization of the network for both intra-and inter-system traffic to mitigate the deterioration of the QoS are discussed. The average value of the jitter of the VoIP traffic traversing through the WiFi-WiMAX network was observed to be higher than that of utilizing WiFi alone at some points in time. It is routinely surmised to be less than that of transiting across the WiFi network only and obviously higher than passing through the increased bandwidth network of WiMAX. Moreover, both the values of the packet end-to-end delay and the Mean Opinion Score (MOS) were considerably higher than expected. The consequences of this optimization, leading to a solution, which can ameliorate the QoS over these networks are analyzed and offered as the conclusion of this ongoing research. Keywords—Voice over Internet Protocol (VoIP); Quality of Service (QoS); Mean Opinion Score (MOS); simulation
Analysis and Modeling of QoS Parameters in VoIP Traffic
Trends and Issues
Our studies have revealed that VoIP jitter can be modeled by self-similar processes, and through a decomposition based on Haar wavelet it is shown a possible reason of the presence of long range dependence (LRD) in VoIP jitter. On the other hand, we used a description of VoIP packet loss based on microscopic and macroscopic packet loss behaviors, where these behaviors can be modeled by 2-state and 4-state Markov chains, respectively. Besides, the distributions of the number of consecutive received and lost packets (namely gap and burst, respectively) are modeled from the transition probabilities of 2-state and 4-state Markov chains. Based on the above mentioned description, we presented a methodology for simulating packet loss and proposed a new model that allows to relate the Hurst parameter (H) with the packet loss rate (PLR). These models can be used by other researchers as input to problems related to the design of VoIP applications, performance evaluation of IP networks, and th...
Performance Evaluation of VoIP Analysis and Simulation
Journal of Engineering Research and Reports
The use of technology has impacted on communication in so many ways. The advent of voice over internet protocols (VoIP) has made the world a global village where one can reach out to any part of the universe. But a challenge exists as to how to make communication and data transmission faster, the volume of traffic, bandwidth and latency in networks, that has to be transmitted between the sender and the receiver. The overall customer experience can be improved by the use of technology, which also makes it simpler to collect client information. Data packets are addressed and routed by the Internet Protocol (IP). This research aimed at deploying jitter, throughput, network traffic delay and bandwidth (JiTTraB) as a performance metrics to measure voice over internet protocols (VoIP) to measure the Quality of Service (QoS) of networks. This method prioritizes network traffic going through a router and providing acceptable service to most users in a quest to address VoIP concerns. In comp...
A Solution for Evaluating the QoS of Voice over IP
Solutions, Methods, and Applications
In this chapter, we present a solution for evaluating the Quality of Services (QoS) of Voice over Internet Protocol (VoIP). First, we present an introduction to the main concepts and mathematical background relating to QoS and Internet Protocol (IP) traffic nature, which subsequently are used in the measurements, analysis, and modeling of VoIP traffic. Secondly, we analyze network measurements and the result of the simulation in order to characterize the VoIP traffic nature. As results of this analysis, it is shown that VoIP jitter can be modeled by alpha-stable distributions and self-similar processes, with either Short or Long Range Dependence (i.e., SRD or LRD). Thirdly, we investigate the packet loss effects on the VoIP jitter, and present a methodology for simulating packet loss on VoIP jitter. Finally, we found an empirical relationship between the Hurst parameter (H) and the Packet Loss Rate (PLR); this relationship is based on voice traffic measurements and can be modeled by means of a power-law function with three fitted parameters.
WSEAS TRANSACTIONS on …, 2009
In this work, the perceived quality of VoIP communications is studied. The distributions of the number of consecutive received and lost packets, respectively named gap and burst, of a VoIP communication are modeled with discrete two-state and four-state Markov chains. Algorithms for estimating the transition probabilities between states and from these, the packet loss rate and the respective gap and burst length distributions, are described. Through a study of monitored VoIP calls, it is shown that these models can adequately represent the geometric-type decay of these distributions and that although two-state model performs well for homogeneous losses, for non-homogeneous losses the four-state model fits better. An analysis of the performance of a packet-level FEC scheme, based on-packet redundancy, is presented. The perceived packet loss rate that results of applying this correction scheme is quantified. For the studied measurements, 1-packet redundancy is sufficient to decrease the perceived loss rate below 1%. Also, the impairments of the perceived quality of voice after the FEC technique and a de-jitter buffer is quantified. The resulting equations can be used to optimize the adjust parameters of the VoIP call, e.g., level of redundancy, type of codec used and de-jitter buffer size. The proposed methodology can be extended if other types of improvements are included.
Performance Evaluation of the QoS for VoIP using Different CODECS
Voice over Internet Protocol (VoIP) service is growing very fast and supported by many applications. Its interactive nature makes it very attractive service. VoIP requires a precise level of quality to be utilized. Quality of Service (QoS) is determined by factors like jitter, traffic sent, traffic received and end-to-end delay. In this paper, we study the performance of different scheduling schemes, like: FIFO, PQ, and WFQ for different codec formats. The implementation of the schemes was carried out using OPNET. VoIP service is deployed using the internet implementing the Resource Reservation Protocol (RSVP). The paper discusses the results through a number of figures for the jitter, end-to-end delay and the traffic sent and received. Figures show the different scheduling schemes PQ, WFQ and FIFO with different codec formats, G.711, G.729A and G.723.15 codec formats. I. INTRODUCTION Nowadays, very huge amounts of voice traffic are transferred between millions of people across the world using different social media applications. Using VoIP over the Internet connection, we should be aware about the quality of the VoIP service. VoIP service requires a precise level of quality to be utilized. The end user perception of the quality is determined by subjective testing as a function of the network impairments such as delay, jitter, packet loss, and blocking probability. The amount of impairment introduced by a packet network depends on the particular QoS mechanism implemented [1] Quality of Service (QoS) is determined by factors like the delay the packet delay variation (jitter), and the data loss rate [2]. The greatest technical problem in supporting multimedia services over IP is that real-time traffic must reach its destination within a preset time interval (delay) and with some tolerance of the delay variation (jitter). This is difficult because the original UDP/IP operates on a best-effort basis and permits dropping of packets on the way to a destination [3]. The simulation model was done using OPNET Modeler [4] [5]. OPNET has gained considerable popularity in academia as it is being offered free of charge to academic institutions. That has given OPNET an edge over DES NS2 in both market place and academia [6]. In this paper, we studied the performance of the most popular scheduling schemes, like: First-In First-Out (FIFO), priority Queuing (PQ), and Weighted Fair Queuing (WFQ). A comparison is carried out between different codecs (G.711, G.729A and G.723.15) which are the most appropriate to improve QoS for VoIP. The rest of the paper is organized as follows. Section II presents a typical WAN network topology that uses RSVP protocol to be used as a case study for deploying VoIP service. Section III describes the VoIP service and its parameters. Section IV presents the OPNET-based simulation approach for deploying VoIP service. Section V describes the results and analysis of the simulation study. Then section VI conclusion.
Improving Quality of Server (QoS) in Voice over Internet Protocol V6 by Using Queue Technique
Voice over Internet Protocol (VoIP) is develop for voice communication system that based from packet transmitted over IP network, with real-time communications for voice across networks by 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. The objective of this research is to proposed a new approach to create Ethernet network scenario with Internet Protocol version 6 (IPv6) for VoIP and compare quality for improving quality of service (QoS) with the simulation results by using OPNET. This paper demonstrates the simulation framework based on OPNET Modeler version 14.5 is used to Improving Quality of Service (QoS) in VoIP by Using Queue Technique. In VoIP process, the performance analysis of difference queuing duplicates namely FIFO, WFQ and PQ are implemented within the OPNET simulate IPv6 along with the parameters like delay, jitter, and packets loss. The performance of the proposed algorithms is analyzed and compared the quality of service for VoIP in IPv6. The final simulate result shows that the voice traffic base on the priority and weighted-far queues were improved the quality of service for VoIP in IPv6.
Study on the QoE for VoIP Networks
Journal of Networks, 2014
In recent years, people are recognizing that the quality in VoIP application should be evaluated according to the QoE (Quality of Experience). The main goal of this paper is to analyze the different factors on the impact of voice quality for VoIP networks. Our contributions are thus threefold: First, we establish a new VoIP simulation platform. The network simulation software is WANem, the voice communication protocol is implemented by Open Phone. This simulation system is more 'real' than other researcher's system. Secondly, we analyze the factors that affect the voice quality of VoIP networks. Thirdly, we use the VoIP networks simulation platform to test the network performance impact on the quality of voice service. Through the experiment result, we can conclude that in order to get the better voice quality, we use the iLBC codec when the VoIP network is exist packet loss. And use the AMR 12.2 kb/s when the VoIP network is exist time delay.
VOIP PERFORMANCE MEASUREMENT USING QoS PARAMETERS
This paper presents an analysis of the performance of VoIP. In our study, three different VoIP communication aspects have been considered. These comprise the call signaling protocols, networking environments in which the VoIP communications take place, and the Virtual Private Network (VPN) protocols for securing the data transmissions. The performance evaluation involves the identification of the QoS parameters, which would be relevance to the VoIP communications. These parameters have then been measured on the RTP packets transmission in VoIP communications. The results gathered from the analysis shows that these communication aspects have a significant impact on the QoS in the VoIP communications and the impact varies according to the parameters and the communication aspects selected for the analysis.
Reference model for VoIP Quality
Jitter buffers in IP phones have various mechanisms for jitter elimination. To measure how effective these implementations are, we need to compare quality mea-sured using IP phone to default quality for each jitter that could occur in real IP network. Reference model specifying the default quality for each jitter does not exist so far, therefore we present this idea in this paper, where we present the methodology that will be used in the future. In this paper we discuss only the worst possible scenario, where the input flow to the jitter buffer is by modelled by Poisson distribution.