Improving Quality of Server (QoS) in Voice over Internet Protocol V6 by Using Queue Technique (original) (raw)
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The Affects of Different Queuing Algorithms within the Router on QoS VoIP application Using OPNET
In this work, simulation tool "OPNET Modeler version 14.0" is used to implement the proposed network (VoIP Network). The proposed network is a private network for a company that has two locations located at two different countries around the world in order to simulate the communications within the same location as a local and the communications between two locations as a long distance and analyze VoIP QoS through measuring the major factors that affect the QoS for VoIP according to international telecommunication union (ITU) standards such as: delay, jitter and packet loss.
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
Enhanced performance of IPv6 packet transmission over VoIP network
2009
Voice over IP (VoIP) supports us to make free of charge or very cheap soft-phone and hard-phone calls locally and globally. VoIP is moving on Wireless Local Area Networks (WLANs) based on IEEE 802.11 standards. For audio speech quality in packet switch applications, the main concerns are end to end delay and packet loss. This paper presents a VoIP-Telephony deployment model using Opnet Simulator. We have configured an experimental setup and analyzed various methods to improve voice quality. Experimental results using IEEE 802.11a/b were analyzed for voice traffic. Experimental results with varying packet loss using Jperf and distributed coordination function (DCF-Voice) presented.
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.
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.
VoIP Performance Analysis over IPv4 and IPv6
International Journal of Computer Network and Information Security, 2014
The advance of technology often requires the emergence of complementary technologies, of which the transition from IPv4 to IPv6 presents a significant example. The move of protocol has focussed attention on the level of performance for associated technologies. Among the many Internet applications, in contemporary digital communications, VoIP stands apart in importance. This paper presents a performance analysis of VoIP using IPv4 and IPv6. Using OPNET to simulate the protocols and to investigate areas of performance weakness.
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.
Cross-Layer Integration Approach for Improving QoS for IPv6 Based VOIP
Voice over IP (VOIP) is today one of the most innovative IP based Communication Technologies in the Telecommunications industry. This has made it to enjoy a high degree of success in its application in small, medium and large scale enterprises, primarily to save cost as well as leve raging on its enhance functionalities such as mobility and scalability. Despite all its successes, VOIP still faces challenges with Quality of Service (QoS) degradation. This paper proposes a cross-layer model to effectively manage interactions in the data, network and transport layers guided by trade-off between three performance metrics that affect QoS of VOIP for an improved QoS for Voice over IPv6 (VOIPv6). The parameters taken into consideration in this proposed model are: packet loss, delay and throughput observe by the end-user.
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.
Influences of Classical and Hybrid Queuing Mechanisms on VoIP's QoS Properties
VoIP Technologies, 2011
Nowadays we can find many TCP/IP based network applications, such as: WWW, e-mail, video-conferencing, VoIP, remote accesses, telnet, p2p file sharing, etc. All mentioned applications became popular because of fast-spreading broadband internet technologies, like xDSL, DOCSIS, FTTH, etc. Some of the applications, such as VoIP (Voice over Internet Protocol) and video-conferencing, are more time-sensitive in delivery of network traffic than others, and need to be treated specially. This special treatment of the time-sensitive applications is one of the main topics of this chapter. It includes methodologies for providing a proper quality of service (QoS) for VoIP traffic within networks. Normally, their efficiency is intensively tested with simulations before implementation. In the last few years, the use of simulation tools in R&D of communication technologies has rapidly risen, mostly because of higher network complexity. The internet is expanding on a daily basis, and the number of network infrastructure components is rapidly increasing. Routers are most commonly used to interconnect different networks. One of their tasks is to keep the proper quality of service level. The leading network equipment manufacturers, such as Cisco Systems, provide on their routers mechanisms for reliable transfer of time-sensitive applications from one network segment to another. In case of VoIP the requirement is to deliver packets in less than 150ms. This limit is set to a level where a human ear cannot recognize variations in voice quality. This is one of the main reasons why leading network equipment manufacturers implement the QoS functionality into their solutions. QoS is a very complex and comprehensive system which belongs to the area of priority congestions management. It is implemented by using different queuing mechanisms, which take care of arranging traffic into waiting queues. Time-sensitive traffic should have maximum possible priority provided. However, if a proper queuing mechanism (FIFO, CQ, WFQ, etc.) is not used, the priority loses its initial meaning. It is also a well-known fact that all elements with memory capability involve additional delays during data transfer from one network segment to another, so a proper queuing mechanism and a proper buffer length should be used, or the VoIP quality will deteriorate. If we take a look at the router, as a basic element of network equipment, we can realise that we are dealing with application priorities on the lowest level. Such level is presented by waiting queues and queuing mechanisms, related with the input traffic connection interface.