Performance analysis on voip for kolej kediaman 2 & 3, Universiti Malaysia Pahang (original) (raw)
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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...
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.
International Journal of Intelligent Systems Design and Computing, 2018
Circuit switched network or packet switched networks are used for both visual and vocal communication. Public switched telephone network (PSTN) is not an affordable option when comparing with existing packet switched network. Voice over internet protocol (VoIP) has become a preferable alternative due to its reduced cost and flexibility compared to voice over LTE (VoLTE) where carriers are still building out their 4G networks. However, despite its reduced cost, it has to face so many challenges which affect its successful deployment. The reason is that the quality of VoIP is mainly affected by jitter, delay, packet loss and various other parameters. This research was carried out to evaluate the quality of voice in VoIP experimentally, under the effect of interference and during conference calls. The simulations were carried out using Riverbed Modeler academic edition 17.5. The results of the analysis and the performance evaluation are presented in this paper. This work can guide researchers and designers to design a network for VoIP services and its deployment based on their requirements.
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.
Enhanced performance of packet transmission using system model over VoIP network
2010
Internet Protocol (VoIP) network is challenging. Packet loss, delay, jitter, etc. highly degrade the performance of the network. Hence more robust techniques are required to provide better level of service quality. One crucial parameter to measure service quality is packet loss. The objective of this paper is to analyze the impact of packet loss and identify proper codec techniques for VoIP traffic. We used a packet analyzer tool (wireshark) to analyze the voice traffic. We found that wireshark could serve as an effective tool for packet loss analysis. It helped us in identifying the proper codec technique that should be used for an enhanced performance in a VoIP network.
In this Paper, the objective of simulation models is presented to investigate the performance of VoIP codecs over WiMAX and FDDI networks that specially design for Aden University. To assure if the University IP network is prepared and adequate for this new type of traffic before adding any new components, Aden University IP network will be simulated by using OPNET simulation software then the new VoIP service will be added to the University networks. Different parameters that represent the QoS like end to end delay, jitter, traffic sends and traffic received, MOS are calculated and analyzed in both network scenarios.
Sweden OPNET simulation of voice over MPLS with considering Traffic Engineering ABSTRACT Multiprotocol Label Switching (MPLS) is an emerging technology which ensures the reliable delivery of the Internet services with high transmission speed and lower delays. The key feature of MPLS is its Traffic Engineering (TE) which is used for effectively managing the networks for efficient utilization of network resources. Due to lower network delay, efficient forwarding mechanism, scalability and predictable performance of the services provided by MPLS technology makes it more suitable for implementing real-time applications such as Voice and video. In this thesis performance of Voice over Internet Protocol (VoIP) application is compared in MPLS network and conventional Internet Protocol (IP) network. OPNET modeler 14.5 is used to simulate the both networks and the comparison is made based on the metrics such as Voice jitter, Voice packet end-to-end delay, voice delay variation, voice packet send and received. The simulation results are analyzed and it shows that MPLS based solution provides better performance in implementing the VoIP application.
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.
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
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.