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 ( I. INTRODUCTIONNowadays, 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.
Voice over Internet Protocol (VoIP) service is growing very fast. Service providers offer VoIP service along with traditional data services using the same IP infrastructure. That is why today it is one of the most dominant technologies for communication. In this paper, simulative investigations have been done for VoIP service in WiFi campus network.Step by step, increasing the number of calls, investigations have been done in terms of important Quality of Service parameters like jitter, packet end-to-end delay, wireless LAN load and wireless LAN throughput.
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