VoIP over WiMAX from QoS Perspectives

Transcription

1 VoIP over WiMAX from QoS Perspectives Muaz Omer Mahmoud Mudawi 1 and Dr. Amin Babiker A/Nabi Mustafa 2 1,2 Deptt. of Communication Engineering, Faculty of Engineering, Alneelain University, Khartoum, Sudan 1 muazomer2000@gmail.com and 2 amin31766@gmail.com Publishing Date: August 28, 2016 Abstract The precious goal for all networks users among the world is to get a network with high data transmutation rate, low cost and high speed. In our paper we investigate a new technology VoIP over WiMAX an OPNET simulator has been used to evaluate the network performance which is depends on numerous variables called performance parameters like jitter, throughput, load, MOS and delay. The investigation was focused on those parameters because it affects the voice during the VoIP call. Three scenarios has been designed and various values for the previous parameters were observed the number of workstations has been tested. Ultimately we find that the VoIP over WiMAX process effected strongly by those factors and its negative impact has been eliminated. Keywords: WiMAX, VoIP CODECs, OPNET. Introduction WiMax System WiMAX was quoted from worldwide interoperability for micro wave access it was introduced by IEEE in 1999 as a part of wireless (MAN) wireless metropolitan area network and it was known commonly by the standard IEEE and this family contain two types of WiMAX, the first IEEE d standard which called fixed WiMAX and the second IEEE e standard or mobile WiMAX. The idea of WiMAX technology wasn t far away from WiFi technology. The construction contain the WiMAX tower, subscriber device or SimCard and the power supply devices, the working concept depends on the uplink and downlink. When the user send signal from his mobile device the base station will receive it and it will broadcast a wireless signal via a unique path or a channel this is uplink. When the base station connect another subscriber or transmit the same one this is downlink. The band width of WiMAX (2-11) GHz and the speed up to 70 mbps even in case of we have many subscribers using 162 the net work it will remain constant in addition of low cost, reliability, privacy and utilization [1]. History of VoIP VoIP is abbreviated as voice over internet protocol it s one of the most famous WiMAX applications also known IP Telephony it embodies transmitting the voice via internet protocol [1]. The analog signals must be converted into digital signals and divided into packets and suitable protocols used to compress and encode the digitized voice then it will be mixed into IP using the real time protocol (RTP) in the other side we have a receiver where the VoIP packets analyzed to extract the digitized voice and convert it into analog signal and transmitted via telephone lines [2]. VoIP Requirement The play out buffer, CODECs and packetizer. - The CODECs:- It is the most important part of VoIP utilized to compression and decompression or sometimes coder and decoder it has various types include G.711, G.723, G.726, G.729 and G The packetizer :- The function of packetizer keep a constant bit rate for the packets of digitized voice. - The PLAOUT BUFFER :- It must be exist in the side of receiver to decrease the effect of jitter but it s not enough. The Performance Metric Parameters - Jitter: So what about Jitter as mentioned before the packets of voice travels between source to destination but the

2 IP network can t guarantee the delivery time for every packet so this variation will produce distortion in voice which has a negative effect toward VoIP [3]. - Throughput: Generally it represent the amount of data processed by time unit in WiMAX network is defined in terms of a number of packets delivered to the network peripherals. It measured by bit/sec and packet/sec or packet/time slot. The value of throughput high that mean good network performance [4]. - Delay: The delay express the time period or how long the packet take when it travel among the network from the source to destination or from end point to another one it must measure in fraction of second it also affected by the distance between the pair of communication and service class [5]. - Load: It a term that express the network traffic or amount of packets cross the network in particular time. Network traffic which provide load has been considered as a mean factor in simulation and both real live because it indicate network collision [5]. - MOS: MOS is coated from mean opinion score it s a criteria to determine if the quality of voice is acceptable or not used in VoIP service class and IP telephone network characterized by ITU to excellent, good, faire, bad and poor [6]. Simulation Setup In this project the version of OPNET simulator 14.5 has been used. The WiMAX module which imbedded with the simulator was used to design the WiMAX network and determined transmutation of VoIP through it. Our design divided into three scenarios. The first one exclusive on using 150 workstation. The second one lies on increasing the number of workstations to be 200 workstations and the last one contain 250 workstations to obtain its impact in the value of jitter, throughput, load, MOS and delay. The first step in design was insert the VoIP application into the workstations by setup the profile configuration and application configuration. IP router has been used to connect the IP cloud (internet) and base station, point to point (PPP DS3) has been used as a connection between the internet and IP router. And the connection between this base station and IP router was Ethernet (100 base T). Figure 1: WiMAX Network 150 Devices 163

3 Figure 2: WiMAX Network 200 Devices Figure 3: WiMAX Network 250 Devices 164

4 Simulation Result In this part the results were taken to show the three parameters of the network in each scenario. The performance of VoIP is analyzed for the three scenarios and the results have been shown. Figure 4: The Average in Voice Jitter We noticed that the values of jitter was fluctuated in case of 150 workstations appears in a green colure but get stable after 1 m and get lowest value and the case of 200 workstations appears in blue colure get lower value and the case of 250 workstations get the low value of jitter appears in red colure so the jitter effect increase by increasing the number of workstation. Figure 5: The Average in WiMAX Throughput 165

5 As it shown above in the result of WiMAX throughput increase by decreasing the number of workstations so we found the case of 150 workstation in the top after that the case of 200 workstation and then the case of 250 get the lowest value of throughput when the throughput rate is high that means the performance is good for the network. Figure 6: The Average in WiMAX Delay The average delay haven t a strong effect in WIMAX networks but as it shown in the results above the red curve which represent the case of 250 device get high delay rate and the case of 200 workstation appear in a blue curve and finally the red curve represent the case of 150 workstation. Figure 7: The Average in WiMAX Load 166

6 It shown that in this section the value of WiMAX load increase by increasing the number of workstations so the scenario which contain 250 workstation is get the highest value and the scenario which contain 150 workstation is the best one. Figure 8: The Average in WiMAX MOS Increasing the value of MOS indicate to good performance for the network and it directly affected by the number of workstation as it clearly shown in the result above, so that the scenario which contain 150 workstation by blue curve represent the best one and vice versa for the green curve. Conclusion In this paper we have been concluded that WIMAX network is the most suitable network for the service class like VoIP. The performance of the network depends on various metric parameters but in this paper the jitter, throughput, load, MOS and delay has been considered as a main parameters because it affect the voice strongly so that three scenarios has been simulated. Our simulation results determined that the VoIP preformed perfect under 150 workstation case compared to 200 and 250 also it shown that bad effect value decreased by decreasing the number of work stations when jitter and throughput were tested and a little bit effected by delay. References [2] N. Bulusu, M. Habib, "Improving QoS of VoIP over WLAN (IQ-VW)", December [3] W.M. Jon, C. Lin, X. Yang and S. Xuemin "VoIP over WLAN: Voice capacity, admission control, QoS, and MAC" May [4] Z. Niu, J. Liu, "An adaptive receiver buffer adjust algorithm for VoIP applications considering voice characters," August [5] L. Murphy, M. Narbutt,"A New VoIP Adaptive Playout Algorithm", [6] M. Liu, J. J. Rodriguez and K. M. McNeill, "An Adaptive Jitter Buffer PlayOut Scheme to Improve VoIP Quality in Wireless Networks", About Authors: Muaz Omer Mahmoud Mudawi: has a B.Sc. in Computer Engineering from Future University (Computer Man College) in Working in Ministry of education (Khartoum State) in Examination Network from 2007 till now. Working in Saudi Arabia in teaching field (maintenance, programming, designed) from 2002 to [1] Ravneet Kaur, Preetinder Singh, "VOIP Over WiMAX: Comprehensive Review," 2014,. 167

7 Amin Babiker A/Nabi Mustafa: obtained his B.Sc. & M.Sc. from University of Khartoum in 1990 & 2001 respectively. He obtained his Ph.D from Alneelain University in He was the head of Computer Engineering Dept. from 2001 to Then he become the vice-dean. He has been the dean, faculty of Engineering Alneelain University since 2009 upto March 2015 and then the Secretary of Academic Affairs from April 2015 upto December His research areas include QoS in Communication Systems, traffic Engineering, Service Costing Disciplines and Networking Assonciate Prof. Dr. Amin is a Consultant Engineer. He is a member of the Sudan Engineering Council. He is also a member of the Executive Committee of the Federation of Sudanese Engineers. Dr. Amin supervised or supervising more than 80 Ph.D or M. Sc. students. He published over 150 papers in International Scientific Journals. 168