PERFORMANCE EVALUATION OF VOICE OVER WIMAX

• Main Authors: Hui-Wen Su, 蘇慧文
• Other Authors: Teng-Pin Lin
• Format: Others
• Language: zh-TW
• Published: 2010
• Online Access: http://ndltd.ncl.edu.tw/handle/86525823614663492644

碩士 === 大同大學 === 電機工程學系(所) === 98 === Nowadays, a new standard IEEE802.16 has become the most popular wireless broadband access technology which has the merit of high bandwidth and mobility. Since the bandwidth of BS is limited, it deserves to do analysis to sacrifice the request of numerous users .Based on the goal of making the simulation corresponding with the actual network operation situation, this research executes the simulation on the platform of NS2 to simulate WiMAX network which is very close to the actual PMP and discuss how to achieve the usage rate of the maximum system capacity for specific web application service on the condition of multiple users sharing the bandwidth. The network system is divided into two parts; one is for simulating Background APP under the circumstances that another user in the same BS occupying the partial bandwidth and the other one is to simulate the level of the system capacity we intent to test supporting Tracing APP (user) for the use of special application service. Five different kinds of queue management are used to do simulation respectively, they are Drop Tail、RED、DRR、FQ and SFQ to analyze and compare the competition situation of the network resource and the efficiency of real-time voice transmission with the increasing of users. In the application of VoIP, two kinds of Codes G.711 and G729 which their data rate are 64kbps and 8kbps respectively are compared. Video and VOIP are combined to make the simulation more closer to the actual network situation. This research establishes a CBR transmission with 160 Kbps data rate to serve as the background for simulation. Parameters in this simulation are as follows: 1ms transmission delay time, 1Mbps bandwidth, OFDM_QPSK_3_4 modulation and DSDV Routing protocol, other parameters that are not defined are used with default values which are put in the ns-default.tcl file. The standard of assessing the communication quality sets the delay time as 150ms and the loss rate as 3%. The result of simulation from this research shows that Codec G.711 can achieve the maximum system capacity with around 21 users getting online simultaneously. Since the amount of packet Codec G.729 transfer each time are smaller, comparing to Codec G.711, it can achieve the maximum system capacity with around 45 users getting online simultaneously.. Finally, the combination of Video and VOIP can achieve the maximum system capacity with 7 users getting online.