首页> 外文OA文献 >Performance analysis of 8kbps voice codec (G.729, G.711 ALAW, G.711 ULAW) for voip over wireless local area network with respective signal-to-noise ratioudNOISE RATIO
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Performance analysis of 8kbps voice codec (G.729, G.711 ALAW, G.711 ULAW) for voip over wireless local area network with respective signal-to-noise ratioudNOISE RATIO

机译:用于具有相应信噪比 ud的无线局域网上的voip的8kbps语音编解码器(G.729,G.711 ALAW,G.711 ULAW)的性能分析噪声比

摘要

In this research, 3 types of speech codec (G.729, G.711 aLaw and G.711 uLaw) in the same sampling rate of 8kbps are put to test in predefined network environment and given respective SNR 10dB, 20dB and 30dB to measure the performance base on R-factor, MOS, packet jitter and packet lost. Speech codec is used to convert the analog voice signals into digital signal. Each speech codec have its own speech quality, minimum bandwidth require etc. There are many manufacturers that have been producing various types of speech codecs in the market. The VoIP users are able to choose the desire codec that will be used or enable in the VoIP call based on the service and hardware that can support the speech codec. But, users will face some difficulty in choosing the best codec to use. All 3 mentioned speech codec will be test base on these criteria; VoIP session over optimum wireless network with 10dB, 20dB and 30dB SNR and VoIP session over wireless network that shared with other traffic with 10dB, 20dB and 30dB SNR. Six testbed will be carry out to complete all the criteria and all of the tests criteria will be carry out on real devices simulation. At the end, the performance measurement such as MOS, r-factor , packet lost and packet jitter will be observe to determine the best speech codec in each scenario. The final results of this research should be able to determine the best speech codec among the four codecs that have been selected and match the suitability with the environments.
机译:在这项研究中,在预定义的网络环境中以相同的8kbps采样率测试了三种类型的语音编解码器(G.729,G.711 aLaw和G.711 uLaw),并分别给出了SNR 10dB,20dB和30dB进行测量性能基于R因子,MOS,数据包抖动和数据包丢失。语音编解码器用于将模拟语音信号转换为数字信号。每个语音编解码器都有自己的语音质量,最小带宽要求等。市场上有许多制造商在生产各种类型的语音编解码器。 VoIP用户能够基于支持语音编解码器的服务和硬件来选择将在VoIP呼叫中使用或启用的所需编解码器。但是,用户在选择最佳编解码器时会遇到一些困难。上述所有3种语音编解码器都将根据这些标准进行测试;在SNR为10dB,20dB和30dB的最佳无线网络上进行VoIP会话,在SNR为10dB,20dB和30dB的无线网络上与其他流量共享VoIP会话。将执行六个测试平台以完成所有标准,并且将在真实设备仿真中执行所有测试标准。最后,将观察诸如MOS,r因子,数据包丢失和数据包抖动之类的性能测量,以确定每种情况下的最佳语音编解码器。这项研究的最终结果应该能够在已选择的四种编解码器中确定最佳的语音编解码器,并使其与环境相匹配。

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    Norani Mohammad Zaki;

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  • 年度 2013
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