Adaptive Multi Lane technique for LTE radio access VoIP

Onsy Abdel Alim, S. Shaaban, M. Hamdy
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引用次数: 1

Abstract

This paper describes a novel radio access channel adaptive technique designed for the Long Term Evolution (LTE) VoIP codecs. In previous 3GPP releases, the circuit switched Adaptive Multi Rate (AMR) voice codec has proven robust performance under different radio channel conditions. With the 3GPP Release 8 LTE standards, the circuit switched domain is cancelled, making LTE an all IP system and VoIP the only way to carry voice. Despite that VoIP codecs are well developed for internet applications, these codecs are not designed to adapt according to the radio channel conditions, as in the case of AMR. With the new Adaptive Multi Lane (AML) approach, the system is to stream 1, 2 or 4 lanes of speech frames in parallel according to the radio channel conditions. Each lane has a different fractional speech frame delay increasing the probability of getting out of fading dips. The AML technique has been modeled and simulated over OMNET++ ver4.1 including the INET framework and VoIP tool extension. The VoIP tool has been modified to implement the lane selection algorithm. The resulting speech quality was evaluated using the ITU P.862 PESQ. Simulation results show improvement in the average PESQ MOS readings exceeding 44% compared to the normal single path model. We conclude that the AML is a useful technique for improving VoIP codecs performance in cellular systems.
LTE无线接入VoIP的自适应多车道技术
本文介绍了一种针对长期演进(LTE) VoIP编解码器设计的新型无线接入信道自适应技术。在之前的3GPP版本中,电路交换自适应多速率(AMR)语音编解码器在不同的无线电信道条件下已经证明了强大的性能。随着3GPP Release 8 LTE标准的发布,电路交换域被取消,LTE成为全IP系统,VoIP成为唯一的语音传输方式。尽管VoIP编解码器已经为互联网应用开发得很好,但这些编解码器并没有像AMR那样根据无线电信道条件进行调整。采用新的自适应多通道(AML)方法,系统将根据无线电信道条件并行传输1、2或4通道的语音帧。每个信道都有不同分数的语音帧延迟,增加了摆脱衰落下降的概率。在包含INET框架和VoIP工具扩展的omnet++ ver4.1上对AML技术进行了建模和仿真。已修改VoIP工具以实现选道算法。使用国际电联P.862 PESQ对产生的语音质量进行了评估。仿真结果表明,与普通单路径模型相比,平均PESQ MOS读数提高了44%以上。我们得出结论,AML是一种有用的技术,可以提高VoIP编解码器在蜂窝系统中的性能。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
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