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Analysis of Skype VoIP traffic in UMTS: End-to-end QoS and QoE measurements

机译:UMTS中的Skype VoIP流量分析:端到端QoS和QoE测量

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In the future Internet, multi-network services will follow a new paradigm in which the intelligence of the network control is gradually moved to the edge of the network. This impacts both the objective Quality of Service (QoS) of the end-to-end connection as well as the subjective Quality of Experience (QoE) as perceived by the end user. Skype already offers such a multi-network Voice-over-IP (VoIP) telephony service today. Due to its ease of use and a high sound quality, it becomes increasingly popular in the wired Internet. UMTS operators promise to offer large data rates which should suffice to support VoIP calls in a mobile environment. However, the success of those applications strongly depends on the corresponding QoE. In this work, we analyze the theoretically achievable as well as the actually achieved quality of IP-based voice calls using Skype. This is done performing measurements in both a real UMTS network and a testbed environment. The latter is used to emulate rate control mechanisms and changing system conditions of UMTS networks. The results show in how far Skype over UMTS is able to keep pace with existing mobile telephony systems and how it reacts to different network characteristics. The investigated performance measures comprise the QoE in terms of the MOS value and the QoS in terms of network-based factors like throughput, packet interarrival times, or packet loss.
机译:在未来的Internet中,多网络服务将遵循一种新的范例,其中网络控制的智能逐渐移至网络边缘。这会影响端到端连接的客观服务质量(QoS)以及最终用户感知的主观体验质量(QoE)。 Skype现在已经提供了这种多网络IP语音(VoIP)电话服务。由于其易用性和高音质,它在有线Internet中变得越来越流行。 UMTS运营商承诺提供大数据速率,这足以在移动环境中支持VoIP呼叫。但是,这些应用程序的成功很大程度上取决于相应的QoE。在这项工作中,我们分析了使用Skype进行IP语音通话的理论上可实现以及实际达到的质量。这是在真实的UMTS网络和测试平台环境中进行测量的。后者用于模拟UMTS网络的速率控制机制和不断变化的系统条件。结果表明,基于UMTS的Skype能够与现有的移动电话系统保持同步,以及它如何对不同的网络特性做出反应。研究的性能指标包括基于MOS值的QoE和根据基于网络的因素(如吞吐量,数据包到达时间或数据包丢失)的QoS。

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