Mobile telecommunications are converging towards all-IP solutions. This is the case of the Long Term Evolution (LTE) technology that, having no circuit-switched bearer to support voice traffic, needs a dedicated VoIP infrastructure, which often relies on the IP Multimedia Subsystem architecture. Most telecom operators implement LTE-A, an advanced version of LTE often marketed as $4{G} +$ , which achieves data rate peaks of 300 Mbps. Yet, although such novel technology boosts the access to advanced multimedia contents and services, telco operators continue to consider the VoIP market as the major revenue for their business. In this work, the authors propose a detailed performance assessment of VoIP traffic by carrying out experimental trials across a real LTE-A environment. The experimental campaign consists of two stages. First, we characterize VoIP calls between fixed and mobile terminals, based on a data-set that includes more than 750,000 data-voice packets. We analyze quality-of-service metrics such as round-trip time (RTT) and jitter, to capture the influence of uncontrolled factors that typically appear in real-world settings. In the second stage, we further consider VoIP flows across a range of codecs, looking at the trade-offs between quality and bandwidth consumption. Moreover, we propose a statistical characterization of jitter and RTT (representing the most critical parameters), identifying the optimal approximating distribution, namely the Generalized Extreme Value (GEV). The estimation of parameters through the Maximum Likelihood criterion, leads us to reveal both the short- and long-tail behaviour for jitter and RTT, respectively.

中文翻译：

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LTE-A 网络上 VoIP 的实验评估和表征

移动电信正在向全 IP 解决方案发展。长期演进 (LTE) 技术就是这种情况，它没有支持语音流量的电路交换承载，需要专用的 VoIP 基础设施，而该基础设施通常依赖于 IP 多媒体子系统架构。大多数电信运营商实施 LTE-A，这是 LTE 的高级版本，通常以 $4{G} +$ 进行销售，可实现 300 Mbps 的数据速率峰值。然而，尽管这种新技术促进了对高级多媒体内容和服务的访问，但电信运营商继续将 VoIP 市场视为其业务的主要收入来源。在这项工作中，作者通过在真实 LTE-A 环境中进行实验性试验，提出了对 VoIP 流量的详细性能评估。实验活动包括两个阶段。第一的，我们根据包含超过 750,000 个数据语音数据包的数据集来表征固定和移动终端之间的 VoIP 呼叫。我们分析服务质量指标，例如往返时间 (RTT) 和抖动，以捕捉通常出现在现实环境中的不受控制的因素的影响。在第二阶段，我们进一步考虑跨一系列编解码器的 VoIP 流，查看质量和带宽消耗之间的权衡。此外，我们提出了抖动和 RTT 的统计特征（代表最关键的参数），确定最佳近似分布，即广义极值 (GEV)。通过最大似然准则估计参数，使我们分别揭示了抖动和 RTT 的短尾和长尾行为。