QoE evaluation of WebRTC-based Mobile Multiparty Video Calls in Light of Different Video Codec Settings

Dunja Vučić, Lea Skorin-Kapov
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引用次数: 7

Abstract

Technological advances in mobile networks and end user devices are paving the way for widespread use of interactive multiparty audiovisual communication services. The emergence of the WebRTC (Web Real-Time Communication) open framework has provided an opportunity to redefine the video conferencing communication landscape. However, in the mobile context, challenges remain in optimizing video encoding parameters due to limited device processing power and dynamic wireless network conditions. In this paper, we investigate how different video encoding parameters in terms of encoding bit rate, resolution, and frame rate, influence Quality of Experience (QoE). While video parameters are manually manipulated, we observe also dynamic video adaptation invoked as a result of the Google Congestion Control (GCC) algorithm. We conduct an experimental study involving twenty-seven users participating in three-party audiovisual WebRTC calls on symmetric mobile devices with a given conversation task to simulate natural conversation. Participants reported subjective ratings for test scenarios with different combinations of video encoding parameters, for which we collect real-time session data using the Chrome WebRTC-internals tool.
基于webrtc的移动多方视频通话在不同视频编解码器设置下的QoE评价
移动网络和终端用户设备的技术进步为广泛使用交互式多方视听通信服务铺平了道路。Web实时通信(webbrtc)开放框架的出现为重新定义视频会议通信领域提供了机会。然而,在移动环境下,由于有限的设备处理能力和动态无线网络条件,优化视频编码参数仍然存在挑战。在本文中,我们研究了不同的视频编码参数在编码比特率,分辨率和帧率方面如何影响体验质量(QoE)。虽然视频参数是手动操作的,但我们也观察到动态视频适应是由于Google拥塞控制(GCC)算法而调用的。我们进行了一项实验研究,涉及27名用户在对称移动设备上参与三方视听WebRTC呼叫,并具有给定的对话任务来模拟自然对话。参与者报告了对不同视频编码参数组合的测试场景的主观评分,为此,我们使用Chrome webtc内部工具收集实时会话数据。
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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