QoS analysis for WebRTC videoconference on bandwidth-limited network

Y. Bandung, L. B. Subekti, Dion Tanjung, C. Chrysostomou
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引用次数: 12

Abstract

This paper presents a quality of service analysis conducted for real-time multimedia communication services particularly over limited capacity network. This research aims to examine quality of videoconference service based on WebRTC technology. We investigate several implementation challenges of videoconferencing in bandwidth limited network by conducting a thorough experimental study. We analyze videoconference quality from the implementation of WebRTC based system for supporting distance learning between two schools in Cianjur, West Java, Indonesia. Further, we study adaptability feature of WebRTC by performing experiments for several network conditions. Result of the experiment shows that fair quality of video (peak signal-to-noise ratio, PSNR, between 30dB to 33dB) can be reached when bandwidth capacity is 2048kbps. On the other hand, the experiment shows acceptable audio and video stream quality with packet loss rate (PLR) less than 1% and jitter less than 100ms can be reached in network condition with bandwidth capacity higher than 128kbps. Outcome of this research supports understanding and addressing of the challenges of real-time videoconferencing on bandwidth limited network.
有限带宽网络下WebRTC视频会议的QoS分析
本文针对有限容量网络下的实时多媒体通信业务进行了业务质量分析。本研究旨在探讨基于WebRTC技术的视讯会议服务品质。我们通过进行全面的实验研究,探讨了在带宽有限的网络中视频会议的几个实现挑战。我们分析了基于WebRTC系统的视频会议质量,以支持印度尼西亚西爪哇省Cianjur的两所学校之间的远程学习。在此基础上,通过多种网络条件下的实验,研究了WebRTC的自适应特性。实验结果表明,当带宽容量为2048kbps时,可以达到较好的视频质量(峰值信噪比PSNR在30dB ~ 33dB之间)。另一方面,实验表明,在带宽容量高于128kbps的网络条件下,可以达到丢包率(PLR)小于1%、抖动小于100ms的可接受音视频流质量。本研究的结果有助于理解和解决带宽有限的网络上实时视频会议的挑战。
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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