Performance Analysis of Receive-Side Real-Time Congestion Control for WebRTC
Performance Analysis of Receive-Side Real-Time Congestion Control for WebRTC
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WebRTC接收端实时拥塞控制性能分析
DOI：
10.1109/pv.2013.6691454
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发表时间：
2013
期刊：
影响因子：
--
通讯作者：
J. Ott
中科院分区：
文献类型：
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作者：
Varun Singh;A. Lozano;J. Ott
在即将部署的WebRTC系统中，我们推测高质量的视频会议将被广泛采用。它目前被部署在b谷歌Chrome和Firefox网络浏览器上，同时桌面和移动客户端正在开发中。如果没有标准化的信令机制，服务提供者可以启用各种类型的拓扑；从全网格到集中式视频会议以及介于两者之间的一切。在本文中，我们使用实现WebRTC的端点来评估各种拓扑的性能。我们特别评估了当前在这些web浏览器中实现和部署的拥塞控制的性能，即接收端实时拥塞控制（RRTCC）。我们使用传输损害，如不同的吞吐量、损失和延迟，以及不同数量的交叉流量来衡量性能。我们的研究结果表明，RRTCC在单独使用时性能良好，但在与TCP竞争时却缺乏性能。当与自相似的媒体流竞争时，后发流暂时使现有的媒体流挨饿。
In the forthcoming deployments of WebRTC systems, we speculate that high quality video conferencing will see wide adoption. It is currently being deployed on Google Chrome and Firefox web-browsers, meanwhile desktop and mobile clients are under development. Without a standardized signaling mechanism, service providers can enable various types of topologies; ranging from full-mesh to centralized video conferencing and everything in between. In this paper, we evaluate the performance of various topologies using endpoints implementing WebRTC. We specifically evaluate the performance of the congestion control currently implemented and deployed in these web-browser, Receive-side Real-Time Congestion Control (RRTCC). We use transport impairments like varying throughput, loss and delay, and varying amounts of cross-traffic to measure the performance. Our results show that RRTCC is performant when by itself, but starves when competing with TCP. When competing with selfsimilar media streams, the late-arriving flow temporarily starves the existing media flows.