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Reducing End-to-End Delays in WebRTC using the FSE-NG Algorithm for SCReAM Congestion Control

机译:使用FSE-NG算法减少WebRTC中的端到端延迟,用于尖叫拥塞控制

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The 2020 Corona pandemic has shown that on-line real-time multimedia communication is of vital importance when regular face-to-face meetings are not possible. One popular choice for conducting these meetings is the open standard WebRTC which is implemented in every major web browser. Even though this technology has found widespread use, there are still open issues with how different congestion control (CC) algorithms of Media- and DataChannels interact. In 2018 we have shown that the issue of self-inflicted queuing delay can be mitigated by introducing a CC coupling mechanism called FSE-NG. Originally, this solution was only capable of linking DataChannel flows controlled by TCP-style CCs and MediaChannels controlled by NADA CC. Standardization has progressed and along with NADA, IETF has also standardized the RTP CC SCReAM. This work extends the FSE-NG algorithm to also incorporate flows controlled by the latter algorithm. By means of simulation, we show that our approach is capable of drastically reducing end-to-end delays while also increasing RTP throughput and thus enabling WebRTC communication in scenarios where it has not been applicable before.
机译:2020电晕大流行表明,当不可能面对面的面对面会议时,在线实时多媒体通信是至关重要的。进行这些会议的一个流行的选择是开放标准WebRTC,其在每个主要的Web浏览器中实现。尽管这项技术已经发现广泛使用,但仍然存在不同的拥塞控制(CC)媒体和DataChannels交互的不同拥塞控制(CC)算法。 2018年,我们表明,通过引入名为FSE-NG的CC耦合机制,可以减轻自我造成的排队延迟问题。最初,此解决方案仅能够将由TCP式CCS和MedcomChanns控制的数据流量链接到由Nada CC控制。标准化已经进展并与Nada一起,IETF还标准化了RTP CC尖叫声。这项工作扩展了FSE-NG算法还将通过后一种算法控制的流融合。通过模拟,我们表明我们的方法能够大大减少端到端延迟,同时还增加了RTP吞吐量,从而在其之前尚未适用的情况下启用WebRTC通信。

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