Abstract

Real-time video conferencing systems have recently become indispensable tools. However, existing commodity video conferencing systems often fail to deliver a satisfactory quality of experience (QoE) due to discrepancies between the actual network throughput and the average bitrate of encoded videos. In this paper, we present a system called Tyrus that aims to achieve efficient real-time video conferencing. Tyrus adapts its delivery of B-frames based on the real network bandwidth and the average bitrate of encoded videos, effectively addressing the QoE loss caused by mismatch issues. This system can be easily implemented on top of prevalent video conferencing systems that utilize standard video encoders. Unlike traditional approaches that treat all frame types equally, Tyrus enables adaptive delivery of B-frames, prioritizing the allocation of bandwidth to other crucial video frames. By doing so, it reduces end-to-end conferencing latency, especially when the network throughput experiences significant fluctuations. Additionally, Tyrus defines the playing deadline for B-frames by considering factors such as delivery time, buffering time, and the progress of preceding frames. It accurately classifies frame types according to video coding standards and estimates bandwidth allocation on a frame-by-frame basis. Moreover, Tyrus proactively handles potential errors in B-frame playing deadline estimation, minimizing their impact on video conferencing performance. Our evaluation results, obtained through real implementation on top of WebRTC, demonstrate the effectiveness of Tyrus. On average, it reduces per-frame latency by 19.03%, video stalls by 22.36%, and improves bandwidth utilization by at least 10.29%.

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