webrtc中SendSideBandwidthEstimation类的设计
The SendSideBandwidthEstimation class in WebRTC is a critical component in its video engine. It’s responsible for deciding the video traffic rate that can be sent without overloading the network and thus maintaining video quality. Earlier algorithms were rudimentary, relying heavily on packet loss to gauge bandwidth. Modern algorithms, like Google Congestion Control used in WebRTC, are more sophisticated, predicting congestion by analyzing delays between packets1.
Recent versions of Chrome have shifted the bandwidth estimation logic to the sender side. This change includes the use of transport wide sequence numbers and transport feedback protocols for better delay information from the receiver, which is crucial for estimating available bandwidth1.
原文地址:https://blog.csdn.net/commshare/article/details/134603793
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