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Why video bitrate is small in WebRTC publish and play?

Silas Getachew


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WebRTC publish and play quality depends on E2E connection reliability/bandwidth and RTT for latency. As an example, imagine publishing and playing in a local environment where you have 100 mbit connection between devices. You will have the best quality since distance is very short and it is directly connected to the device itself. On the other hand, assume that you still have 100 mbit connection and you publish/play and your peer is very far from you and only have limited connection. In this case, the quality/latency will be scaled to the worst part of the equation which is your peer.

If my explanation wasn't enough, I will try to elaborate in a better way.

Best Regards,


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