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The bitrate of a live stream is generally described in the metadata, but this may not be accurate. Therefore, accurate data is usually obtained by analyzing the actual stream. Moreover, the bitrate can be adjusted during the live broadcast without disconnecting the RTMP link. Even with live streams that convert WebRTC to RTMP, the bitrate is constantly fluctuating. As a result, there is no way to know the bitrate of a live stream in advance. SRS can only determine the accurate bitrate after receiving the stream for a certain period. This is similar to FFmpeg, which also requires analyzing the stream for a certain duration before it can provide bitrate data. Note: The term "accurate bitrate" can be misleading because the bitrate is an average over a certain period, and the bitrate can vary across different time dimensions.
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Is there a way to block RTMP streams that are incoming with more than a set amount of bitrate? I would prefer to not have to poll the API.
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