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or more frames to eliminate twitching without viewable latency.
Incoming RTMP stream buffering
RTMP stream published with high resolution and bitrate may be played non smoothly via WebRTC with freezes or low FPS if publishers channel is unstable. Incoming stream should be buffered to prevent playback issues
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rtmp_in_buffer_enabled=true |
Adaptive RTMP incoming buffer has the following parameters to tune:
| Parameter | Description | Default value |
|---|---|---|
| rtmp_in_buffer_start_size | Minimum buffer volume to start, ms | 300 |
| rtmp_in_buffer_initial_size | Maxumum buffer volume, ms | 2000 |
| rtmp_in_buffer_max_bufferings_allowed | Maximum bufferings amout allowed | -1 (unlimited) |
| rtmp_in_buffer_polling_time | Buffer polling period, ms | 100 |
| rtmp_in_buffer_overflow_allowed_deviation | Maximum difference between minimum and maximum buffer volumes used, ms | 1000 |
| rtmp_in_buffer_overflow_deviation_window | Window size to collect the difference, ms | 30000 |
| rtmp_in_buffer_overflow_rate | Maximum buffer overflow rate | 0.15 |
| rtmp_in_buffer_clear_threshold | Clear all the data exceeding maxumum buffer size when buffer reaches the threshold, ms | 30000 |
Known issues
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