Versions Compared


  • This line was added.
  • This line was removed.
  • Formatting was changed.


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

Code Block

Adaptive RTMP incoming buffer has the following parameters to tune:

ParameterDescriptionDefault value
rtmp_in_buffer_start_sizeMinimum buffer volume to start, ms300
rtmp_in_buffer_initial_sizeMaxumum buffer volume, ms2000
rtmp_in_buffer_max_bufferings_allowedMaximum bufferings amout allowed-1 (unlimited)
rtmp_in_buffer_polling_timeBuffer polling period, ms100
rtmp_in_buffer_overflow_allowed_deviationMaximum difference between minimum and maximum buffer volumes used, ms1000
rtmp_in_buffer_overflow_deviation_windowWindow size to collect the difference, ms30000
rtmp_in_buffer_overflow_rateMaximum buffer overflow rate0.15
rtmp_in_buffer_clear_thresholdClear all the data exceeding maxumum buffer size when buffer reaches the threshold, ms30000

Known issues

Excerpt Include
From another server via RTMP
From another server via RTMP