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Real-time mixer can be tuned using the following parameters
Parameter | Default value | Description |
|---|---|---|
| mixer_audio_silence_threshold | -50.00 | Incoming stream audio silence level in Db |
| mixer_debug_mode | false | Adds some debug info to stream picture caption |
mixer_in_buffering_ms | 200 | Incoming stream video buffer in milliseconds |
| mixer_incoming_time_rate_lower_threshold | 0.95 | Relative incoming stream time to mixer time rate lower threshold |
| mixer_incoming_time_rate_upper_threshold | 1.05 | Relative incoming stream time to mixer time rate upper threshold |
| mixer_video_stable_fps_threshold | 15 | Incoming stream FPS threshold, video buffer will be disabled for streams with low fps |
Testing
1. For test we use
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Call flow for conferencing example based on real-time mixer with MCU function is described on MCU Client page.
Incoming streams tuning recommendations
When delay occurs in the incoming stream from one of participants, realtime mixer will freeze that stream. The following is recommended to minimize incoming stream delays:
1. For RTMP streams, adjust encoding parameter so that:
- client encoder perfomance to be enough to send picture frames in time
- stream resolution and bitrate to fit to publishers channel from client to server
2. For WebRTC streams, do not raise minimum video bitrate threshold higher than webrtc_cc_min_bitrate server configuration parameter defines. By default, lower video bitrate threshold is set to 30 kbps
| Code Block | ||
|---|---|---|
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webrtc_cc_min_bitrate=30000 |
The publisher client browser will adopt the stream to channel quality drops. The lower bitrate the lower picture quality, but the participant stream will not be freezed in this case.