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Note that ffmpeg sends orientattion value but not angle itself.
Known issues
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4. Some RTMP functions does not supported and will be ignored:
- FCSubscribe
- FCPublish
- FCUnpublish
- onStatus
- onUpstreamBase
- releaseStream
5. Some RTMP-encoders does not support KeepAlive.
Symptoms: disconnection occurs often while stream publishing with RTMP-encoder.
Solution: switch KeepAlive off for RTMP on the server using the following parameter in flashphoner.properties file
| Code Block | ||||
|---|---|---|---|---|
| ||||
keep_alive.enabled=websocket,rtmfp |
7. When stream published with RTMP encoder is played as HLS, freezes may occur if GOP is not multiple of FPS of file published
Symptoms: freezes occur when RTMP stream is played as HLS
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Decoding buffer size management
When RTMP stream is published from software encoder supportin hardware acceleration on NVIDIA GPU, and contains B-frames, the stream picture can twitch in some RTMP or HLS players. To workaround this, since build 5.2.863 the parameter was added to set the maximum decoding buffer size passed in SPS
| Code Block | ||
|---|---|---|
| ||
h264_sps_max_dec_frame_buffering=-1 |
By default, buffer size is not limited. This eliminates picture twitching, but can inscrease latency due to excessive buffering. In this case, the buffer size can be limited to 2 frames (default value before build 5.2.863)
| Code Block | ||
|---|---|---|
| ||
h264_sps_max_dec_frame_buffering=2 |
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 | ||
|---|---|---|
| ||
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
| Excerpt Include | ||||||
|---|---|---|---|---|---|---|
|
4. Some RTMP functions does not supported and will be ignored:
- FCSubscribe
- FCPublish
- FCUnpublish
- onStatus
- onUpstreamBase
- releaseStream
5. Some RTMP-encoders does not support KeepAlive.
Symptoms: disconnection occurs often while stream publishing with RTMP-encoder.
Solution: switch KeepAlive off for RTMP on the server using the following parameter in flashphoner.properties file
| Code Block | ||||
|---|---|---|---|---|
| ||||
keep_alive.enabled=websocket,rtmfp |
7. When stream published with RTMP encoder is played as HLS, freezes may occur if GOP is not multiple of FPS of file published
Symptoms: freezes occur when RTMP stream is played as HLS
Solution: in RTMP encoder settings, assign GOP to value equal or multiple of FPS of file published. For example, when publishing file with FPS 25 set GOP to 50.
8. When stream published with RTMP encoder is played as WebRTC, stream audio has low quality
Symptoms: sound is good enogh for speech but not for music when RTMP is played as WebRTC
Solution: set Opus encoding bitrate as published one, for example
| Code Block | ||
|---|---|---|
| ||
opus.encoder.bitrate=160000 |
if RTMP stream is published with audio bitrate 160 kbps
9. High CPU load when stream H264+speex is published (for example, using Flash client) with audio transcoding
Symptoms: high CPU load while audio transcoding from speex to AAC or Opus
Solution: use native speex decoder implementation
| Code Block | ||
|---|---|---|
| ||
use_speex_java_impl=false |
10. Stream with unsupported codecs cannot be published
Symptoms: RTMP stream with MP3 or AC3 audio is not publishing with the following warnings in server log
| Code Block | ||
|---|---|---|
| ||
11:01:00,921 WARN ServerHandler - RTMP-pool-15-thread-1 Codecs not supported! audio: NoCodec, video: NoCodec |
Solution: reencode stream to supported codecs using corresponding encoder settings