This article provides an overview of how to set up and use Phenix's Real-time RTMP ingest feature.
Since an RTMP stream is sent from an encoder to Phenix via RTMP, the encoder must be set up to send its output to Phenix. This means that, while the Phenix Customer Portal can be used to obtain information such as the stream key, the Portal’s publishing tab cannot be used to publish RTMP to the Phenix platform.
Encoder Configuration for RTMP Ingest
Your encoder's configuration must be set up to ensure that it is not introducing unnecessary latency. The number one issue that causes high latency is not using the correct low-latency settings on the local encoder.
For details on setting up your source, please refer to Optimal RTMP Encoder Settings
Measure Latency
To characterize the latency of your source, try measuring the lag using ffplay with the "nobuffer" setting. Using ffplay is always recommended to determine the baseline latency when evaluating encoder settings. The argument will be similar to:
ffplay -fflags nobuffer <url>
If the latency is not extremely low, that indicates that the encoder is not configured correctly for low latency.
Encoding Settings
Check that encoder settings are set to the lowest possible values that will provide acceptable results, such as baseline profile, CBR, 5 Mbps, 1-second GOP size.
Phenix supports H.264-encoded video as ingest, and recommends using Constrained Baseline Profile up to Level 5.2 with a maximum frame size of 36864 macro-blocks for best performance. Be sure to verify that features such as scene change detection are not enabled to keep the encoding consistent. Some video feeds may appear to work initially, but will fail when certain unsupported features are used (e.g., when the encoding is optimized for a high motion scene).
The source bitrate must be less than 8Mbps, with no spikes in the bitrate exceeding that value. There must be zerolatency interleaving, and no grouping of audio or video packets. The audio and video packets should be in strict increasing order based on the timestamp in the signal.
The audio must be 48kHz AAC, either mono or stereo.
When using single-bitrate output (i.e., when the multi-bitrate capability is not set), the video quality/resolution is not changed; that is, the same video quality input is the quality that is output.
For example, if FHD video is provided to Phenix via SRT, and the output quality is set to SD, the output will be FHD.
AWS Elemental Live Encoder
Various parameters may need to be modified to optimize for smooth playback. For Elemental Live version="2.22.4.1632837480838", use the following:
Protocol="RTMP"
Key points for smooth playback:
Scene Change Detection="off"
Use Baseline Profile of H.264
GOP Size(keyframe interval)="1 Sec"
Repeat PPS=True #Places a PPS header on each encoded picture, even if repeated.
Slices="1"
Adaptive Quantization="Auto"
Use AAC-LC Stereo 128kbps Sample Rate "48kHz"
FFMPEG
When using ffmpeg, be sure to use the zerolatency option to get real-time output. This is not set automatically. Without the zerolatency setting the encoder will introduce 0.5 seconds of end-to-end latency out of the gate. Please refer to the ffmpeg documentation for details.
An example ffmpeg command to create test content is:
ffmpeg -re -f lavfi -i testsrc=size=1280x720:rate=30 -f lavfi -i aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)" -pix_fmt yuv420p -vcodec libx264 -profile:v baseline -deblock 1:0 -bitrate 500k -tune zerolatency -x264opts keyint=30:min-keyint=30 -acodec aac -ar 48000 -ac 2 -b:a 128k -f mpegts "udp://239.0.0.1:1234?pkt_size=1316"
To measure latency, add a timestamp overlay:
ffmpeg -re -f lavfi -i testsrc=size=1280x720:rate=30 -f lavfi -i aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)" -pix_fmt yuv420p -vf "drawtext=text='timestamp: %{pts \: hms}': x=20: y=20: fontsize=72:fontcolor=white@0.9: box=1: boxcolor=black@0.6" -vcodec libx264 -profile:v baseline -deblock 1:0 -bitrate 500k -tune zerolatency -x264opts keyint=30:min-keyint=30 -acodec aac -ar 48000 -ac 2 -b:a 128k -f mpegts "udp://239.0.0.1:1234?pkt_size=1316"
API Usage
For details on API usage, please refer to https://phenixrts.com/docs/api/?http#rtmp-ingest . To take advantage of Real-time RTMP, you must include the mpegts-multicast-ingest capability.
Example
An example URL is:
rtmp://ingest.phenixrts.com:80/ingest/96characterStreamKey;capabilities=hd,multi-bitrate,prefer-h264,mpegts-multicast-ingest;tags=my-awesome-stream-id,MyAppId
Where:
96characterStreamKey is the stream key to be used when publishing. This can be found in the Customer Portal under the Properties tab for a Channel or Room.
MyAppId is your application ID