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Choose the stream’s resolution and frame rate first, then set FFmpeg’s video bitrate to YouTube’s recommendation for that combination and codec. For a straightforward H.264 starting point, use 6 Mbps at 720p30 or 10 Mbps at 1080p30, with CBR guidance, a two-second keyframe interval, and RTMPS ingest. Those settings do not guarantee that a small VPS can encode the stream in real time: sustained CPU and network capacity must be tested.
Choose a resolution and bitrate
YouTube’s recommended bitrate depends on the ingestion codec, resolution, and frame rate. The figures below are YouTube’s H.264 recommended video bitrates; they are not total stream rates and do not include audio or network overhead. YouTube’s current encoder-settings guidance was accessed in 2026.
| Resolution and frame rate | H.264 recommended video bitrate |
|---|---|
| 360p30 | 4 Mbps |
| 480p30 | 4 Mbps |
| 720p30 | 6 Mbps |
| 720p60 | 8 Mbps |
| 1080p30 | 10 Mbps |
| 1080p60 | 17 Mbps |
See YouTube’s encoder settings for its current recommendations, including other codecs and minimums. Specify the codec whenever using a bitrate figure: these H.264 recommendations should not be assumed to apply to HEVC or AV1.
A practical first target
If the source is suitable and the VPS can keep up, 720p30 H.264 is a conservative example to test: 6 Mbps video, AAC stereo at 128 kbps, and a two-second GOP. Choose 1080p or 60 fps only if the source, CPU capacity, and outbound connection can sustain the corresponding load. A platform recommendation is not a VPS sizing guarantee.
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Configure bitrate, keyframes, and audio
YouTube recommends constant bitrate (CBR), a keyframe frequency of two seconds (not exceeding four seconds), and frame rates up to 60 fps. Its guidance lists AAC or MP3 audio and recommends 128 kbps for stereo AAC/MP3. In FFmpeg, -b:v sets video bitrate, -b:a sets audio bitrate, and -maxrate and -bufsize configure rate control. A configured target does not mean the instantaneous bitrate will be perfectly flat; check stream health in YouTube Live Control Room.
For 30 fps, a 60-frame GOP corresponds to two seconds. At 60 fps, use 120 frames for the same interval. Confirm the actual output frame rate rather than relying on the source filename or an assumed rate.
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Start a looping file stream
This illustrative command loops a file in real time and publishes H.264 video and AAC audio using RTMPS. It assumes a 30 fps source or output, a compatible input containing audio and video, and an encoder that can sustain real-time output. It is an example, not a tested universal command.
ffmpeg -re -stream_loop -1 -i input.mp4
-c:v libx264 -b:v 6000k -maxrate 6000k -bufsize 12000k -g 60
-c:a aac -b:a 128k
-f flv "rtmps://<ingest-address>/<stream-key>"- Get the ingest details. In YouTube Live Control Room, copy the ingest address and stream key for the broadcast. Replace both placeholders in the command with those values. Treat the key like a password; do not publish it or place it in a publicly accessible script.
- Check the input. Verify that the file has the audio and video streams the command expects, and inspect its actual frame rate. If audio is absent, map or generate an audio source as appropriate instead of assuming
-c:acan encode a nonexistent stream. - Set the target. For this 720p30 example,
-b:v 6000kis the 6 Mbps video target.-maxrate 6000kand-bufsize 12000kshape rate control; the buffer is not extra network bandwidth. The 60-frame GOP is two seconds only at 30 fps. - Publish and verify. Run the command, then check the preview, audio, and ingest health in YouTube Live Control Room. Resolve any reported bitrate, codec, audio, or keyframe warnings before leaving the stream unattended.
FFmpeg documents -re as real-time input pacing, which is useful when using a file as a live source. Its documentation warns against applying a low read rate to an actual live input because it can cause packet loss. For a compatible, correctly encoded source, stream copy may lower CPU demand, but verify codec and container compatibility and confirm output health before depending on it. See the FFmpeg documentation for options and behavior.
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Budget for network capacity
YouTube recommends leaving about 20% upload-bandwidth headroom and says to account for primary and backup streams if both are being sent. For the example settings, 6 Mbps video plus 128 kbps audio is about 6.13 Mbps of stream payload before overhead. With 20% headroom, plan for roughly 7.4 Mbps of sustained outbound capacity. This is arithmetic based on the example, not a provider guarantee.
Measure the VPS’s actual route and check the provider’s traffic allowance and port limits. A fast advertised port does not by itself establish sustained throughput to YouTube. If you send a backup stream, include its bitrate in the capacity calculation. YouTube’s recommendations and streaming guidance are available at Encoder settings and Streaming tips.
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Check whether a small VPS can keep up
There is no reliable universal vCPU or RAM minimum for this job. The load varies with resolution, frame rate, codec, x264 preset, motion and visual complexity, other processes sharing the VPS, and whether FFmpeg is encoding or merely forwarding compatible pre-encoded video.
- Test a representative segment with motion and audio similar to the material you intend to loop.
- Watch FFmpeg’s reported output speed and CPU use for a sustained period; a stream that falls below real time will not maintain the intended live cadence.
- Monitor YouTube’s preview and stream-health indicators as well as the VPS’s network and resource use.
- If encoding cannot sustain real time, lower resolution or frame rate, choose a faster encoder preset, or move to a plan with more dedicated compute; repeat the test after changing settings.
YouTube recommends testing with audio and motion similar to the real event and checking stream health. A VPS plan should be assessed on sustained or dedicated CPU, outbound bandwidth and traffic caps, network quality, and whether the provider permits long-running processes. No named VPS plan is established here as suitable for a particular encode.
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Use RTMPS and test before leaving it unattended
For a conventional H.264 stream, RTMPS is a sensible default: YouTube recommends it as the encrypted extension of RTMP. HLS and DASH are segment-based alternatives that generally have greater latency, though they support other codec and high-resolution use cases. Consult YouTube’s protocol and encoder guidance if your workflow needs a different ingest method.
- Start a test broadcast and confirm that both video and audio appear in the Live Control Room preview.
- Check the stream-health panel for warnings, including bitrate, codec, audio, or keyframe problems.
- Keep the test running long enough to reveal CPU, network, and process issues under representative conditions.
- For a nonstop setup, use process supervision and alerting so you can detect a stopped FFmpeg process or a failed connection. Platform settings and bitrate alone do not guarantee uninterrupted service.
Or let it run in the cloud
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Quick Recap
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