sebiulabs realtime Documentation

Contents

Overview

Sebiulabs Realtime is a low-latency live streaming platform with sub-second delay. Stream from your browser, OBS, or any WHIP-compatible encoder and deliver to viewers worldwide via WebRTC.

Architecture

  • Origin servers -- receive and process your stream
  • Edge servers -- distributed across Europe and North America for fast delivery to viewers
  • Geo-router -- automatically sends each viewer to the closest server
  • Auto-scaler -- adds or removes edge servers based on viewer demand

Key Features

  • Sub-second delay via WebRTC (WHIP/WHEP)
  • Viewers are automatically connected to the closest server
  • Edge servers scale up and down with demand
  • Each user gets their own stream key for secure access
  • Real-time dashboard with server and stream monitoring
  • H.264 and H.265/HEVC codec support

Web Broadcaster

Stream directly from your browser — no software to install. The Web Broadcaster uses the WHIP protocol to send video and audio from your camera or screen to any compatible server.

Quick Start

  1. Open the Web Broadcaster
  2. Select your server type from the dropdown (Sebiulabs Realtime, Gateway Pro, Twitch, and more)
  3. Enter the connection details or paste the full WHIP URL
  4. Choose your source (camera or screen share)
  5. Click Go Live

Supported Platforms

The broadcaster includes built-in presets for:

Settings

H.264 + Opus is the safest combination for compatibility across all platforms. Unsupported codecs are automatically greyed out based on your browser.

Requirements

Press Escape to stop a broadcast. You will be warned before leaving the page while live.

Want the full documentation? Sign in to see ingest protocols, OBS/vMix/FFmpeg setup guides, stream keys, embedding, and more.

Sign In Create Account

Getting Started

  1. Log in to the dashboard with credentials provided by your admin.
  2. Go to the Send tab and copy your stream key (starts with sk_).
  3. Configure your encoder (OBS, FFmpeg, etc.) using the settings below.
  4. Hit Start Streaming — your stream appears on the dashboard within seconds.
First time streaming? We recommend the WHIP protocol for the lowest delay. See the OBS Setup section below for step-by-step instructions.

Ingest Protocols

The dashboard automatically detects your region and shows the nearest ingest server. Americas users connect to cdn-ca.clouds3.uk (Canada), everyone else to cdn.clouds3.uk (EU). Use whichever is shown in your Send tab.

WHIP WebRTC Ingest

Sub-second latency. Recommended for interactive streams.

URL formathttps://cdn.clouds3.uk/live/{name}?direction=whip&key={key}
Latency< 500ms
CodecsH.264, VP8, Opus

RTMP Classic Ingest

Works with virtually every streaming encoder.

Serverrtmp://cdn.clouds3.uk:1935/live
Stream Key{name}?key={key}
Latency3-5 seconds
CodecsH.264, H.265, AAC

SRT Low-Latency Ingest

Built-in error correction. Ideal for unreliable or remote connections.

URL formatsrt://cdn.clouds3.uk:9999?streamid=default/live/{name}?key={key}
Port9999/UDP
Latency1-2 seconds
CodecsH.264, H.265, AAC, Opus

In all URLs above, replace {name} with your chosen stream name (e.g. my-stream) and {key} with your stream key from the dashboard.

OBS Studio Setup

OBS Studio (version 30 and later) supports all three streaming protocols. Follow the step-by-step instructions for your preferred protocol below.

WHIP OBS → WebRTC (Recommended — Lowest Latency)

  1. Open OBS → SettingsStream
  2. Set Service to WHIP
  3. In the Server field, paste the full URL including your stream name and key:
https://cdn.clouds3.uk/live/my-stream?direction=whip&key=sk_your_key
  1. Leave the Bearer Token field empty
  2. Go to Settings → Output and set Output Mode to Advanced
  3. On the Streaming tab, set:
Encoderx264 or NVENC H.264
Rate ControlCBR
Bitrate2500 kbps (720p) or 4500 kbps (1080p)
Keyframe Interval1 s
CPU Presetveryfast
Profilebaseline
Tunezerolatency
Audio EncoderOpus (preferred for WHIP) or AAC
  1. Click OK, then click Start Streaming
WHIP streams appear on the dashboard within 1-2 seconds. The total delay from your camera to your viewers is typically under 500ms.

RTMP OBS → Classic

  1. Open OBS → SettingsStream
  2. Set Service to Custom...
  3. Fill in these two fields exactly:
Serverrtmp://cdn.clouds3.uk:1935/live
Stream Keymy-stream?key=sk_your_key

Replace my-stream with your chosen stream name and sk_your_key with your actual key from the Send tab.

  1. Go to Settings → Output → Advanced → Streaming and configure the encoder settings (see table below)
  2. Click OK, then click Start Streaming

SRT OBS → Low-Latency

SRT provides reliable, low-delay delivery with built-in error correction. Use it when streaming over unstable or remote connections.

  1. Open OBS → SettingsStream
  2. Set Service to Custom...
  3. In the Server field, paste the full SRT URL:
srt://cdn.clouds3.uk:9999?streamid=default/live/my-stream?key=sk_your_key
  1. Leave the Stream Key field empty (the stream ID is embedded in the URL)
  2. Go to Settings → Output → Advanced → Streaming and configure encoder settings (see table below)
  3. Click OK, then click Start Streaming
SRT Latency: OBS defaults to 120ms SRT latency. For unstable networks, increase the latency parameter: srt://cdn.clouds3.uk:9999?streamid=default/live/my-stream?key=sk_your_key&latency=200000 (200ms, value in microseconds).

Recommended OBS Output Settings

Maximum ingest bitrate: 10 Mbps. Streams exceeding 10 Mbps will be automatically disconnected. All recommended settings below are well within this limit.

Go to Settings → Output, set Output Mode to Advanced, then configure the Streaming tab:

H.264 (Universal)

Encoderx264 or NVENC H.264
Rate ControlCBR
Bitrate2500-4000 kbps (720p) / 4500-6000 kbps (1080p)
Keyframe Interval1 s
CPU Presetveryfast
Profilebaseline
Tunezerolatency

H.265/HEVC (Better Quality)

EncoderNVENC HEVC or QSV HEVC
Rate ControlCBR
Bitrate2000-3000 kbps (720p) / 3500-5000 kbps (1080p)
Keyframe Interval1 s
PresetP4 (quality)
Profilemain

Requires GPU encoder (NVIDIA GTX 900+ / Intel 6th gen+). ~30% better quality at the same bitrate vs H.264.

Important: Set Keyframe Interval to 1 second. Leaving it at the default (0 or 2s) will cause noticeably higher delay for your viewers.

vMix Setup

vMix supports both SRT and RTMP output. Configure one of the following in vMix's output settings.

SRT vMix → SRT Output (Recommended)

  1. In vMix, click the gear icon below the stream button, or go to Settings → Outputs
  2. Click Output Settings on one of the output channels
  3. Set Output Type to SRT (Caller)
  4. Fill in the following fields:
Hostnamecdn.clouds3.uk
Port9999
Stream IDdefault/live/my-stream?key=sk_your_key
Latency120 ms (increase to 200 for unstable connections)
Key Length0 (no encryption)
  1. Under Video, set the codec and quality:
Video CodecH.264 (or H.265 if your GPU supports it)
Bitrate4000 kbps (1080p) / 2500 kbps (720p)
ProfileBaseline (H.264) or Main (H.265)
Keyframe Interval1 s
Audio CodecAAC
Audio Bitrate128 kbps
  1. Click OK, then click the Stream button to go live

RTMP vMix → RTMP Output

  1. In vMix, click the gear icon below the stream button
  2. Set Destination to Custom RTMP Server
  3. Fill in the following fields:
URLrtmp://cdn.clouds3.uk:1935/live
Stream Keymy-stream?key=sk_your_key
  1. Set video quality: H.264, CBR at 4000 kbps, Keyframe every 1s, Profile: Baseline
  2. Click OK, then click the Stream button to go live
Tip: vMix SRT Caller mode is recommended over RTMP because it offers lower latency (~1-2s vs 3-5s) and handles packet loss better on unreliable networks.

Restreamer Setup

Restreamer (by datarhei) is an open-source tool for relaying video sources to streaming platforms. Use it to push an existing source (IP camera, HDMI capture, file) to sebiulabs realtime.

Step-by-Step Configuration

  1. Open the Restreamer UI in your browser (usually http://your-server:8080)
  2. Set up your video source (RTSP camera, screen capture, etc.) on the Video Setup page
  3. On the Streaming / Publication page, click Add new process or edit the existing output
  4. Set Output Type to one of the following:

RTMP Restreamer Output

In the Restreamer output configuration:

ProtocolRTMP
Address / URLrtmp://cdn.clouds3.uk:1935/live/my-stream?key=sk_your_key

Or if Restreamer asks for server and key separately:

Serverrtmp://cdn.clouds3.uk:1935/live
Stream Keymy-stream?key=sk_your_key

SRT Restreamer Output

In the Restreamer output configuration:

ProtocolSRT
ModeCaller
Address / URLsrt://cdn.clouds3.uk:9999?streamid=default/live/my-stream?key=sk_your_key&mode=caller
Latency120 ms
  1. Set the video/audio encoding parameters:
Video CodecH.264 (libx264)
Video Bitrate2500-4000 kbps
Framerate30 fps
GOP / Keyframe30 frames (= 1 second at 30fps)
Audio CodecAAC
Audio Bitrate128 kbps
  1. Click Save and Start the process

Restreamer with FFmpeg Process (Advanced)

If using Restreamer's FFmpeg process directly, use this command template:

{ffmpeg.binary} -i {input} \
  -c:v libx264 -preset veryfast -tune zerolatency \
  -b:v 3000k -g 30 -keyint_min 30 \
  -c:a aac -b:a 128k \
  -f flv "rtmp://cdn.clouds3.uk:1935/live/my-stream?key=sk_your_key"

FFmpeg Command-Line Examples

FFmpeg can be used to ingest from files, cameras, screen captures, or any other source. Below are copy-pasteable commands for each protocol.

RTMP FFmpeg → RTMP

Stream a file or live source via RTMP:

# Stream a video file to RTMP
ffmpeg -re -i input.mp4 \
  -c:v libx264 -preset veryfast -tune zerolatency \
  -b:v 3000k -maxrate 3000k -bufsize 6000k \
  -g 30 -keyint_min 30 \
  -c:a aac -b:a 128k -ar 48000 \
  -f flv "rtmp://cdn.clouds3.uk:1935/live/my-stream?key=sk_your_key"
# Stream a webcam (Linux) to RTMP
ffmpeg -f v4l2 -i /dev/video0 -f pulse -i default \
  -c:v libx264 -preset veryfast -tune zerolatency \
  -b:v 2500k -g 30 -keyint_min 30 \
  -c:a aac -b:a 128k -ar 48000 \
  -f flv "rtmp://cdn.clouds3.uk:1935/live/my-stream?key=sk_your_key"

SRT FFmpeg → SRT

Stream via SRT for lower latency with error correction:

# Stream a file via SRT
ffmpeg -re -i input.mp4 \
  -c:v libx264 -preset veryfast -tune zerolatency \
  -b:v 3000k -g 30 -keyint_min 30 \
  -c:a aac -b:a 128k -ar 48000 \
  -f mpegts "srt://cdn.clouds3.uk:9999?streamid=default/live/my-stream?key=sk_your_key&latency=120000&mode=caller"
# Stream with H.265/HEVC via SRT (better quality, lower bitrate)
ffmpeg -re -i input.mp4 \
  -c:v libx265 -preset veryfast -tune zerolatency \
  -b:v 2000k -g 30 -keyint_min 30 \
  -c:a aac -b:a 128k -ar 48000 \
  -f mpegts "srt://cdn.clouds3.uk:9999?streamid=default/live/my-stream?key=sk_your_key&latency=120000&mode=caller"
SRT latency parameter: The value is in microseconds. 120000 = 120ms (default). For unstable networks, try 200000 (200ms) or 500000 (500ms).

WHIP FFmpeg → WHIP (WebRTC)

FFmpeg 7.0+ supports WHIP output for sub-second latency:

# Stream via WHIP (requires FFmpeg 7.0+ built with libdatachannel or libwebrtc)
ffmpeg -re -i input.mp4 \
  -c:v libx264 -preset veryfast -tune zerolatency \
  -b:v 2500k -g 30 -keyint_min 30 \
  -c:a libopus -b:a 128k -ar 48000 \
  -f whip "https://cdn.clouds3.uk/live/my-stream?direction=whip&key=sk_your_key"
Note: WHIP output in FFmpeg requires a build compiled with WebRTC support. Most standard FFmpeg packages do not include this. If unavailable, use RTMP or SRT instead, or use OBS for WHIP ingest.

FFmpeg Quick Reference

FlagPurposeRecommended Value
-reRead input at native framerate (required for files, omit for live sources)Include for files
-preset veryfastEncoding speed/quality tradeoffveryfast or ultrafast
-tune zerolatencyDisables lookahead for lowest encoding latencyAlways include
-g 30GOP size (keyframe interval in frames)30 for 30fps, 60 for 60fps (= 1 second)
-b:v 3000kVideo bitrate2500k (720p) / 4500k (1080p)
-c:a aacAudio codecaac for RTMP/SRT, libopus for WHIP
-f flvOutput format for RTMPflv (RTMP), mpegts (SRT)

Playback & Sharing

WebRTC Built-in Player

Share this link with viewers for sub-second latency:

https://dashboard.clouds3.uk/watch.html#my-stream

The player automatically connects to the nearest edge node.

WebRTC Playback URL

For direct WebSocket signalling (custom players):

wss://cdn.clouds3.uk/live/my-stream

Edge Playback

You can replace cdn.clouds3.uk with a specific edge server hostname if needed. In most cases, the geo-router at play.clouds3.uk handles this automatically and sends each viewer to the best server.

Embedding on Your Website

You can place a live stream on any website by adding an iframe. The embedded player automatically connects viewers to the nearest server for the best performance.

Basic Embed (iframe)

<iframe
  src="https://dashboard.clouds3.uk/embed.html?s=my-stream"
  width="640" height="360"
  frameborder="0" allowfullscreen
  allow="autoplay"></iframe>

Responsive Embed

<div style="position:relative;padding-bottom:56.25%;height:0;overflow:hidden;">
  <iframe
    src="https://dashboard.clouds3.uk/embed.html?s=my-stream"
    style="position:absolute;top:0;left:0;width:100%;height:100%;"
    frameborder="0" allowfullscreen
    allow="autoplay"></iframe>
</div>

Options

ParameterDescriptionDefault
sStream name (required)
nodeEdge server hostname (e.g. edge1.clouds3.uk)play.clouds3.uk (geo-routed)

The embedded player connects via WebRTC through the geo-router for sub-second delay. You can override the edge server with the node parameter if needed.

The embedded player is publicly accessible. Viewers do not need to log in -- they just need the stream name.

Stream Keys

Every user gets a unique stream key (e.g. sk_a1b2c3d4e5f6). Your stream key proves you are authorized to publish a stream. Think of it like a password for your stream.

Dashboard

The dashboard at dashboard.clouds3.uk provides:

Troubleshooting

Stream not appearingCheck your stream key is correct and included as ?key=sk_... in the URL. Verify the server URL has no typos.
Connection failed in OBSCheck firewall allows outbound on port 443 (WHIP), 1935 (RTMP), or 9999 (SRT). Try a different protocol.
Too much delaySet keyframe interval to 1s. Use WHIP protocol for the lowest delay. Use CBR rate control, not VBR.
Viewers are bufferingTry lowering your bitrate. 720p at 2500 kbps works well for most connections.
Stream disconnected automaticallyYour ingest bitrate may have exceeded the 10 Mbps limit. Lower your encoder bitrate and try again.
Stream drops or reconnectsCheck your upload speed. RTMP needs stable upload of at least 1.5x your bitrate. Try SRT for unreliable connections.
No audioMake sure the audio encoder is set to AAC (RTMP/SRT) or Opus (WHIP). Also check that your OBS audio mixer is not muted.
Need a new stream keyGo to Send tab → Regenerate. Old key stops working immediately.
Forgot passwordContact your admin to reset your password.

FAQ

Which protocol should I use?

Use WHIP for the lowest delay (under 500ms). Use RTMP if you need the widest encoder compatibility. Use SRT if your internet connection is unreliable (mobile, satellite, remote locations).

Can multiple people stream at the same time?

Yes. Each user has their own stream key and can stream to their own stream name simultaneously.

What resolution/framerate should I use?

1080p30 is a good default. Use 720p30 if your upload is limited. 1080p60 for fast-motion content (gaming, sports) if you have the bandwidth. Keep your total bitrate under 10 Mbps.

How do viewers watch my stream?

Share the watch link: https://dashboard.clouds3.uk/watch.html#your-stream-name. Viewers do not need to log in.

Is H.265/HEVC supported?

Yes. HEVC gives ~30% better quality at the same bitrate. Requires a GPU encoder (NVENC or QSV) and a compatible browser (Chrome 107+, Safari, Edge).

sebiulabs realtimeDashboard