Publish Streams to Red5 Cloud via RTMP and WebRTC

Configure RTMP and WebRTC (WHIP) encoders to publish live streams to Red5 Cloud, including OBS settings, codec recommendations, and FFmpeg command examples.

Red5 Cloud accepts live streams over two ingest protocols: RTMP for traditional software and hardware encoders, and WebRTC via WHIP for browser-based or WebRTC-native publishers. Both deliver low-latency streams to your Red5 Cloud deployment. Your publishing endpoint for either protocol is available in the Pub/Sub Details of your deployment.

RTMP

RTMP encoding

Real-Time Messaging Protocol (RTMP) is a TCP-based protocol with persistent connections and broad support across encoders, hardware capture cards, and streaming software. It is the most common way to send a live stream from a desktop or on-premises encoder to a streaming server.

Getting your RTMP endpoint

1. Open Pub/Sub Details

   In the Management Console, open the context menu for your deployment and select Pub/Sub Details.

2. Copy the RTMP Publishing Endpoint

   The RTMP Publishing Endpoint is shown in the format:

   rtmp://<server-ip>/live
   

   The stream key you use in your encoder becomes the stream name on Red5 Cloud.

Recommended video settings

Setting	Recommendation
Video codec	H.264
H.264 profile	Baseline (disables B-frames)
Keyframe interval	2 seconds

Do not use H.264 profiles that include B-frames (such as High or Main) when delivering to WebRTC subscribers. B-frames cause playback to jump back and forth. Use the Baseline profile or explicitly disable B-frames in your encoder.

Recommended video bitrates

Resolution / FPS	Bitrate
360p @ 30fps	500–800 kbps
480p @ 30fps	800–1,200 kbps
720p @ 30fps	2,000–2,500 kbps
720p @ 60fps	2,500–3,050 kbps
1080p @ 30fps	3,500–4,500 kbps
1080p @ 60fps	4,500–6,000 kbps
4K @ 30fps	10,000–12,000 kbps
4K @ 60fps	12,000–14,000 kbps

Recommended audio settings

Setting	Recommendation
Preferred codec	AAC
Fallback codec	MP3 (only if AAC is unavailable)
Stereo bitrate	128 kbps minimum; 192 kbps for higher quality

Publishing with OBS Studio

1. Open OBS Settings

   In OBS Studio, go to File → Settings → Stream.

2. Set service to Custom

   Set Service to Custom....

3. Enter the server URL

   Paste your RTMP Publishing Endpoint into the Server field.

4. Set the stream key

   Enter your desired stream name as the Stream Key (for example, myStream).

5. Configure output settings

   Go to Settings → Output → Encoding. Select H.264, set the profile to Baseline, and choose a bitrate from the table above that matches your target resolution.

6. Start streaming

   Click Start Streaming in the OBS main window.

Publishing with FFmpeg

ffmpeg \
  -re \
  -i input.mp4 \
  -c:v libx264 \
  -profile:v baseline \
  -preset veryfast \
  -b:v 2500k \
  -g 60 \
  -c:a aac \
  -b:a 128k \
  -f flv \
  rtmp://<server-ip>/live/myStream

Replace <server-ip> with your RTMP Publishing Endpoint host and myStream with your chosen stream name.

WebRTC (WHIP)

WebRTC (WHIP) encoding

WebRTC HTTP Ingest Protocol (WHIP) is a standardized way to push a WebRTC stream to a server over a simple HTTP POST. WHIP works natively in modern browsers, making it the easiest path for browser-based publishing without any plugins or additional software.

Getting your WHIP endpoint

1. Open Pub/Sub Details

   In the Management Console, open the context menu for your deployment and select Pub/Sub Details.

2. Copy the WHIP Publishing Endpoint

   The WHIP Publishing Endpoint is shown in the format:

   https://<server-host>/live/myStream
   

   Replace myStream with the stream name you want to use.

Recommended video settings

Setting	Recommendation
Video codec	H.264
H.264 profile	Baseline (disables B-frames)
Keyframe interval	2 seconds

Recommended video bitrates

Resolution / FPS	Bitrate
360p @ 30fps	500–800 kbps
480p @ 30fps	800–1,200 kbps
720p @ 30fps	2,000–2,500 kbps
720p @ 60fps	2,500–3,050 kbps
1080p @ 30fps	3,500–4,500 kbps
1080p @ 60fps	4,500–6,000 kbps
4K @ 30fps	10,000–12,000 kbps
4K @ 60fps	12,000–14,000 kbps

Recommended audio settings

Setting	Recommendation
Preferred codec	AAC
Fallback codec	Opus (Red5 Cloud transcodes Opus to AAC for delivery)
Stereo bitrate	128 kbps minimum; 192 kbps for higher quality

If your browser or WebRTC client sends Opus audio, Red5 Cloud automatically transcodes it to AAC before delivering to subscribers. No additional configuration is required on your end.

Publishing from a browser

Most modern browsers (Chrome, Firefox, Safari, Edge) support WHIP publishing using the browser’s getUserMedia API combined with RTCPeerConnection. Point your WHIP client at the WHIP Publishing Endpoint from your Pub/Sub Details.

The embedded player code in Pub/Sub Details also provides a ready-made starting point for browser-based publishers and subscribers.

Publishing with FFmpeg (WHIP)

Recent versions of FFmpeg support WHIP output natively:

ffmpeg \
  -re \
  -i input.mp4 \
  -c:v libx264 \
  -profile:v baseline \
  -preset veryfast \
  -b:v 2500k \
  -g 60 \
  -c:a aac \
  -b:a 128k \
  -f whip \
  https://<server-host>/live/myStream

Replace <server-host> and myStream with the values from your WHIP Publishing Endpoint.

Choosing between RTMP and WHIP

	RTMP	WebRTC (WHIP)
Best for	Desktop encoders, OBS, hardware	Browser-based apps, low-dependency clients
Protocol	TCP	UDP / HTTP
Latency	Low (~2–5 s)	Sub-second
Encoder support	Very broad	Growing rapidly

For the lowest possible glass-to-glass latency, use WHIP for ingest and WHEP for playback. For maximum encoder compatibility and a proven workflow, use RTMP.