WebRTC Server - Monibuca WHIP/WHEP Low-Latency

WebRTC is the standard technology for ultra-low latency audio/video communication in browsers. Monibuca V6 implements publishing and subscribing based on the WHIP (WebRTC-HTTP Ingestion Protocol) and WHEP (WebRTC-HTTP Egress Protocol) standards, powered by the rustrtc engine for full ICE/DTLS/SRTP support.

Basic Information

Property	Value
Signaling Transport	HTTP (WHIP/WHEP)
Media Transport	UDP / optional TCP ICE (ICE/DTLS/SRTP)
Publish	✅ Supported (WHIP)
Subscribe	✅ Supported (WHEP)
Latency	< 500ms
Config section	webrtc

Configuration

Enable plugin

The plugin is included in official binaries; configure it in config.yaml:

Configuration File

webrtc:
  enable: true
  # UDP only (default). Dual ICE: "udp:9000-9100,tcp:9101-9200"
  port_range: "udp:9000-9100"
  pli_interval: 2
  nack_buffer_size: 200
  prefer_srflx_over_natted_host: false
  public_ip: ""
  enable_cors: true
  cors_origins:
    - "*"
  ice_servers:
    - urls:
        - "stun:stun.l.google.com:19302"

Option	Type	Default	Description
enable	bool	true	Whether to enable the WebRTC plugin
port_range	string	"udp:9000-9100"	ICE media ports: udp:A-B, tcp:A-B, or dual udp:A-B,tcp:C-D (default UDP-only)
pli_interval	u64	2	PLI (keyframe request) interval (seconds)
nack_buffer_size	usize	200	NACK retransmit history size; RFC 4588 RTX used when browser offers apt= (rustrtc ≥0.3.92)
prefer_srflx_over_natted_host	bool	false	Prefer srflx over private host in ICE check order
public_ip	string	""	Public IP rewritten into host candidates
enable_cors	bool	true	Whether to enable CORS
cors_origins	Vec	["*"]	Allowed cross-origin sources
ice_servers	Vec	[]	STUN/TURN list

Dual ICE (UDP + TCP)

When UDP is blocked, gather both UDP host and TCP passive candidates (opt-in):

webrtc:
  port_range: "udp:9000-9100,tcp:9101-9200"
  public_ip: "203.0.113.10"

Open 9000-9100/udp and 9101-9200/tcp on the firewall. TCP-only: port_range: "tcp:9101-9200".

ICE / TURN servers

ice_servers:
  - urls:
      - "stun:stun.l.google.com:19302"
  - urls:
      - "turn:turn.example.com:3478?transport=udp"
      - "turn:turn.example.com:3478?transport=tcp"
    username: "user"
    credential: "pass"

Minimal coturn sidecar (not embedded in Monibuca):

services:
  coturn:
    image: coturn/coturn:latest
    network_mode: host
    command:
      - "-n"
      - "--log-file=stdout"
      - "--listening-port=3478"
      - "--fingerprint"
      - "--lt-cred-mech"
      - "--user=user:pass"
      - "--realm=example.com"

Configure the same iceServers on the browser WHIP/WHEP client.

Full Compose reference stack (origin + coturn): repo deploy/compose/.

Tip

If the server has a public IP with no NAT, you can skip ICE servers and set public_ip. For restrictive networks prefer dual port_range and/or TURN.

Codec Support

Type	Codec	Description
Video	H.264	Widely supported
Video	H.265	Supported by some browsers
Audio	Opus	WebRTC standard audio codec

Note

WebRTC uses Opus audio encoding. When streams need to be forwarded to protocols such as RTMP/FLV/HLS, the engine will automatically transcode Opus to AAC (requires the transcode feature).

WHIP Publishing

API Endpoint

POST /webrtc/whip/{streamPath}
Content-Type: application/sdp

The request body is an SDP Offer, and the response returns an SDP Answer.

Publishing with FFmpeg

FFmpeg 6.1+ supports WHIP publishing:

ffmpeg -re -i input.mp4 \
  -c:v libx264 -preset ultrafast -tune zerolatency \
  -c:a libopus \
  -f whip "http://localhost:8080/webrtc/whip/live/test"

Publishing with OBS

OBS 30+ has built-in WHIP support:

1. Go to Settings → Stream
2. Select WHIP for the service
3. Enter the server URL: http://your-server:8080/webrtc/whip/live/test
4. Click Start Streaming

WHEP Subscribing

API Endpoint

POST /webrtc/whep/{streamPath}
Content-Type: application/sdp

The request body is an SDP Offer, and the response returns an SDP Answer.

Browser Subscribing Example

async function play(streamPath) {
  const pc = new RTCPeerConnection({
    iceServers: [{ urls: 'stun:stun.l.google.com:19302' }]
  });

  pc.addTransceiver('video', { direction: 'recvonly' });
  pc.addTransceiver('audio', { direction: 'recvonly' });

  pc.ontrack = (event) => {
    document.getElementById('player').srcObject = event.streams[0];
  };

  const offer = await pc.createOffer();
  await pc.setLocalDescription(offer);

  const response = await fetch(`/webrtc/whep/${streamPath}`, {
    method: 'POST',
    headers: { 'Content-Type': 'application/sdp' },
    body: offer.sdp,
  });

  const answer = await response.text();
  await pc.setRemoteDescription({ type: 'answer', sdp: answer });
}

play('live/test');

Built-in Test Pages

Monibuca provides built-in WebRTC test pages for quick feature verification:

Page	URL	Description
Publish Test	/webrtc/test/publish	Publish using camera/screen
Subscribe Test	/webrtc/test/subscribe	Play an existing WebRTC stream

Simply access the URLs in your browser to use them.

NAT Traversal

In network environments with NAT, proper configuration is required to ensure WebRTC connectivity:

Option 1: Configure Public IP

webrtc:
  public_ip: "203.0.113.1"  # Server's public IP

Option 2: Configure STUN/TURN Servers

webrtc:
  ice_servers:
    - urls:
        - "stun:stun.l.google.com:19302"
    - urls:
        - "turn:turn.example.com:3478"
      username: "user"
      credential: "pass"

Option 3: Port Mapping

Ensure the UDP port range configured in port_range is open in your firewall:

# Open UDP port range
sudo ufw allow 9000:9100/udp

Cross-Protocol Forwarding

After publishing via WebRTC, you can subscribe through other protocols:

# After WebRTC WHIP publishing...

# Subscribe via RTMP (Opus automatically transcoded to AAC)
ffplay rtmp://localhost:1935/live/test

# Subscribe via HTTP-FLV
ffplay http://localhost:8080/flv/live/test.flv

# Subscribe via HLS
ffplay http://localhost:8080/hls/live/test/index.m3u8