WebRTC Protocol

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 (ICE/DTLS/SRTP)
Publish	✅ Supported (WHIP)
Subscribe	✅ Supported (WHEP)
Latency	< 500ms
Feature Flag	webrtc

Configuration

Feature Activation

Cargo.toml

[features]
webrtc = ["dep:plugin-webrtc"]

Configuration File

webrtc:
  enable: true
  port_range: "udp:9000-9100"
  pli_interval: 2
  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"	UDP port range
pli_interval	u64	2	PLI (keyframe request) interval (seconds)
public_ip	string	""	Public IP (used for NAT traversal)
enable_cors	bool	true	Whether to enable CORS
cors_origins	Vec	["*"]	Allowed cross-origin sources
ice_servers	Vec	[]	ICE server list

ICE Server Configuration

ice_servers:
  # STUN server (for NAT traversal)
  - urls:
      - "stun:stun.l.google.com:19302"

  # TURN server (for relay)
  - urls:
      - "turn:turn.example.com:3478"
    username: "user"
    credential: "pass"

Tip

If the server has a public IP with no NAT, you can skip configuring ICE servers. Simply set public_ip to the server’s public IP address.

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