Usage Guide

This guide covers local development, container deployment, API usage, and troubleshooting for live-webrtc-go.

Usage Guide

Requirements

Dependency	Version	Notes
Go	1.22+	Required for compilation
Docker	20.10+	Optional, for containerized deployment
Docker Compose	2.0+	Optional, for multi-container orchestration
Browser	Chrome 90+ / Firefox 88+	WebRTC support required

WebRTC Port Requirements

When running behind NAT, you need to:

Configure STUN/TURN servers
Ensure UDP ports are not blocked by firewalls

Local Development

Method 1: Direct Execution

# Clone the repository
git clone https://github.com/LessUp/go-live.git
cd go-live

# Download dependencies
go mod tidy

# Run the server
go run ./cmd/server

Method 2: Using Development Script (Recommended)

# Basic startup (loads .env.local if exists)
./scripts/start.sh

# Run go mod tidy before starting
RUN_TIDY=1 ./scripts/start.sh

Script features:

Creates records, .gocache, .gomodcache directories
Sets GOCACHE and GOMODCACHE environment variables
Loads .env.local file (if exists)
Starts the server

Environment Variables File

Create .env.local file:

# Server configuration
HTTP_ADDR=:8080
ALLOWED_ORIGIN=*

# Authentication
AUTH_TOKEN=your-secret-token
# ROOM_TOKENS=room1:token1;room2:token2
# JWT_SECRET=jwt-signing-secret

# WebRTC configuration
# STUN_URLS=stun:stun.l.google.com:19302
# TURN_URLS=turn:turn.example.com:3478
# TURN_USERNAME=username
# TURN_PASSWORD=password

# Recording
RECORD_ENABLED=1
RECORD_DIR=records

# Rate limiting
RATE_LIMIT_RPS=10
RATE_LIMIT_BURST=20

Verification After Startup

# Health check
curl http://localhost:8080/healthz
# Output: ok

# List rooms
curl http://localhost:8080/api/rooms
# Output: []

# Get frontend configuration
curl http://localhost:8080/api/bootstrap | jq

Configuration Reference

Core Settings

Variable	Default	Description
`HTTP_ADDR`	`:8080`	HTTP listen address, format `host:port`
`ALLOWED_ORIGIN`	`*`	CORS allowed origin, `*` means any

Authentication

Variable	Format	Description
`AUTH_TOKEN`	string	Global access token
`ROOM_TOKENS`	`room1:tok1;room2:tok2`	Per-room token mapping
`JWT_SECRET`	string	JWT HMAC signing key
`JWT_AUDIENCE`	string	Required JWT audience claim
`ADMIN_TOKEN`	string	Admin API access token

Authentication Priority:

Room-specific Token → 2. Global Token → 3. JWT → 4. No authentication

WebRTC/ICE Configuration

Variable	Default	Description
`STUN_URLS`	`stun:stun.l.google.com:19302`	STUN server list
`TURN_URLS`	-	TURN server list
`TURN_USERNAME`	-	TURN username
`TURN_PASSWORD`	-	TURN password

Example:

STUN_URLS=stun:stun1.l.google.com:19302,stun:stun2.l.google.com:19302
TURN_URLS=turn:turn.example.com:3478,turns:turn.example.com:5349
TURN_USERNAME=myuser
TURN_PASSWORD=mypassword

Recording Configuration

Variable	Default	Description
`RECORD_ENABLED`	`0`	Enable recording, set to `1` to enable
`RECORD_DIR`	`records`	Recording file storage directory

S3 Upload Configuration

Variable	Default	Description
`UPLOAD_RECORDINGS`	`0`	Enable upload, set to `1` to enable
`DELETE_RECORDING_AFTER_UPLOAD`	`0`	Delete local file after upload
`S3_ENDPOINT`	-	S3/MinIO endpoint address
`S3_REGION`	-	S3 region
`S3_BUCKET`	-	Target bucket name
`S3_ACCESS_KEY`	-	Access Key ID
`S3_SECRET_KEY`	-	Secret Access Key
`S3_USE_SSL`	`1`	Use HTTPS connection
`S3_PATH_STYLE`	`0`	Use path-style addressing
`S3_PREFIX`	-	Object key prefix

MinIO Example:

S3_ENDPOINT=minio.example.com:9000
S3_ACCESS_KEY=minioadmin
S3_SECRET_KEY=minioadmin
S3_BUCKET=recordings
S3_USE_SSL=0
S3_PATH_STYLE=1

Rate Limiting Configuration

Variable	Default	Description
`RATE_LIMIT_RPS`	`0`	Requests per second per IP, `0` disables
`RATE_LIMIT_BURST`	`0`	Burst capacity

TLS Configuration

Variable	Description
`TLS_CERT_FILE`	TLS certificate file path
`TLS_KEY_FILE`	TLS private key file path

TLS_CERT_FILE=/etc/ssl/cert.pem
TLS_KEY_FILE=/etc/ssl/key.pem

Debug Configuration

Variable	Default	Description
`PPROF`	`0`	Enable pprof endpoints
`OTEL_SERVICE_NAME`	`live-webrtc-go`	OpenTelemetry service name

Docker Deployment

Build Image

docker build -t live-webrtc-go:latest .

Basic Run

docker run --rm -p 8080:8080 live-webrtc-go:latest

Enable Recording

docker run --rm -p 8080:8080 \
  -e RECORD_ENABLED=1 \
  -e RECORD_DIR=/records \
  -v $(pwd)/records:/records \
  live-webrtc-go:latest

Full Configuration Example

docker run --rm -p 8080:8080 \
  -e HTTP_ADDR=:8080 \
  -e AUTH_TOKEN=mytoken \
  -e RECORD_ENABLED=1 \
  -e RECORD_DIR=/records \
  -e UPLOAD_RECORDINGS=1 \
  -e S3_ENDPOINT=s3.amazonaws.com \
  -e S3_BUCKET=my-bucket \
  -e S3_ACCESS_KEY=$AWS_ACCESS_KEY_ID \
  -e S3_SECRET_KEY=$AWS_SECRET_ACCESS_KEY \
  -v $(pwd)/records:/records \
  live-webrtc-go:latest

Docker Compose

docker-compose.yml:

version: '3.8'

services:
  live-webrtc:
    build: .
    ports:
      - "8080:8080"
    environment:
      - HTTP_ADDR=:8080
      - AUTH_TOKEN=${AUTH_TOKEN}
      - RECORD_ENABLED=1
      - RECORD_DIR=/records
      - RATE_LIMIT_RPS=10
      - RATE_LIMIT_BURST=20
    volumes:
      - ./records:/records
    restart: unless-stopped

Start:

docker compose up -d

Kubernetes Deployment

Deployment Example

apiVersion: apps/v1
kind: Deployment
metadata:
  name: live-webrtc
spec:
  replicas: 3
  selector:
    matchLabels:
      app: live-webrtc
  template:
    metadata:
      labels:
        app: live-webrtc
    spec:
      containers:
      - name: live-webrtc
        image: live-webrtc:latest
        ports:
        - containerPort: 8080
        env:
        - name: HTTP_ADDR
          value: ":8080"
        - name: AUTH_TOKEN
          valueFrom:
            secretKeyRef:
              name: live-webrtc-secret
              key: auth-token
        volumeMounts:
        - name: records
          mountPath: /records
      volumes:
      - name: records
        persistentVolumeClaim:
          claimName: records-pvc

Service Example

apiVersion: v1
kind: Service
metadata:
  name: live-webrtc
spec:
  selector:
    app: live-webrtc
  ports:
  - port: 80
    targetPort: 8080
  type: LoadBalancer

API Usage Examples

Using curl

# Set variables
HOST=http://localhost:8080
TOKEN=your-token

# Health check
curl $HOST/healthz

# Get frontend configuration
curl $HOST/api/bootstrap | jq

# List rooms
curl $HOST/api/rooms | jq

# List recording files
curl $HOST/api/records | jq

# Close room (admin API)
curl -X POST \
  -H "Authorization: Bearer $TOKEN" \
  $HOST/api/admin/rooms/myroom/close

Stream with OBS

Open OBS → Settings → Stream
Select “WHIP” as the service
Server URL: http://localhost:8080/api/whip/publish/myroom
Bearer Token: your-token (if authentication is configured)
Start streaming

Play with JavaScript

// Get configuration
const config = await fetch('/api/bootstrap').then(r => r.json());

// Create PeerConnection
const pc = new RTCPeerConnection({
  iceServers: config.iceServers
});

// Create transceivers (receive only)
pc.addTransceiver('video', { direction: 'recvonly' });
pc.addTransceiver('audio', { direction: 'recvonly' });

// Handle remote tracks
pc.ontrack = (event) => {
  const video = document.getElementById('video');
  video.srcObject = event.streams[0];
};

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

// Send WHIP request
const response = await fetch('/api/whep/play/myroom', {
  method: 'POST',
  headers: {
    'Content-Type': 'application/sdp',
    'Authorization': 'Bearer your-token'
  },
  body: offer.sdp
});

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

Frontend Integration

Bootstrap API Response

{
  "authEnabled": true,
  "recordEnabled": true,
  "iceServers": [
    {
      "urls": ["stun:stun.l.google.com:19302"]
    }
  ],
  "features": {
    "rooms": true,
    "records": true
  }
}

Authentication Header Format

Two formats supported:

Authorization: Bearer <token>
X-Auth-Token: <token>

Troubleshooting

Common Issues

Issue	Possible Cause	Solution
`publisher already exists`	Room already has a publisher	Use a different room name or wait for publisher to disconnect
`unauthorized`	Authentication failed	Check token or JWT configuration
`too many requests`	Rate limit triggered	Increase `RATE_LIMIT_BURST` or wait
`room not found`	Room does not exist	Ensure publisher is connected
`ICE connection failed`	NAT traversal issue	Configure TURN server
No video/audio	Codec mismatch	Check browser-supported codecs

Debugging Steps

Check Service Status

curl http://localhost:8080/healthz
curl http://localhost:8080/api/rooms

View Metrics
```
curl http://localhost:8080/metrics
```

Enable pprof

PPROF=1 go run ./cmd/server
# Access http://localhost:8080/debug/pprof/

Check Logs
- View console slog output
- Focus on ERROR level logs

WebRTC Connection Issues

Symptom: Browser shows “ICE connection failed”

Troubleshooting:

Check STUN server reachability
Configure TURN server (for NAT environments)
Confirm UDP ports are not blocked

TURN Configuration Example:

TURN_URLS=turn:turn.example.com:3478
TURN_USERNAME=username
TURN_PASSWORD=password

Authentication Issues

Symptom: API returns 401 Unauthorized

Troubleshooting:

Confirm correct Authorization header is sent
Check token is correct
If using JWT, verify signature and expiration

Recording Issues

Symptom: Recording directory is empty

Troubleshooting:

Confirm RECORD_ENABLED=1
Check directory permissions: ls -la records/
Check logs for write errors
Confirm publisher is using VP8/VP9/Opus codecs

Upload Issues

Symptom: Files not uploaded to S3

Troubleshooting:

Confirm UPLOAD_RECORDINGS=1
Check all S3_* variables are configured
Verify S3 credentials are valid
Check network connectivity

# Test S3 connection
aws s3 ls --endpoint-url http://minio:9000

Performance Issues

Symptom: High latency, stuttering

Troubleshooting:

Check server CPU/memory usage
Check subscriber count
Consider setting MAX_SUBS_PER_ROOM limit
Check network bandwidth

# View metrics
curl http://localhost:8080/metrics | grep live_

Development Commands Reference

# Build
make build

# Format
make fmt

# Lint
make lint

# Security scan
make security

# Run tests
make test          # Unit + Integration + Security
make test-all      # Include e2e + Performance
make test-unit     # Unit tests only

# Coverage
make coverage
open coverage.html

Design Documentation - Architecture and module details
API Reference - Complete API documentation
GitHub Repository - Source code

Usage Guide

Table of Contents

Requirements

WebRTC Port Requirements

Local Development

Method 1: Direct Execution

Method 2: Using Development Script (Recommended)

Environment Variables File

Verification After Startup

Configuration Reference

Core Settings

Authentication

WebRTC/ICE Configuration

Recording Configuration

S3 Upload Configuration

Rate Limiting Configuration

TLS Configuration

Debug Configuration

Docker Deployment

Build Image

Basic Run

Enable Recording

Full Configuration Example

Docker Compose

Kubernetes Deployment

Deployment Example

Service Example

API Usage Examples

Using curl

Stream with OBS

Play with JavaScript

Frontend Integration

Bootstrap API Response

Authentication Header Format

Troubleshooting

Common Issues

Debugging Steps

WebRTC Connection Issues

Authentication Issues

Recording Issues

Upload Issues

Performance Issues

Development Commands Reference

Related Documentation