Dananeer TURN Server - WebRTC Configuration

📡 What is a TURN Server?

A TURN (Traversal Using Relays around NAT) server is a critical component for WebRTC applications. It acts as a relay server that forwards media traffic between peers when direct peer-to-peer connections are not possible due to network restrictions.

When is TURN needed?

4G/LTE mobile networks (carrier restrictions)
5G networks with strict NAT
Corporate firewalls
Symmetric NAT environments
Restrictive network policies

How it works: When WebRTC cannot establish a direct connection, it automatically falls back to using the TURN server as an intermediary, ensuring reliable communication across all network types.

⚙️ Server Configuration

Domain

turn.dananeer.io

IP Address

165.22.71.105

Ports

UDP: 3478 - For fast LAN/WiFi connections
TCP: 3478 - For moderate firewalls
TLS: 443 - For 4G/5G and restrictive networks

Transport Types Supported

UDP (User Datagram Protocol) - Fastest, lowest latency
TCP (Transmission Control Protocol) - More reliable
TLS (Transport Layer Security) - Most secure, works everywhere

🔧 Backend Configuration

Add these environment variables to your backend .env file:

# STUN servers (optional - defaults to Google STUN servers)
RTC_STUN_URLS=stun:stun.l.google.com:19302,stun:stun1.l.google.com:19302

# TURN servers - Include all three transport types for maximum compatibility
# Supports: 4G, 5G, Firewalls, Symmetric NAT, Corporate Networks
RTC_TURN_URLS=turn:turn.dananeer.io:3478?transport=udp,turn:turn.dananeer.io:3478?transport=tcp,turns:turn.dananeer.io:443?transport=tcp

# TURN server authentication (required if TURN_URLS is set)
RTC_TURN_USERNAME=dananeer_turn
RTC_TURN_CREDENTIAL=rZBSJ0z2mrDU4ja6j9DT5yPG

# Maximum participants per call (optional, default: 4)
RTC_MAX_PARTICIPANTS=4

📋 Configuration Details

RTC_STUN_URLS Optional

STUN (Session Traversal Utilities for NAT) servers for discovering public IP addresses.

Default: Google STUN servers are used automatically if not specified.

RTC_TURN_URLS Optional

TURN server URLs with transport types. Include all three for maximum compatibility:

turn:turn.dananeer.io:3478?transport=udp - For LAN/WiFi
turn:turn.dananeer.io:3478?transport=tcp - For moderate firewalls
turns:turn.dananeer.io:443?transport=tcp - For 4G/5G and restrictive networks

Default: Empty array (no TURN servers). Works for LAN/WiFi only.

RTC_TURN_USERNAME Required if TURN_URLS set

TURN server username for authentication.

Current value: dananeer_turn

RTC_TURN_CREDENTIAL Required if TURN_URLS set

TURN server password/credential for authentication.

Note: Keep this secure and never commit to version control.

RTC_MAX_PARTICIPANTS Optional

Maximum number of participants allowed in a WebRTC call.

Default: 4

Range: 2-10 (enforced automatically)

🌐 Network Support

This TURN server configuration supports all network types:

✅ LAN Networks UDP transport

✅ WiFi Networks UDP/TCP transport

✅ 4G/LTE Networks TLS transport

✅ 5G Networks TLS transport

✅ Restrictive Firewalls TLS transport

✅ Symmetric NAT TURN relay

✅ Corporate Networks TLS transport

🔌 API Endpoint

Your backend exposes the RTC configuration via HTTP endpoint:

GET /api/v1/rtc/config
Authorization: Bearer <your_token>

Response:

{
  "turnUrls": [
    "turn:turn.dananeer.io:3478?transport=udp",
    "turn:turn.dananeer.io:3478?transport=tcp",
    "turns:turn.dananeer.io:443?transport=tcp"
  ],
  "turnUsername": "dananeer_turn",
  "turnCredential": "rZBSJ0z2mrDU4ja6j9DT5yPG",
  "stunUrls": [
    "stun:stun.l.google.com:19302",
    "stun:stun1.l.google.com:19302"
  ],
  "maxParticipants": 4
}

💻 Frontend Usage

In your frontend WebRTC code, fetch the configuration and use it:

const config = await fetch('/api/v1/rtc/config', {
  headers: { Authorization: `Bearer ${token}` }
});

const pc = new RTCPeerConnection({
  iceServers: [
    ...config.stunUrls.map(url => ({ urls: url })),
    ...config.turnUrls.map(url => ({
      urls: url,
      username: config.turnUsername,
      credential: config.turnCredential
    }))
  ]
});

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