WebRTC Architecture — Browser-Based Real-Time Communication

WebRTC: The Browser Becomes a Phone

WebRTC turns web browsers into VoIP endpoints without plugins. iIt's how Telnyx's WebRTC SDK works, enabling voice calls directly from browser-based applications.

WebRTC APIs

Three JavaScript APIs provide the functionality:

getUserMedia()

Captures local audio/video:

navigator.mediaDevices.getUserMedia({
  audio: true,
  video: false
}).then(stream => {
  // Attach to audio element
  localAudio.srcObject = stream;
});

Permissions: Browser prompts user for microphone access. HTTPS required for production.

RTCPeerConnection

The core of WebRTC — handles:

• ICE candidate gathering
• DTLS handshake for SRTP keys
• Media encoding/decoding (Opus mandatory)
• Network adaptation (bandwidth estimation)

const pc = new RTCPeerConnection({
  iceServers: [
    { urls: 'stun:stun.telnyx.com:3478' },
    { urls: 'turn:turn.telnyx.com:3478', credential: '...' }
  ]
});

// Add local stream
stream.getTracks().forEach(track => {
  pc.addTrack(track, stream);
});

RTCDataChannel

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