Webrtc Implementation

WebRTC Implementation

Implementing WebRTC for real-time audio, video, and data communication — from signaling and peer connection through NAT traversal, media streams, and data channels.

When to Use

• Building video/audio conferencing applications
• Implementing peer-to-peer file transfer
• Adding real-time data channels to web applications
• Building screen sharing or remote desktop tools
• Creating real-time collaboration features

WebRTC Connection Flow

WEBRTC_FLOW = {
    'signaling': 'Peers exchange session metadata (SDP offers/answers) via server',
    'ICE': 'Interactive Connectivity Establishment — finds best network path',
    'STUN': 'Session Traversal Utilities for NAT — determines public IP/port',
    'TURN': 'Traversal Using Relays around NAT — relay when P2P fails',
    'DTLS': 'Datagram Transport Layer Security — encrypts all media/data',
}

class SignalingServer:
    """WebSocket-based signaling server for WebRTC."""
    def __init__(self):
        self.clients = {}  # client_id -> websocket
    
    def relay_sdp(self, sender: str, receiver: str, sdp: dict):
        """Relay SDP offer/answer between peers."""
        if receiver in self.clients:
            self.clients[receiver].send_json({
                'type': 'sdp',
                'from': sender,
                'sdp': sdp
            })
    
    def relay_ice(self, sender: str, receiver: str, candidate: dict):
        """Relay ICE candidates between peers."""
        if receiver in self.clients:
            self.clients[receiver].send_json({
                'type': 'ice_candidate',
                'from': sender,
                'candidate': candidate
            })

Common Pitfalls

1. TURN server not configured — only STUN works for 80% of connections; TURN needed for the rest
2. Signaling bottleneck — signaling must be fast (WebSocket); slow signaling breaks connections
3. No connection state handling — WebRTC connections change state; handle all states (disconnected, failed)
4. Codec compatibility — not all browsers support all codecs; check in advance (H.264, VP8, VP9, AV1)
5. Bandwidth estimation — sending HD video on slow connections causes buffering; use SVC or simulcast

Verification Checklist

• STUN server configured (Google's stun.l.google.com or custom)
• TURN server configured for NAT traversal fallback
• Signaling via WebSocket (reliable, low-latency)
• ICE candidate gathering and exchange working
• Media stream constraints (resolution, frame rate) configured
• Data channel (if used) established and reliable/unreliable mode chosen
• Connection state monitoring (oniceconnectionstatechange)
• Codec negotiation (H.264/VP8 for compatibility, AV1 for quality)
• Bandwidth estimation (cc.bitrate) and adaptation