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Voice infrastructure

NAT (Network Address Translation)

What is NAT (Network Address Translation)?

NAT (Network Address Translation) is the technique a router or firewall uses to let many devices on a private network share one public IP address, rewriting the source address and port on outbound packets and reversing the change on the way back. It's what lets an office share one internet connection, but it also hides each device's real address — which is why a WebRTC client or SIP phone behind NAT needs STUN, TURN, and ICE to find a reachable public address before a real-time call's media can flow.

More detail

NAT rewrites addresses transparently for ordinary web traffic, but it breaks the private address a SIP or WebRTC endpoint embeds inside its own signaling — a remote party that tries to send media to that embedded address hits an unroutable private IP. Real-time voice and video work around this with STUN (discover a usable public address), TURN (relay media when no direct path exists), and ICE (negotiate whichever route actually works).

An SBC (Session Border Controller) performs carrier-grade NAT traversal for SIP trunks at the network edge, the same job STUN/TURN/ICE do for a browser-based WebRTC session — both exist because NAT alone isn't enough for two-way, real-time media to connect reliably.

Frequently asked

Why does NAT cause problems for VoIP and video calls?
NAT only rewrites the address in a packet's header, not the private address a VoIP or WebRTC endpoint embeds inside its own SIP/SDP signaling. A remote party trying to reach that embedded address hits a private IP it can't route to, so real-time voice and video need STUN, TURN, and ICE to discover and negotiate an address that's actually reachable.
Is NAT the same thing as a firewall?
No. NAT's job is address sharing and translation; a firewall's job is filtering traffic by rule. Many routers do both, but a device can perform NAT without any firewall policy, and vice versa.

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