NAT and STUN in VoIP
Short answer: NAT (network address translation) allows many devices to share a single public IP address, while STUN (Session Traversal Utilities for NAT) is the protocol that reports back which public IP and port a device actually appears from. SIP clients rely on STUN so the far side can send audio to an address that is actually reachable.
Most home and office routers run NAT, which explains why one-way audio and dropped calls are such a common first-time problem with a new softphone.
How it works
SIP messages carry the IP address and port where media is supposed to go. Behind NAT, a device only knows its own private address, so the far end's RTP audio can end up going nowhere useful. A STUN server (RFC 8489) reflects back the public mapping, and the client advertises that instead. Keep-alives, rport and ICE all help keep that mapping open over time.
NAT traversal, the short version
| Item | What it does |
|---|---|
| STUN server (RFC 8489) | Mirrors back the public address and port so the client knows what to announce |
| Keep-alives, rport, and ICE | Keep the NAT mapping from closing |
| One-way audio issue | Turn on STUN or NAT traversal inside the SIP client |
| Browser web dialer | Requires no setup on the client side at all |
How it applies to SMS Conecta
Registering a SMS Conecta number in Zoiper, Linphone or MicroSIP behind a router? Turn on STUN or NAT traversal in the client if audio is only working one way. Registration goes through sip.smsconecta.com on UDP/TCP port 5060 or TLS port 5061. The browser web dialer skips all of this client-side setup entirely.
Related terms
- SIP
- Softphone
- G.711 codec
- Web dialer
Frequently asked questions
What typically causes one-way audio?
Most often NAT, or a firewall blocking RTP; turning on STUN or switching to TCP/TLS tends to fix it.
Are STUN and TURN the same thing?
No — TURN relays media through a server, whereas STUN only figures out your public address.
Is turning off SIP ALG on my router a good idea?
Often, yes. SIP ALG frequently rewrites packets incorrectly, which is why so many troubleshooting guides suggest disabling it.
Related pages
- SIP
- softphone
- web dialer
- SIP troubleshooting
- Zoiper integration
- Linphone integration
- Mobile Phone Number
- Non-VoIP Number
Sources
- IETF: RFC 8489, Session Traversal Utilities for NAT (STUN) (as of September 2026)
- IETF: RFC 3581, Symmetric Response Routing (rport) for SIP (as of September 2026)