What the test checks
- Public IPv4 / IPv6 — the address the internet sees, via ipify.
- UDP / STUN — your browser asks two public STUN servers (Google and Cloudflare) "what address do you see me from?" over UDP. A reply proves outbound UDP works.
- NAT behavior — if the same local port is mapped to different public ports for different servers, the NAT is likely symmetric, which some VoIP setups struggle with.
Why one-way audio happens
SIP tells the other end where to send audio by writing an IP address and port into the call setup. Behind NAT, that's a private address the internet can't reach. Phone systems solve this with STUN, session border controllers, or keep-alives. Problems start when:
- SIP ALG on the router rewrites SIP messages and gets it wrong. Most hosted VoIP providers tell customers to turn it off.
- The firewall's UDP timeout is shorter than the phone's keep-alive or re-registration interval, so incoming calls stop ringing or calls drop at a fixed time.
- The firewall blocks the RTP port range the provider uses.
- There are two routers in a row (double NAT).
Ports VoIP commonly uses
| Traffic | Common default |
|---|---|
| SIP signaling | 5060 UDP/TCP, 5061 TLS (RFC 3261) |
| RTP voice | A UDP range chosen by the vendor — ranges such as 10000–20000 are common; check your provider's documentation |
| STUN | 3478 UDP (19302 on Google's servers) |
Frequently asked questions
What causes one-way audio on VoIP calls?
Usually NAT or firewall problems: SIP ALG on the router rewriting call setup incorrectly, a firewall blocking the RTP voice ports, double NAT, or the phone advertising a private IP address. Turning off SIP ALG and confirming the provider's RTP port range is allowed fixes most cases.
What is SIP ALG and should I turn it off?
SIP ALG (Application Layer Gateway) is a router feature that tries to rewrite SIP messages to help them pass through NAT. Many implementations do it badly, causing one-way audio, failed transfers and registration drops. Most hosted VoIP providers recommend turning it off unless they say otherwise.
What is STUN in VoIP?
STUN (Session Traversal Utilities for NAT, RFC 8489) lets a device behind NAT ask a public server which public IP address and port it appears to come from. VoIP and WebRTC use that answer to tell the other end where to send audio.
What is symmetric NAT?
A symmetric NAT assigns a different public port for each destination a device talks to. That breaks the assumption that the address learned from a STUN server will also work for media from the other party, so calls may need a relay (TURN) or session border controller.
Do I need a static IP for VoIP?
Most hosted VoIP phones don't need one. SIP trunks to an on-site phone system often require a static IP so the carrier can authenticate and route calls to your address. Ask your provider which authentication method it uses.
Why do my VoIP calls drop after exactly 15 or 30 minutes?
A fixed-interval drop is usually a timer, not the network: a SIP session timer that isn't refreshed, or a firewall's UDP/NAT timeout closing the connection. Adjusting the firewall's UDP timeout or the phone's keep-alive interval typically fixes it.
What is double NAT and does it affect VoIP?
Double NAT happens when two routers each perform address translation, such as an ISP gateway plus your own router. It makes NAT traversal harder and can cause one-way audio. Put the ISP device in bridge mode or put your router in its DMZ/passthrough setting.
Sources & references
- IETF RFC 8489 — STUN
- IETF RFC 3261 — SIP: Session Initiation Protocol
- IETF RFC 3550 — RTP: A Transport Protocol for Real-Time Applications
Written by Eric · Reviewed November 2026