SIP behind NAT is one of the most common sources of VoIP call quality issues. This guide explains the protocols involved and how to configure your network for reliable one-way or no-audio resolution.
Understanding SIP and NAT
SIP signalling carries IP addresses in the message body (SDP). When you are behind NAT, the private IP address in SDP does not match the public IP the server sees, causing audio to be directed to the wrong address.
STUN Configuration
STUN (Session Traversal Utilities for NAT) allows the VoIP client to discover its public IP address. In iTel Dialer, set the STUN server to stun.itelvoip.com:3478 in Settings → Network → STUN Server.
Router/Firewall Rules
Open the following ports on your router/firewall:
- UDP 5060 — SIP signalling
- UDP 5061 — SIP TLS (if using TLS transport)
- UDP 10000-20000 — RTP media streams
SIP ALG (Application Layer Gateway)
Most consumer routers have SIP ALG enabled by default. This feature often corrupts SIP headers and causes one-way audio. Disable SIP ALG on your router — this is the most common fix for audio issues.
Testing
After configuration, use the built-in Echo Test number (*47 in iTel Dialer) to verify bidirectional audio. A successful echo test confirms correct NAT traversal.