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VoIP & telephony

NAT type test for VoIP

Find out what NAT your router or ISP uses: a regular one or symmetric NAT, the usual cause of one-way audio. Shows your public IP and port, whether UDP passes, signs of CGNAT, and what to set in Asterisk and FreeSWITCH.

Your browser sends STUN requests over UDP to public Cloudflare and Google servers: a few tiny packets, no call and no recording. Test from the network where the phone sits.

Why NAT breaks calls

SIP carries the address and port for voice inside its messages (in SDP). Behind NAT the phone only knows its private address, so the PBX answers into the void: one-way audio or silence. A regular NAT keeps the same public port for every destination, so the PBX just needs to reply where the packet came from. A symmetric NAT gives each destination a new port; STUN cannot help, you need symmetric RTP on the PBX, an SBC or TURN.

How the test works

From one and the same socket the browser asks four STUN servers which address and port they see. If they all report the same address:port, the NAT does not depend on the destination. If the ports differ, it is symmetric.

What else to check

VoIP quality test — loss and jitter; SDP analyzer — which address the phone put in SDP; pcap analyzer — whether RTP arrived.