One-Way Audio

One-way audio is one of the most common VoIP complaints. The call connects, the timer runs, and one person can hear perfectly while the other hears silence. Because the signaling worked, the problem is not with SIP itself. It lies with the RTP media, which travels on separate UDP ports and can be blocked independently in each direction.

The usual cause is NAT. The client behind a router advertises its private IP address in the SDP of the INVITE. The far end dutifully sends audio to that private address, which is unreachable from outside, so nothing arrives. Meanwhile the client sends its own audio to the far end's public address, which works. The result is that the far end hears the client but not the reverse. A firewall that allows outbound UDP but blocks the inbound reply produces the same symptom.

Fixes work at both ends. On the client, enable STUN so it learns its public address and ICE so it can test candidate paths, as RFC 8445 describes. On the server, enable NAT handling so the PBX sends audio back to the address the packets actually came from rather than the advertised one; in Asterisk this is nat=yes for chan_sip or rtp_symmetric=yes and force_rport=yes for PJSIP. Disable SIP ALG on the router. A VPN sidesteps the issue entirely.

In SIPdia, turn on the STUN server and ICE options under Advanced as the first step.