One-way audio

One-way audio is exactly what it sounds like: one person on the call can be heard, but cannot hear the other side. The call connects, both parties think it worked, and then one of them is talking into silence. It is one of the most common and most confusing voice problems, because the signaling looks completely healthy — the trouble is entirely in the media path.

To understand it, separate two things. The SIP (Session Initiation Protocol) signaling sets up the call, and a separate stream carries the actual voice using the Real-time Transport Protocol, or RTP. One-way audio means the RTP packets are flowing in one direction but not the other. The most frequent cause is a firewall or router that maps internal addresses, a situation handled by NAT traversal. If that is misconfigured, audio leaves your network but the return stream gets dropped at the edge.

To diagnose, first take a SIP trace to confirm the signaling really did succeed, then look at where the RTP is supposed to go. Check that the public address advertised in the call matches the address that can actually receive packets. Mismatched Codec settings can also cause silence, and heavy Packet loss can make one direction sound like it cut out entirely.

The practical fix is usually about addresses and ports. Make sure your server knows its real public IP, that the RTP port range is open both ways in the firewall, and that any NAT settings reflect your actual network. Get those right and one-way audio almost always disappears, because the return voice stream finally has a path home.