VoIP.info Troubleshooting

One-way VoIP audio

Trace the missing direction from advertised media address to receiving endpoint.

Reviewed 2026-09-07Asterisk 22 LTS · FreePBX 17 / Debian 12 checksIntermediateDocumentation based · not lab tested

Define the missing direction

Write down who cannot hear whom. If the customer cannot hear the office, follow audio sent by the office toward the customer. Avoid the ambiguous statement that the customer has no audio.

One-way audio: find the missing hop
First identify who cannot hear. If packets do not leave the sender, inspect the device or media source. If they leave but do not arrive, inspect SDP, NAT and firewall. If they arrive, inspect playback and decoding.

Follow one missing direction. Repeat on each PBX call leg and verify hold and transfer after a repair.

Most useful first checks

Compare an internal call, an external call and a call from another endpoint. A working internal call narrows the problem toward the path that changes at the carrier or remote-network boundary. Check mute and the selected headset before changing network settings.

Follow the media

Inspect the SDP address and port on each affected call leg, then compare actual RTP destinations. Determine whether packets leave the sender, reach the PBX and leave toward the receiver. Check NAT translation, firewall policy and whether direct media changes the expected path.

In Asterisk, inspect the loaded endpoint and topology-specific PJSIP NAT settings. In FreePBX, review the managed SIP/network settings through its supported interface. Do not paste standalone transport fragments over generated files.

Confirm the repair

Change only the setting supported by the evidence. Re-test both call directions, hold and transfer, because renegotiation can introduce a different media path. Keep the firewall enabled and remove temporary broad logging. A successful registration is not the acceptance criterion for this problem.

Watch the explanation

Detecting Packet Loss in RTP Phone Calls Using WiresharkPlaintext Packets · 13:43 · Conceptual

Compares an intact RTP capture with a copy containing deliberately removed packets. The stream list and detailed analysis show how gaps become loss indicators and how burst patterns differ from scattered loss.

Read applicability and editorial notes →
Watch this video here

A 2024 Wireshark demonstration; menu placement can change. The second capture is a teaching fixture with simulated loss. Review used the available transcript and primary references; audio and screen readability remain unverified.

The player loads only when you choose Watch here. Playback uses YouTube’s privacy-enhanced embed; YouTube processes playback data.

Sources & applicability

Primary references for the technical details above. Operational examples and planning checklists are VoIP.info editorial guidance.

Examples require adaptation to your topology. No live PBX or hardware testing is claimed.