VoIP.info Troubleshooting

VoIP call connects but has no audio

Separate endpoint audio selection from a media path that fails in both directions.

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

Confirm that the call was answered

A ringing indication or progress announcement is not the same as an established call. Identify the final signaling result, then confirm whether both participants have silence after answer. Ask whether the problem affects handset, speakerphone and headset equally.

Signaling and media answer different questions
SIP signaling passes from caller through call control to the other endpoint. RTP media can pass directly between endpoints or through a media relay.

Illustrative topology: direct RTP is shown below. Many PBXs and session border controllers anchor media instead.

Isolate endpoints first

Check mute, output selection and microphone permissions for softphones. Compare another endpoint on the same network and an internal call. If one headset fails while the handset works, changing carrier codecs is unlikely to help.

Inspect negotiation and delivery

Compare the accepted codec, media-security policy and SDP destinations. Look for RTP or SRTP on both relevant interfaces. Packets that reach the device with an incompatible security context require a different investigation from packets blocked before reaching the PBX.

On a PBX relaying media, examine both legs. An incoming stream at the carrier interface is not proof it was forwarded to the extension. Review recent firewall, VPN or direct-media changes.

Avoid a false success

Do not turn off encryption or the firewall permanently to produce sound. Correct the mismatched setting and repeat incoming, outgoing and transferred calls under the intended policy. Record which boundary was responsible so later network changes can be reviewed against it.

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.