VoIP.info Troubleshooting

VoIP call connects but has no audio

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

Reviewed 2026-10-30Asterisk 22 LTS · FreePBX 17 / Debian 12 checksIntermediateDocumentation based · not lab tested

Confirm that the call was answered

A ringing indication, ringback or progress announcement is not the same as an established call. Record whether silence occurs before answer, after a final 2xx response and ACK, or only after hold, transfer or another renegotiation. Then state whether both participants hear silence or one direction is missing. Early-media failure, no audio after answer and one-way audio require different evidence paths.

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, handset, speaker, output selection, input selection and operating-system microphone permissions. 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. If every local audio path fails, retain the endpoint as a suspect until a known-good endpoint reproduces the same call-path symptom.

Inspect negotiation, both legs and both directions

Read the final SDP offer and answer for each leg. Compare accepted codecs, payload mappings, send or receive direction, media address and port, and RTP versus SRTP policy. A negotiated codec name alone is not proof that decodable media arrived. Packets reaching a device with an incompatible security context require a different investigation from packets blocked before the PBX.

On a PBX relaying media, examine carrier-to-PBX, PBX-to-endpoint, endpoint-to-PBX and PBX-to-carrier observations separately. An incoming carrier stream is not proof it was bridged to the extension. If Asterisk is intended to stay in the media path, confirm whether direct_media changed that topology. Review recent firewall, NAT, VPN, codec, certificate or media-policy changes without assuming that the newest change is causal. For FreePBX PJSIP, the managed media-path investigation adds the required comparison among GUI values, generated configuration, loaded objects, final SDP and observed RTP on both legs.

Interpret packet evidence carefully

RTP sequence numbers and timestamps can expose continuity and timing, while SSRC values help distinguish sources. With SRTP, a capture can still show endpoints, ports, packet timing and many RTP headers, but it does not prove that either endpoint possessed the correct key or produced audible output. An empty capture does not prove that the network carried no packets until the interface, mirror direction, filter, direct-media path and capture-drop counters are checked.

Avoid a false success

Do not disable encryption, authentication or the firewall as the production repair. Correct the specific negotiation, route, NAT or endpoint-audio fault and repeat incoming, outgoing, hold, resume and transferred calls under the intended policy. Record the failing boundary and the exact evidence that changed. A decoded stream, answered dialog and user-confirmed bidirectional audio remain separate acceptance results.

Use the WebRTC media path when Asterisk registration succeeds

For a registered browser endpoint on Asterisk, use the WebRTC-specific diagnostic to distinguish WSS and SIP state from ICE selection, DTLS-SRTP establishment, browser-facing media counters and playback.

When a Zen trunk call answers in silence

The Zen no-audio diagnostic follows the answered dialog through SDP, B2BUA media addresses, both RTP directions and G.711 decoding without treating registration as media proof.

Verify Baresip audio modules and devices before codec changes

The Baresip registration and audio workflow keeps driver-module loading, global and account device selection, SDP, NAT traversal and RTP as separate evidence.

Branch to Linphone ICE only when the media path requires it

When the affected endpoint is Linphone and the evidence shows missing candidates, failed checks or an unexpected direct-versus-relay path, use the Linphone ICE diagnostic. A completed registration or configured server is not a media result.

Keep VOICE frames, RTP packets and audible output distinct

The Asterisk frame-trace diagnostic adds a channel-internal checkpoint without treating one VOICE frame as proof of continuous RTP, decoding or audible speech.

Map a silent VoIPstudio call beyond the green icon

A green application indicator and registered endpoint do not prove negotiated or rendered media. The VoIPstudio Desktop diagnostic follows SDP, both RTP directions, operating-system permission and selected call output independently.

Compare VoIP Much codec claims with the negotiated call

VoIP Much’s current product and codec pages do not list exactly the same codecs. The provider diagnostic therefore preserves SDP, packetization and both RTP directions rather than enabling every published codec.

Compare VoIPline SDP with the ordered service

The VoIPline diagnostic preserves the negotiated codec, packetization and both RTP directions rather than converting a published codec list into media proof.

Do not let a green Webex service state stand in for media

The Webex App workflow separates cloud status, phone-service connection, local device tests, call statistics and each media direction.

Classify 6940w SIP, PCLink and MobileLink audio first

When a 6940w is paired but silent or controlled by the wrong device, use the path-specific diagnostic.

SIPTRUNK signaling gateways may not be the RTP sources

Use the SIPTRUNK diagnostic to read SDP and trace each media direction before changing firewall scope.

Trace a V52 speaker path from the selected host device

The V52 diagnostic separates meeting receive, host output selection, local playback and bar speaker output.

Trace V72 host and expansion audio separately

The V72 diagnostic distinguishes host selection and meeting receive from the supported one-microphone expansion topology.

Include Trio C60 microphones, host selection and daisy chain

The focused Trio C60 workflow separates selected host, microphone, speaker, mute, expansion and daisy-chain state from call signaling.

A Cloudli answer does not prove media

The Cloudli diagnostic follows negotiated SDP and both RTP directions rather than treating an answered call as media acceptance.

An Orange call answer does not establish media

The Orange Business Talk diagnostic separates answer, early media, RTP or SRTP packets, decoding and audibility.

A Vodafone call answer does not establish media

The Vodafone Evolved Voice diagnostic separates answer, packets, decryption, decoding and audibility.

A Swisscom call answer does not establish media

The Swisscom Enterprise SIP diagnostic separates answer, packets, protection, decoding and audibility.

A Sunrise call answer does not establish media

The Sunrise Business Voice SIP diagnostic separates answer, packets, protection, decoding and audibility.

A Proximus call answer does not establish media

The Proximus Business Trunking diagnostic separates answer, packets, decoding and audibility.

A Crazytel dialog does not establish audio

The Crazytel SIP trunk diagnostic separates response, packets, decoding, audibility and the standard-versus-Hybrid SRTP boundary.

A MaxoTel registration or dialog does not prove media

The MaxoTel SIP trunk diagnostic separates published codecs, negotiation, packets, decoding and audibility.

Request the selected Aussie Broadband media contract

The Aussie Broadband SIP trunk diagnostic avoids inventing codecs, media ports, fax or encryption from adjacent product documentation.

Do not equate a Wave connected state with audible media

The Grandstream Wave diagnostic separates call-state UI, answer, SDP, packet flow, decoding and selected local devices.

Do not equate RingCentral answer or green quality with audio

The RingCentral diagnostic distinguishes local device tests, answer, media packets, decoding and remote intelligibility.

Request the selected Symbio media contract

The Symbio Enterprise SIP trunk diagnostic avoids inventing codecs, DTMF, media ranges, fax or protection from adjacent documentation.

Request the current Telstra media contract

The Telstra SIP Connect diagnostic avoids inventing codecs, DTMF, media ranges or encryption from adjacent Business SIP records.

Request the current Optus media contract

The Optus Evolve Voice diagnostic avoids inventing codecs, DTMF, media ranges or protection from older material.

Request the current One NZ media contract

The One NZ SIP diagnostic avoids inventing transport, codecs, DTMF, media ranges or encryption from marketing language.

Request the current 2degrees media contract

The 2degrees SIP diagnostic avoids inventing transport, codecs, DTMF, media ranges or encryption from the overview.

Request the current Vocus media contract

The Vocus SIP Voice diagnostic keeps product-specific codec and protection language separate from observed media.

Request the current Superloop media profile

The Superloop diagnostic keeps public quality language separate from negotiated and decoded media.

Do not equate a Zoom call state or normal statistic with audible media

The Zoom Workplace diagnostic keeps answer, packet directions, codec, decoding and what each party heard independently observable.

Trace the VP59 selected audio path and both directions

The VP59 diagnostic distinguishes handset, speaker, wired, USB and Bluetooth capture, send, receive and audibility.

Request the current Enreach media contract

The Enreach diagnostic does not generalize one profile’s codecs, DTMF, fax or encryption.

Do not equate NextivaONE answer state with two-way media

The NextivaONE diagnostic preserves packet directions, decoding and what each party heard.

Do not equate Teams answer state or a test call with two-way media

The Teams Phone diagnostic preserves packet directions, decoding and what each party heard.

Do not equate a Dialpad test or answer state with two-way media

The Dialpad app diagnostic preserves packet directions, decoding, authorized metrics and what each party heard.

A MetTel portal or dialog state does not prove media

The MetTel SIP trunk diagnostic separates published codecs, negotiation, packets, decoding and audibility.

A NUSO portal or TLS state does not prove media

The NUSO diagnostic separates signaling, selected protection, packets, decoding and audibility.

A Telesystem portal state does not prove audio

The Telesystem diagnostic separates configured trunk, answered dialog, packets and usable two-way media.

A TPx continuity event does not prove audio

The TPx diagnostic separates path selection, answered signaling, packets and usable two-way media.

Lingo redundancy does not prove audio

The Lingo diagnostic separates path selection, answered signaling, packets and usable two-way media.

An accepted Voyant dialog does not prove audio

The Voyant diagnostic separates answered signaling, packets, decoding and usable two-way media.

Do not equate an 8x8 connection indicator with two-way media

The 8x8 Work diagnostic preserves packet directions, decoding, selected devices and what each party heard.

Do not equate a VBC echo test with two-way call media

The VBC diagnostic preserves packet directions, decoding, selected devices and what each party heard.

A configured Windstream QoS class does not prove audio

The Windstream diagnostic separates signaling, class, observed packets, decoding and usable two-way media.

Do not equate an Aircall device test with two-way media

The Aircall diagnostic preserves call ID, packet directions, decoding, selected devices and what each party heard.

A loaded Granite route does not prove audio

The Granite diagnostic separates signaling, observed packets, decoding and usable two-way media.

FirstLight G.711 wording does not prove audio

The FirstLight diagnostic separates documentation, signaling, observed packets, decoding and usable media.

Include GAC2570 host and microphone selection

The focused GAC2570 workflow separates SIP or GVC state, host selection, local and GMD1208 microphones, mute, packets and audible results.

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.