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VoIP call connects but has no audio

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

Reviewed 2026-10-19Asterisk 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.

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.

Continue your research

Protocol reference

SDP for VoIP: read addresses, ports and media direction

Interpret the session description without confusing an offered parameter with a working network path.

Read guide →
Guide

G.711, PCMA and PCMU

Recognize G.711 payloads and calculate their network cost correctly.

Read guide →
Guide

Firewall rules for VoIP

Define management, signaling and media access separately using actual deployment requirements.

Read guide →
Troubleshooting · Asterisk 22 LTS · FreePBX 17 / Debian 12 checks

One-way VoIP audio

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

Read guide →
Troubleshooting · RFC 3960, RFC 3262 and current Asterisk 22 Progress, Ringing and res_pjsip documentation reviewed October 1, 2026; no endpoint, carrier, PBX, provisional response, packet, announcement or call was tested

No ringback or early announcement before a SIP call is answered

Trace local ringback, 180 or 183 signaling, early SDP, RTP and reliable provisional responses without confusing pre-answer media with an answered call.

Read guide →
Guide · Wireshark 4.x / current upstream tool documentation; inspect installed versions

VoIP packet-capture methodology

Select interfaces, time windows and paired capture points that distinguish network loss from missing evidence.

Read guide →
Guide · Wireshark 4.x / current upstream tool documentation; inspect installed versions

Find RTP streams and interpret loss and jitter

Separate captured packet statistics from what the receiving listener actually experienced.

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Troubleshooting · Sangoma PBX GUI Asterisk SIP Settings and legacy FreePBX 12 SIP-audio guidance plus current Asterisk res_pjsip NAT, configuration, troubleshooting, CHANNEL and bounded-debug references reviewed October 6, 2026; the legacy Sangoma page is chan_sip-era symptom and settings context, not a FreePBX 17 PJSIP configuration recipe, and no PBX, firewall, endpoint, trunk, packet flow or live call was inspected or tested

FreePBX call connects but has one-way or no audio

Trace the final SDP and observed RTP through both FreePBX call legs before changing NAT, direct media, symmetric RTP, Strict RTP, codecs or security.

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Diagnostic · Current Asterisk WebRTC and res_pjsip documentation, RFC 8445, RFC 5764, RFC 8842 and the W3C WebRTC Statistics API were reviewed October 7, 2026; no Asterisk or FreePBX host, browser, endpoint, certificate, ICE server, TURN relay, firewall, NAT, media path, packet, call, compatibility or availability was inspected or tested

Asterisk WebRTC registers but the call has no audio

Separate WSS and SIP success from SDP, ICE, DTLS-SRTP, packet flow and browser playback before changing PJSIP or firewall policy.

Read guide →
Troubleshooting · Zen Business SIP Trunking, General SIP Settings and Business Talk SIP specific terms reviewed October 12, 2026; the terms document carries an issue date of March 3, 2011 and must be confirmed against the selected current order, exact account, PBX, registration, dialog, SDP, firewall, NAT, provider address, RTP stream, codec, DTMF, fax and audio result remain deployment-specific, and no account, credential, device, PBX, packet, call, media stream or emergency call was inspected or tested

Zen SIP trunk registers but call audio is missing

Follow one Zen Business Talk SIP call from registration and dialog establishment through SDP, the documented B2BUA media ranges, both RTP directions, codec selection and decoded audio.

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Troubleshooting · Current Baresip 4.12.0 release, repository, account, configuration and module documentation reviewed October 13, 2026; the selected package, core, libre library, build options, module path, loaded modules, configuration directory, account line, credential, registrar, PBX, transport, audio devices, codec offer, SDP, RTP, protection mode and call remain deployment specific, and no package, process, file, module, account, registration, packet, device, media stream or call was inspected or tested

Baresip account loads but registration or audio fails

Trace the selected configuration directory, compiled and loaded modules, account User-Agent, registration transaction, SDP, RTP and exact audio devices before changing NAT or codecs.

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Networking guide · Current Liblinphone NatPolicy API, Liblinphone 5.5 network-parameter reference and Linphone ICE feature documentation reviewed October 14, 2026 with RFC 8445, RFC 8489, RFC 8656 and RFC 8839; library capabilities do not prove that one distributed Linphone application exposes the same controls, that an account or peer accepts ICE, that a server resolves or authenticates, that a relay was allocated or selected, or that media was delivered, and no application, SDK build, account, device, network, NAT, firewall, STUN or TURN service, DNS result, credential, candidate, SDP body, connectivity check, relay allocation, packet, media stream or call was inspected or tested

Linphone NAT traversal: ICE, STUN and TURN policy

Keep Linphone NAT policy, candidate gathering, TURN allocation, ICE checks, selected media path and audible result separately observable.

Read guide →
Diagnostic · Current Liblinphone NatPolicy API, Liblinphone 5.5 network-parameter reference and Linphone ICE feature documentation reviewed October 14, 2026 with RFC 8445, RFC 8489, RFC 8656 and RFC 8839; library capabilities do not prove that one distributed Linphone application exposes the same controls, that an account or peer accepts ICE, that a server resolves or authenticates, that a relay was allocated or selected, or that media was delivered, and no application, SDK build, account, device, network, NAT, firewall, STUN or TURN service, DNS result, credential, candidate, SDP body, connectivity check, relay allocation, packet, media stream or call was inspected or tested

Linphone ICE fails or a TURN relay is not selected

Trace one Linphone media failure from effective NatPolicy through server resolution, candidate gathering, SDP exchange, ICE checks, selected pair and RTP before changing the account.

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Guide · Asterisk Latest API branch 23 FRAME_TRACE, FRAME_DROP and CHANNEL documentation plus current Asterisk logging and debug-collection guidance reviewed October 14, 2026; FRAME_TRACE is documented since Asterisk 10.0.0 and FRAME_DROP since 16.21.0 and 18.7.0, while installed-version availability, loaded functions, channel ownership, frame direction, frame type, media topology, logger state and observed output remain deployment specific; no PBX, module, dialplan, channel, frame, DTMF event, control event, media stream, log, packet or call was inspected or tested

Asterisk FRAME_TRACE and FRAME_DROP: observe frames without masking the fault

Trace internal frame classes on one channel, keep read and write directions explicit, and reserve temporary frame dropping for controlled behavior tests with a defined restoration point.

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Troubleshooting · Asterisk Latest API branch 23 FRAME_TRACE, FRAME_DROP and CHANNEL documentation plus current Asterisk logging and debug-collection guidance reviewed October 14, 2026; FRAME_TRACE is documented since Asterisk 10.0.0 and FRAME_DROP since 16.21.0 and 18.7.0, while installed-version availability, loaded functions, channel ownership, frame direction, frame type, media topology, logger state and observed output remain deployment specific; no PBX, module, dialplan, channel, frame, DTMF event, control event, media stream, log, packet or call was inspected or tested

Asterisk frame trace shows no DTMF, voice or expected control event

Locate the first mismatch among function availability, channel leg, trace filter, RX or TX viewpoint, logger output and the external signaling or media path before dropping frames.

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Client research · Current VoIPstudio download, Desktop Softphone, user-introduction, endpoint-administration and network-troubleshooting documentation reviewed October 16, 2026; the public download surface lists Windows, Windows MSI, macOS Intel, macOS Silicon and Linux paths but does not publish one common desktop build number, and no package, workstation, operating-system permission, user account, endpoint record, VPN, firewall, WebRTC test, SIP Test, MTR result, packet, protected session, media stream or live call was inspected or tested

VoIPstudio Desktop Softphone: account, audio and network evidence

Evaluate the service-specific Windows, macOS and Linux desktop client without treating an online icon, selected device or WebRTC test as proof of a completed two-way call.

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Troubleshooting · Current VoIPstudio download, Desktop Softphone, user-introduction, endpoint-administration and network-troubleshooting documentation reviewed October 16, 2026; the public download surface lists Windows, Windows MSI, macOS Intel, macOS Silicon and Linux paths but does not publish one common desktop build number, and no package, workstation, operating-system permission, user account, endpoint record, VPN, firewall, WebRTC test, SIP Test, MTR result, packet, protected session, media stream or live call was inspected or tested

VoIPstudio Desktop is online but the call or audio fails

Separate application login, endpoint registration, call signaling, selected devices, WebRTC reachability and two-way media before changing credentials or network policy.

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Provider research · Current VoIP Much SIP-trunk product, 3CX, FreePBX, UniFi Talk, IP-blocking, SIP ALG, codec, quality, status and service-agreement pages reviewed October 16, 2026. Registration-based account examples, account-issued server and credentials, paid channel scope, supported codec lists, SIP-server port guidance and several network cautions are documented; selected-account DNS, transport and proxy profile, number inventory, caller identity, routing, failover, RTP range, DTMF, fax, emergency service, messaging, encryption, capacity, price, SLA, availability and compatibility remain account, route, contract or observation specific. No account, order, credential, trunk, number, registration, packet, media stream, status incident, emergency call or live call was inspected or tested

VoIP Much SIP trunks: registration, channels and acceptance evidence

Qualify a VoIP Much trunk against the account-issued server and credentials, paid channel scope, number routes, signaling, media, network controls and contract rather than a generic PBX template.

Read guide →
Troubleshooting · Current VoIP Much SIP-trunk product, 3CX, FreePBX, UniFi Talk, IP-blocking, SIP ALG, codec, quality, status and service-agreement pages reviewed October 16, 2026. Registration-based account examples, account-issued server and credentials, paid channel scope, supported codec lists, SIP-server port guidance and several network cautions are documented; selected-account DNS, transport and proxy profile, number inventory, caller identity, routing, failover, RTP range, DTMF, fax, emergency service, messaging, encryption, capacity, price, SLA, availability and compatibility remain account, route, contract or observation specific. No account, order, credential, trunk, number, registration, packet, media stream, status incident, emergency call or live call was inspected or tested

VoIP Much trunk registers but calls or audio fail

Separate registration, source-address policy, number and route selection, channel scope, signaling, SDP and both media directions before changing credentials or firewall rules.

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Provider research · Current VoIPline Telecom Australian SIP-trunk product, pricing, legal index, Voice and SMS (IP) service schedule, SIP-trunk critical-information summary, network, number-porting and service-status material reviewed October 16, 2026. Registration, SIP URI diversion and Microsoft Teams gateway are documented as distinct operation modes; the service schedule documents public-internet and private-IP delivery, ordered concurrent-call channels, default bidirectional service and selected codec boundaries. The selected order, registrar values, number inventory, caller identity, routing, DTMF, fax, messaging, emergency service, signaling and media protection, failover, price, SLA, availability and compatibility remain order, location, contract or observation specific. No account, order, credential, trunk, number, registration, port, packet, media stream, emergency call or live call was inspected or tested

VoIPline Telecom SIP trunks: modes, channels and call evidence

Qualify one Australian VoIPline SIP trunk by its ordered operation mode, interconnect, channels, numbers, caller identity, negotiated media, contractual boundary and observed calls.

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Troubleshooting · Current VoIPline Telecom Australian SIP-trunk product, pricing, legal index, Voice and SMS (IP) service schedule, SIP-trunk critical-information summary, network, number-porting and service-status material reviewed October 16, 2026. Registration, SIP URI diversion and Microsoft Teams gateway are documented as distinct operation modes; the service schedule documents public-internet and private-IP delivery, ordered concurrent-call channels, default bidirectional service and selected codec boundaries. The selected order, registrar values, number inventory, caller identity, routing, DTMF, fax, messaging, emergency service, signaling and media protection, failover, price, SLA, availability and compatibility remain order, location, contract or observation specific. No account, order, credential, trunk, number, registration, port, packet, media stream, emergency call or live call was inspected or tested

VoIPline trunk registers but calls or audio fail

Separate operation mode, interconnect, number and caller-identity routing, channel scope, SIP response, SDP and both media directions before changing credentials or firewall rules.

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Client research · Current Cisco Webex App calling-option, assigned-service, system-requirement, download, audio-setting, Health Checker, local Diagnostics, Calling-error and Control Hub troubleshooting documentation reviewed October 16, 2026. Call on Webex, Unified CM, Webex Calling and Calling hosted by a Service Provider remain distinct calling services with different feature and evidence boundaries. Platform, client build, entitlement, dial plan, emergency behavior, signaling and media protection, availability and compatibility remain deployment or observation specific. No organization, account, client, phone service, device, network, packet, media stream, administrator view or call was inspected or tested

Webex App calling: service modes, clients and evidence

Qualify Webex App as an assigned calling client by service, platform, entitlement, identity, device path, diagnostic evidence and observed call rather than by its shared interface alone.

Read guide →
Troubleshooting · Current Cisco Webex App calling-option, assigned-service, system-requirement, download, audio-setting, Health Checker, local Diagnostics, Calling-error and Control Hub troubleshooting documentation reviewed October 16, 2026. Call on Webex, Unified CM, Webex Calling and Calling hosted by a Service Provider remain distinct calling services with different feature and evidence boundaries. Platform, client build, entitlement, dial plan, emergency behavior, signaling and media protection, availability and compatibility remain deployment or observation specific. No organization, account, client, phone service, device, network, packet, media stream, administrator view or call was inspected or tested

Webex App phone service connects but calls or audio fail

Find the first divergence among assigned calling service, entitlement, registration, address or number, call control, selected devices and both media directions.

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Device reference · Current Mitel 6900 product and document-center pages, release 6.4 SP3 6940 and 6940w SIP user guide, release 6.4 administrator guide, accessories catalog and MX-ONE learning center reviewed October 16, 2026; Mitel exposes a release 6.7.0 documentation lane, but no phone, order, software image, call manager, WLAN, PC, mobile device, accessory, power source, provisioning service, SIP account, protected session, emergency workflow, availability, compatibility, price, audio result or call was inspected or tested

Mitel 6940w: touch screen, SIP, PCLink and power boundaries

Evaluate the Mitel 6940w without merging its SIP line, PCLink, MobileLink, Wi-Fi, Bluetooth, touch controls, accessory power or call-manager evidence.

Executive touch-screen SIP desk phone with embedded Wi-Fi and Bluetooth
Mitel lists the 6940w in its current 6900-series product page and retains model documentation in the live SIP phone document center, including a 6.7.0 lane. That does not establish regional stock, installed software, future support, entitlement or call-manager approval.

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Troubleshooting · Current Mitel 6900 product and document-center pages, release 6.4 SP3 6940 and 6940w SIP user guide, release 6.4 administrator guide, accessories catalog and MX-ONE learning center reviewed October 16, 2026; Mitel exposes a release 6.7.0 documentation lane, but no phone, order, software image, call manager, WLAN, PC, mobile device, accessory, power source, provisioning service, SIP account, protected session, emergency workflow, availability, compatibility, price, audio result or call was inspected or tested

Mitel 6940w PCLink or MobileLink pairs but audio control fails

Separate Bluetooth pairing from connection, line control, operating-system device selection, application audio, phone audio routing and the independent SIP line.

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Provider research · Current first-party SIPTRUNK device, registration, inbound-call, IP-authentication, failover, registration-status, interconnection and direct-media documentation was reviewed October 19, 2026. Credential registration and static-IP authentication remain different service paths; portal registration, Contact address, observed public source, inbound route, signaling gateways, direct-media RTP sources and failover treatments remain separate evidence. No SIPTRUNK account, trunk, credential, registration, number, route, gateway, DNS answer, PBX, firewall, packet, media stream, failover event, price, availability, compatibility or live call was inspected or tested

SIPTRUNK registration, inbound Contact routing and direct-media evidence

Keep credential registration or IP authentication, Contact delivery, inbound number matching, gateway redundancy, direct-media RTP and failover treatment independently observable.

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Troubleshooting · Current first-party SIPTRUNK device, registration, inbound-call, IP-authentication, failover, registration-status, interconnection and direct-media documentation was reviewed October 19, 2026. Credential registration and static-IP authentication remain different service paths; portal registration, Contact address, observed public source, inbound route, signaling gateways, direct-media RTP sources and failover treatments remain separate evidence. No SIPTRUNK account, trunk, credential, registration, number, route, gateway, DNS answer, PBX, firewall, packet, media stream, failover event, price, availability, compatibility or live call was inspected or tested

SIPTRUNK is registered but inbound calls or audio fail

Trace the selected authentication mode, registered Contact, provider attempt, PBX route and direct-media RTP without treating portal registration as end-to-end proof.

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