Symptom → cause → fix
| Symptom | Likely cause | Test | Fix |
|---|---|---|---|
| Choppy, robotic, "underwater" | Jitter, packet loss, Wi-Fi | Jitter, Loss | Wire the phone, QoS, replace cable, ISP ticket |
| Breaks up only when the office is busy | Bufferbloat, no QoS | Bufferbloat | SQM router, voice QoS, move backups off-hours |
| One-way or no audio | SIP ALG, firewall RTP ports, double NAT | NAT & STUN | Disable SIP ALG, open provider's RTP range, remove double NAT |
| Delay, talking over each other | High latency, VPN, satellite | Latency | Wired, local provider servers, keep voice off long VPN routes |
| Caller hears echo of themselves | Your speakerphone or headset | Mic test | Headset, lower speaker volume, replace headset |
| You hear echo of yourself | Far end's speakerphone or analog line equipment | Ask the other party | Usually on their side |
| Calls drop at 15 / 30 / 32 minutes | Session timer or firewall UDP timeout | Note exact times | Adjust firewall UDP timeout or phone keep-alive |
| Phone doesn't ring; calls go to voicemail | Registration lost, NAT timeout | DNS, Status lights | Check registration, keep-alive, firewall |
| Static, hum, buzz | Analog headset/cable, ground loop, electrical | Swap headset | USB/DECT headset, separate from power cables |
| Wrong or missing caller ID | Carrier CNAM / STIR-SHAKEN attestation | — | See FixCallerID.com |
| Cell calls drop in the building | Weak indoor signal | Cellular test | Wi-Fi calling, softphone app, signal booster |
The six-step method
Infotel Systems uses a Six Step Troubleshooting Method for every service call. The core of any method like it: define the exact symptom, gather evidence before changing anything, isolate which piece is at fault, fix one thing at a time, verify the fix with the same test that found the problem, and write it down.
What to collect before you call for help
- Date and time of a few bad calls, and the numbers called
- Who heard the problem — you, the other party, or both
- Results from the VoIP test run at a bad time
- Photos of the modem/router status lights
- Whether it happens on every phone or just one
Then send it to a tech or call 804-222-8009.
Frequently asked questions
Why are my VoIP calls choppy?
Choppy or robotic audio means voice packets are arriving late or not at all — jitter or packet loss. The usual causes are Wi-Fi, a congested internet connection without QoS, bad cables, or problems on the internet provider's network. Run the Jitter and Packet Loss tests to confirm.
How do I fix one-way audio on VoIP?
Turn off SIP ALG on the router, make sure the firewall allows the RTP port range your provider uses, eliminate double NAT, and confirm the phone isn't advertising a private IP address. If it still happens, the provider can check whether its session border controller sees your media.
Why do I hear an echo on my VoIP calls?
If the other person hears their own voice, the echo is usually from your speakerphone or headset — use a headset and lower the speaker volume. If you hear yourself, the cause is usually on the far end, often their speakerphone or analog line equipment.
Why do my VoIP calls keep dropping?
Calls that drop at the same time into every call are usually a SIP session timer or firewall UDP timeout. Random drops are usually packet loss, Wi-Fi problems or an internet outage. Note the exact times of several drops; the pattern points to the cause.
Why are there delays on my phone calls?
Delay comes from high latency — distance, a VPN, satellite or cellular links — or from bufferbloat when the connection is busy. Run the Latency and Bufferbloat tests. ITU-T G.114 recommends keeping one-way delay below 150 ms.
Is bad call quality my internet provider's fault or my phone system's?
Test on a wired computer at the time calls go bad. If latency, jitter or loss are bad even with nothing else running, and mtr shows the problem beyond your router, it's the internet path. If the network tests are clean, look at the phone configuration, NAT/SIP ALG, headsets and the VoIP provider.
Why does my VoIP phone go straight to voicemail?
The phone has probably lost its registration with the provider, often because a firewall closed the NAT mapping before the phone's next keep-alive or re-registration. Check the phone's registration status, then the firewall's UDP timeout and SIP ALG setting.
Sources & references
- ITU-T G.114 — One-way transmission time
- IETF RFC 3261 — SIP: Session Initiation Protocol
- IETF RFC 8489 — STUN
- Infotel Systems — Six Step Troubleshooting Method
Written by Eric · Reviewed November 2026