TL;DR

Drops at 32 seconds and one-way audio are usually NAT or SIP edge issues.
Use Voyced’s SIP server via DNS NAPTR, stick with UDP, and do not enable STUN or any proxy option.
Turn off SIP ALG on the router. If needed, add simple RTP/SIP port forwards.
Voyced already runs NAT-friendly trunks. You change the bits only you control, and calls stabilise.

Users… experience call drops and choppy sound quality during important conversations

A short story

Jay starts a demo.
At 00:32 the call dies.
Next call, the client hears Jay, but Jay hears silence.
Pressure rises. The deal cools.

What is really happening

Router does NAT and “helps” SIP, then breaks it
ACK never lands, so the call ends at about 32 seconds
Media takes a path the router does not remember, so you get one-way audio

What Voyced already does on our side

NAT-friendly defaults on trunks and Hosted IPPBX
Clean SIP edge and session behaviour in the EU
Redundant SIP reachability by DNS
Diagnostics to confirm the fix on a quick remote session

What you do on your side

Follow Voyced’s setup and simple NAT hygiene.

1) Use the recommended SIP server method

Host: sip.voyced.eu
Port: 0
Transport: DNS NAPTR
If your device cannot do NAPTR, try SRV with port 5060.
Fallback if needed: sip.voyced.eu:5060, then sipnl.voyced.eu:5060.
Use UDP. Do not enable STUN or any Proxy option.

2) Stop the router from “helping” SIP

Disable SIP ALG or any SIP “helper” feature
Keep firewall stateful, not deep-inspection on SIP/RTP
If you still see one-way audio, add simple forwards:
- SIP: a unique UDP port per phone
- RTP: a fixed UDP range per phone (for example 10000–20000)
  These steps remove the guesswork for the return path.

3) Keep the audio steady

Prioritise voice in QoS or Smart Queue
Avoid double NAT, or bridge the ISP modem so your router is the only NAT
Wire key desks, or use strong 5 GHz Wi-Fi for softphones

4) Sensible device settings

Registration: keep default intervals unless support advises
Codecs: start simple
- PCMU (u-law), PCMA (a-law), then G729
- If you test extras and audio fails, return to G711 or G729

5) Quick test plan

Place a call and watch past 00:40
Hold and resume, check both ways audio
Do an attended transfer, speak to both sides
Repeat once on Ethernet and once on Wi-Fi

If it fails at a fixed time, that points to SIP timers or ACK not passing. If you hear only one side, that points to NAT path or RTP blocked.

Two fast wins

32-second drops
ALG off on the ISP router and device on DNS NAPTR to sip.voyced.eu. Call runs clean past a minute.
One-way audio after transfer
No proxy, no STUN, UDP only, and fixed RTP range forwarded to the phone. Both sides hear each other on first try.

FAQs

Do I need an outbound proxy
No. Do not enable any proxy option with Voyced’s setup. Use DNS NAPTR or SRV to reach sip.voyced.eu.

Should I switch on STUN
No. STUN is not needed with Voyced’s recommended setup. Keep it off.

Why do calls drop at 32 seconds
A missing or broken ACK is the classic cause. Fix ALG and NAT handling so ACK and media flow correctly.

What if I still get one-way audio
Add simple SIP/RTP forwards to the phone’s IP, avoid double NAT, and keep SIP “helpers” off.

Make your calls stable today

Use Voyced’s exact SIP settings
Disable SIP ALG
Keep STUN and proxy off
If needed, add small, clear forwards

Get this done with Voyced

Reply with your router model and phone brand.
We will guide you, step by step, on a short remote session and stay on until test calls pass.

Stop VoIP call drops and choppy sound with simple NAT and SIP fixes