When VoIP calls go wrong, the network is the usual suspect. The good news is that most problems come from a short list of firewall and NAT settings that are easy to check once you know what to look for.
Two kinds of traffic
- SIP signalling sets up and ends calls. It normally uses port 5060 (UDP or TCP), or 5061 for encrypted TLS.
- RTP media carries the actual audio, over a wide range of UDP ports chosen per call.
A call can be set up perfectly over SIP and still have no audio if RTP is blocked. That is why "the call connects but nobody can hear anything" is such a common complaint.
Allow the right IPs and ports
- Allow your provider's signalling IP ranges on the SIP ports.
- Allow the provider's media IP range on the RTP port range.
- Restrict SIP to those IPs only — open SIP ports attract constant scanning and fraud attempts.
On ReachMeLine, the exact signalling and media ranges to allow are listed on your SIP trunks page.
Fix NAT properly
If your PBX sits behind a router, it may advertise its private address (like 192.168.x.x) inside SIP messages. The other side then sends audio to an address it cannot reach.
- Set your PBX's external address (for example
external_media_addressandexternal_signaling_addressin Asterisk PJSIP) to your public IP. - Set the local network ranges so the PBX knows which addresses are internal.
- Forward the RTP port range from the router to the PBX.
Turn off SIP ALG
Many routers have a "SIP ALG" feature that rewrites SIP messages. In practice it often breaks calls — causing one-way audio, failed transfers or registrations dropping. Unless your provider tells you otherwise, disable it.
Quick symptom guide
| Symptom | Likely cause |
|---|---|
| One-way or no audio | RTP blocked, wrong external address, SIP ALG |
| Calls drop after about 30 seconds | Acknowledgement messages not reaching your server (NAT/firewall) |
| Inbound calls never ring | Signalling IPs blocked or wrong inbound destination |
| Choppy audio | Bandwidth, jitter or Wi-Fi — test on a wired connection |
