6 min read
VoIP vs SIP trunking
Hosted VoIP replaces your phone system entirely — the exchange lives in the cloud and your handsets connect to it.
SIP trunking keeps your existing on-site phone system but replaces the old ISDN lines feeding it with internet-delivered channels. It's often the cheaper route if your PBX is modern and you want to extend its life.
What VoIP needs from your network
- Roughly 100kbps per concurrent call in each direction, so 10 simultaneous calls needs about 1Mbps up and down of headroom
- Low jitter and packet loss — quality problems are almost always jitter, not bandwidth
- QoS on the router so voice traffic is prioritised over file downloads
- Wired handsets where possible; Wi-Fi calling needs a well-designed wireless network
- A router that handles SIP correctly (badly implemented SIP ALG causes one-way audio)
Troubleshooting call quality
- Choppy audio: usually jitter or contention — check QoS and whether backups run during office hours
- One-way audio: usually NAT or SIP ALG on the firewall
- Echo: normally a headset or handset volume issue, not the line
- Dropped calls: check for a router rebooting or an unstable broadband line
The cost picture
VoIP typically removes line rental entirely and reduces per-minute costs, with many plans bundling UK landline and mobile minutes. The saving over ISDN is usually significant, but the real benefit is flexibility: adding a user is a licence change, not an engineer visit.
Key takeaways
- Voice quality is a network design problem, not a bandwidth-only problem
- SIP trunking extends the life of a good existing PBX
- Bundled minutes and no line rental usually beat legacy ISDN pricing
