About VoIP Toolbox

VoIP Toolbox is a free collection of tools and references for people who build, run and troubleshoot voice-over-IP systems.

Anyone who has chased a 488 Not Acceptable Here at 2am, or tried to explain to a client why their "fast" internet still gives choppy calls, knows the information is out there, but scattered across RFCs, vendor forums and old blog posts. We bring the most useful parts together in one fast, clean place that works on a phone in a server room.

What you'll find here

• SIP response code reference: every common code with real-world causes and fixes, not just the RFC definition.
• VoIP bandwidth calculator: accurate per-call and total figures including every layer of overhead.
• MOS estimator: see how latency, jitter and loss affect call quality before your users complain.
• Guides: practical walk-throughs of the problems we see most often.

Who writes it

The content is written and reviewed by people who work hands-on with SIP trunks, PBXs (Asterisk, FreePBX and others), SIP proxies such as OpenSIPS and Kamailio, and contact centre platforms. Calculations follow published standards (IETF RFCs and ITU-T recommendations), and we note where a tool uses a simplified model.

Accuracy and corrections

We work hard to keep everything accurate, but VoIP equipment varies and vendors don't always follow the standards. If you spot an error or have a fix that worked for you, please get in touch. Corrections are always welcome.

Free to use

All tools run in your browser and need no account. The site is supported by advertising, which keeps it free.