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VoIP & telephony

VoIP diagnostics

One button checks everything that affects calls: the NAT type, loss and jitter in a real UDP "call", bufferbloat under load, line speed and whether your SIP server is reachable. A plain verdict, what to fix, and a report link for your carrier or integrator.

Run it from the network where the phones are, and keep this tab in front. It takes about 30 seconds and downloads up to ~100 MB for the load check.

What is checked

  1. NAT. The browser asks several STUN servers which address and port they see. Symmetric NAT is the main cause of one-way audio.
  2. A "call". For 21 seconds voice-sized packets travel over UDP via the nearest Cloudflare node and back. In the middle the line is loaded with a download, revealing bufferbloat and the speed.
  3. SIP server. We look up SRV records, open TCP or TLS and send SIP OPTIONS: is it alive and how fast it answers.

Who to send the report to

Share gives a short link to the report. Send it to your carrier, PBX integrator or ISP: it has all the numbers: loss, jitter, delay, NAT type and your SIP server's answer.

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