diagnostics: System → Diagnostics with a network test probe; RTP_PORT_START/END
The server checks itself (public IP, DNS, local networks, phone ACL, SIP transports, free audio ports) and says how to fix each problem. A quick or complete network test asks the probe (apps/probe, own image) to test this server from outside: HTTP/HTTPS, SIP over UDP/TCP/TLS, the browser phone's /ws, sampled audio ports. The probe only tests the address that asks. The audio port range comes from RTP_PORT_START/END (default 10000-20000).
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@@ -35,7 +35,7 @@ server is behind one; see [Network](network.md)):
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| 80 | TCP | Only with the bundled Caddy: getting the HTTPS certificate |
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| 5060 | UDP (+TCP) | SIP: voip.ms and desk phones |
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| 5061 | TCP | Only if you turn on SIP over TLS |
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| 10000–20000 | UDP | Call audio (RTP) |
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| 10000–20000 | UDP | Call audio (RTP); another range with `RTP_PORT_START`/`RTP_PORT_END` ([Network](network.md#call-audio-ports)) |
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Everything else (the database, the api on 8080, Asterisk's 5038 and 8088) listens on
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127.0.0.1 only. Don't open them.
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