> ## Documentation Index
> Fetch the complete documentation index at: https://docs.firetone.com.au/llms.txt
> Use this file to discover all available pages before exploring further.
# Connecting to your PBX
> The SIP account screen: credentials, transport, codecs, and proving it registered.
One account, one extension. **Settings › FireTone SIP** opens it, and the card
at the top of Settings is where you switch the line on and off.
## The four things you must have
| Field | What goes in it |
| ---------------- | -------------------------------------------------------------------------------------------------------------------------------------------------------- |
| **Display name** | What the far end sees. Yours to choose. |
| **Username** | Your extension. Write it as `extension@sip-domain` when the SIP domain differs from the host you register against — which it usually does on FreeSWITCH. |
| **Password** | From your administrator. It is held in the phone's secure storage, not in ordinary settings. |
| **Server** | A host, a `host:port`, or a full WebSocket URL such as `wss://pbx.example.com:7443`. |
Your SIP password is a credential like any other. Nobody should ask you to
read it out, and it is never shown back to you once saved — the field holds
dots, and the eye reveals only what you have just typed.
## Transport
Three choices, and they are not equivalent.
* **TLS** registers over a secure WebSocket, port 7443 unless you give another.
Encrypted, and the right answer wherever the PBX offers it.
* **TCP** registers over plain SIP on 5060. Works, and is what most FreeSWITCH
installations hand out first.
* **UDP** also uses 5060, and **cannot carry a FireTone call**. The app warns
you in red when you pick it. The registration will succeed and then every
call will fail, which is a worse outcome than refusing outright.
## Codecs
The list comes from the media stack in the app, not from a table written here,
so it shows exactly what this build can negotiate. Drag to reorder — the order
is the preference order sent in the offer. **More codecs** reveals the rest.
The defaults are **Opus** and **G.729**: the best-sounding one and the cheapest
one. Opus is 48 kHz and adapts to the network; G.729 is 8 kHz and uses the
least bandwidth of anything on the list, which is why PBX trunks are full of
it.
At least one codec must stay on. Turn them all off and a call cannot be
offered at all — the app will refuse to save, and will refuse to join an
intercom channel with a message saying so.
**Low bandwidth** roughly halves what a call uses by running Opus narrow and
capping the send rate. It needs Opus, and switches itself off if you turn Opus
off.
## Advanced
**Local port range** sets the source ports the phone binds for SIP over UDP and
TCP. The default 16000–32000 suits almost everybody; change it only if your
network team has asked you to.
## Proving it worked
**Test registration** at the bottom of the screen registers once and reports
back — `Test registration succeeded · 240 ms`, or the reason it did not. The
card at the top of Settings then reads `registered`, with the transport, the
codecs in force and how long the line has been up.
If it does not, [the SIP log](/mobile/audio-and-diagnostics) has the reason in
full.