> ## Documentation Index
> Fetch the complete documentation index at: https://docs.getsmartalex.com/llms.txt
> Use this file to discover all available pages before exploring further.

# Standard SIP trunk

> The default way to connect SmartAlex: a standard SIP trunk from your PBX to our SIP ingress. Right for most customers, set up in minutes.

A standard SIP trunk is how most customers connect. Your phone system points a trunk at our SIP ingress, calls flow across it, and the AI answers. Nothing about your setup changes: your PBX keeps its numbers, queues, extensions, recording, and carrier.

<Note>
  This is the default connection mode. If you are not sure which mode you need, this is almost certainly the one. Other options exist for special cases, and they are covered on the [connection modes](/telephony/connection-modes) page.
</Note>

## What it is

A trunk is a persistent SIP relationship between two systems. In this case the two systems are your PBX (or carrier) and our SIP ingress at `sip.voice.getsmartalex.com`.

Two things travel over the trunk:

* **Signalling**, on ports 5060 and 5061, which sets up, controls, and tears down each call.
* **Media**, the actual audio, carried as RTP on UDP ports 10000 to 60000.

When a call arrives, your PBX sends it down the trunk to our ingress, the voice engine answers and talks to the caller, and if a human is needed the call is handed back to your PBX so it rings the right extension. Your phone system stays in charge of everything it already does.

## When to use it

<CardGroup cols={2}>
  <Card title="You run a normal PBX">
    3CX, Yeastar, FreePBX, Grandstream, Avaya, Mitel, or anything that speaks SIP. A standard trunk is the intended path.
  </Card>

  <Card title="You want the fastest setup">
    One trunk, one firewall rule, one test call. No extra hardware, no session border controller.
  </Card>

  <Card title="You keep your numbers and carrier">
    Your DIDs and your carrier are untouched. The AI answers on numbers you already own.
  </Card>

  <Card title="You want your existing routing to stay">
    Queues, ring groups, voicemail, and CDRs keep working exactly as they do today.
  </Card>
</CardGroup>

Reach for a different connection mode only if you have a specific constraint that a standard trunk cannot meet, such as a private peering requirement or a platform that cannot originate an outbound trunk. See [connection modes](/telephony/connection-modes) for the full comparison.

## Set it up

<Steps>
  <Step title="Point a trunk at our ingress">
    In your PBX, create an outbound SIP trunk whose destination host is `sip.voice.getsmartalex.com`. Use the credentials from your SmartAlex account for digest authentication. Signalling uses port 5060 for UDP or TCP, and port 5061 for TLS.
  </Step>

  <Step title="Allow the media range on your firewall">
    Open outbound UDP to ports 10000 to 60000 for RTP audio, with the stateful return path allowed. This is the single most common cause of one-way audio when it is missed. The full firewall detail, including SIP ALG and session timeouts, is on the [network requirements](/telephony/network-requirements) page.
  </Step>

  <Step title="Test a call">
    Route a test number, or a temporary inbound rule, to the new trunk and place a call. The AI should answer, hold a conversation, and transfer back to a human when asked. If audio is missing in one direction, revisit the media range and confirm SIP ALG is off.
  </Step>
</Steps>

## How the trunk fits together

The diagram below shows the two paths that make up the trunk: signalling on 5060 and 5061, and media on the 10000 to 60000 UDP range. Both run between your PBX and our SIP ingress.

```mermaid theme={null}
flowchart LR
  Caller["Caller"] --> PBX["Your PBX or carrier"]
  PBX -->|"SIP signalling: 5060 / 5061"| Ingress["Our SIP ingress (sip.voice.getsmartalex.com)"]
  PBX <-->|"RTP media: 10000-60000 UDP"| Ingress
  Ingress --> Engine["The voice engine (AI answers, understands, transfers)"]
  Engine -->|"Transfer back via SIP"| PBX
```

The AI handles the first touch. When a caller needs a person, the call is handed back down the trunk so your PBX rings the extension, queue, or ring group using its own rules.

## Common questions

<AccordionGroup>
  <Accordion title="Do I need to open anything inbound?">
    No. The trunk is originated outbound from your PBX to our ingress. You allow outbound signalling and the media range, and stateful firewalls handle the return path automatically. Inbound rules from the internet to your PBX are not required.
  </Accordion>

  <Accordion title="Which port should I use for signalling?">
    Use UDP or TCP on 5060 for standard signalling, or TLS on 5061 if you want encrypted signalling. All three reach the same ingress. Start with 5060 unless your policy requires encryption in transit.
  </Accordion>

  <Accordion title="Will this survive my existing call recording and CDRs?">
    Yes. Transfers come back to your PBX as standard SIP, so your call detail records, recording, and reporting continue to work as they do today.
  </Accordion>
</AccordionGroup>

## Next steps

<CardGroup cols={2}>
  <Card title="Network requirements" href="/telephony/network-requirements">
    Firewall rules, ports, SIP ALG, and the change request template your network team needs.
  </Card>

  <Card title="Connection modes" href="/telephony/connection-modes">
    Compare the standard trunk with the other ways to connect, and when each one applies.
  </Card>
</CardGroup>