> For the complete documentation index, see [llms.txt](https://docs.voipnumber.net/llms.txt). Markdown versions of documentation pages are available by appending `.md` to page URLs; this page is available as [Markdown](https://docs.voipnumber.net/components/trunk.md).

# Trunk

{% hint style="info" %}
With a Trunk you can receive inbound calls from PSTN/mobile or `SIP` and you can send calls to `PSTN/mobile` or `SIP` from multiple `SIP` users through the single number. You can also run a single Call Flow for multiple `SIP` users multiplexed on a single `SIP` domain

There is no limit on how many concurrent calls (aka. *channels*) can run through a single trunk. The only limiting factor may be your internet bandwidth (for calls made on a single host)
{% endhint %}

Setup your SIP trunks at [app.voipnumber.net/trunks](https://app.voipnumber.net/trunks)

<figure><img src="/files/mL2DGUQWVPM22YyO0KIn" alt=""><figcaption><p>SIP trunk named <strong>sales</strong></p></figcaption></figure>

## Trunk setup

Trunk has 3 configurable sections

|  `Trunk IN`  | Defines trunk **input.** It can be SIP or/and PSTN/mobile                                        |
| :----------: | ------------------------------------------------------------------------------------------------ |
| `Processing` | You can select a Call Flow to be executed on a trunk                                             |
|  `Trunk OUT` | Defines trunk **output**. You can send traffic to PSTN/mobile networks or to another SIP address |

### Trunk IN

Name your trunk

For `SIP` input type, define a subdomain for your trunk

<figure><img src="/files/PrmhlU6GQu7u8His9fCm" alt=""><figcaption></figcaption></figure>

You can also protect your trunk with SIP authentication. Set credentials that incoming SIP traffic must present before call is accepted

<figure><img src="/files/HHsznDpnnptXfnQC9Ggi" alt=""><figcaption></figcaption></figure>

For `PSTN` input type, select a number

<figure><img src="/files/qEBiZsBkvvXoOPiVMOge" alt=""><figcaption></figcaption></figure>

### Processing

Select a [Call Flow](/components/call-flow.md) to be executed on a traffic passing through your trunk. It will make effect in parallel or in series with audio flowing through this channel, depending on the nature of your actions designed in a Call Flow. For example:

* If [Call Flow](/components/call-flow.md) is nonblocking action like `Translate speech` it will run on all audio in the call (in parallel)
* If [Call Flow](/components/call-flow.md) is a blocking action like `Play audio` or `Play text with AI` it will play that audio and execute `Trunk OUT` section once playback is completed

<figure><img src="/files/vZ4ay7piSqbM8gP8sdNL" alt=""><figcaption></figcaption></figure>

### Trunk OUT

Define destination for traffic passing through your trunk. It may be sent to PSTN/mobile networks or to a SIP address. You can also keep it disabled

For `PSTN` destination type, select a source number that will be used for making calls

<figure><img src="/files/wQIaWnG5bGWBfgvFMtt7" alt=""><figcaption></figcaption></figure>

For `SIP` destination type, define destination SIP Forward URI (domain) and port. Configure SIP authentication credentials if Forward URI is protected on remote SIP

✅ All done. You have now created elastic SIP Trunk capable of processing any number of SIP extensions

<figure><img src="/files/3swBazUj4doW84sRRnwg" alt=""><figcaption><p>With elastic SIP trunking you are in full control of SIP routing and call processing</p></figcaption></figure>

&#x20;&#x20;


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