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

# Custom Telephony Overview

> A step by step guide to integrate custom telephony providers

## Overview

In this guide, we will show how to integrate Retell agents with your telephony providers
and use your numbers. Please note that this is independent of the agent type you are using.

We will provide two ways to integrate with your telephony provider:

1. **Elastic SIP trunking**: This is the **recommended** way to integrate with
   your telephony provider that supports elastic SIP trunking. You will need to set up
   a SIP trunking and configure your number to point to it, and then import that number
   to Retell.
2. **Dial to SIP URI**: If your telephony provider does not support elastic SIP trunking,
   or you have more complicated telephony setup that cannot use elastic SIP trunking,
   you can dial the call to a specific SIP uri.

<Note>
  * Retell SIP server uri: `sip:sip.retellai.com`
  * IP block for traffic: `18.98.16.120/30` (All regions), `143.223.88.0/21` (certain United States traffic), `161.115.160.0/19` (certain United States traffic)

  You can use the IP block to whitelist the traffic to Retell's SIP server. It's commonly needed in a bunch of telephony providers.
</Note>

Transport method supported:

* TCP (Recommended)
* UDP
* TLS
* mTLS ([Learn more](#mutual-tls-mtls))

For using the different transport for inbound calls, you can postpend the transport method to the sip server url:

* For TCP, the sip server url needs to be `sip:sip.retellai.com;transport=tcp`
* For UDP, the sip server url needs to be `sip:sip.retellai.com;transport=udp`
* For TLS, the sip server url needs to be `sip:sip.retellai.com;transport=tls`

Media encryption supported:

* SRTP (Transport should be set to TLS to use SRTP)

Audio Codecs supported:

* PCMU
* PCMA
* G.722(HD)

## Method 1: Elastic SIP trunking (Recommended)

Elastic SIP Trunking is a service offered by cloud communications platforms that enables
organizations to connect their existing PBX (Private Branch Exchange) or VoIP (Voice over IP)
infrastructure to the Public Switched Telephone Network (PSTN) over the internet using the
SIP (Session Initiation Protocol).

In this context, the elastic SIP trunking is used to connect Retell's VoIP with
PSTN so that your agents can make and receive calls. When using this method, all the telephony functionalities that are
supported by Retell numbers will also be supported here, assuming that your telephony provider supports it.

SIP trunking is a popular service offered by many telephony providers, so most likely
your telephony provider would be able to support this.

Here're detailed guides for some telephony providers:

* [Twilio](/deploy/twilio)
* [Telnyx](/deploy/telnyx)
* [Vonage](/deploy/vonage)

Other telephony providers that support SIP trunks are also supported, feel free to use these guides as a reference.

### FAQ

<AccordionGroup>
  <Accordion title="Will it throw errors if the details or configuration is incorrect?">
    No, Retell will not be able to know if the setup you provide works or not until a call is made.
  </Accordion>

  <Accordion title="My inbound call is not connecting, what should I do?">
    Please check your origination setting in your SIP trunking provider. Also check the logs in your telephony provider,
    and perhaps open a ticket with them.
  </Accordion>

  <Accordion title="My outbound call is not connecting, what should I do?">
    Please check your termination setting in your SIP trunking provider, and make sure you provide the right termination url to Retell.
    Also check the logs in your telephony provider, and perhaps open a ticket with them.
  </Accordion>

  <Accordion title="Will I be able to transfer call using SIP trunking?">
    Yes you can use the transfer call feature with SIP trunking. Please note that if you intend to use SIP REFER (cold transfer
    with the transferee's number), you will need to configure that in your SIP trunking provider to allow SIP REFER and PSTN transfer,
    with the transferee's number showing as caller id.
  </Accordion>

  <Accordion title="Will I be able to press digit using SIP trunking?">
    Yes, you can.
  </Accordion>

  <Accordion title="Can I view and manage the numbers I import using SIP trunking?">
    Yes, you can.
  </Accordion>

  <Accordion title="I need to change the configuration of the imported number, how can I do that?">
    Right now you'd need to delete and re-import the number.
  </Accordion>
</AccordionGroup>

## Method 2: Dial to SIP URI

If your telephony provider does not support elastic SIP trunking, or you have more complicated
telephony setup that cannot use elastic SIP trunking, you can use this method.

When using this method, Retell does not directly make or receive calls, but instead
relies on your system to dial the call to the respective SIP URI. This would require you to
have some codes to handle integration with your telephony provider. All traffic looks like
inbound to Retell in this case, so it's up to you to specify the call direction.

<Warning>When using this method, you will not be able to utilize Retell's transfer call feature, as we do not
have access and control over the telephony provider and number, and cannot initiate transfer for you. You can however implement your own transfer logic and use a custom function to trigger a call transfer.</Warning>

Here we assume you already have your call handling setup.
For Retell to know what agent to use to handle the call, and for you to
**obtain SIP URI to use** for this call, you would call the
[Register Phone Call API](/api-references/register-phone-call).
You will get a `call_id` back from this API, and you would use it to piece
together the SIP URI.

<Note>SIP URI: `sip:{call_id}@sip.retellai.com`</Note>

<Warning>
  You must dial the call to the SIP URI within **5 minutes** of calling Register Phone Call. If the call is not connected within that window, it disconnects with `registered_call_timeout`. See [Debug call disconnection](/reliability/debug-call-disconnect) for disconnection reasons.
</Warning>

### Example Code

Here's an overly simplified example code to handle the inbound call
webhook from Twilio and dialing the call to the SIP URI.

```Typescript Node theme={null}
const client = new Retell({
  apiKey: 'YOUR_RETELL_API_KEY',
});

server.app.post(
  "/voice-webhook",
  async (req: Request, res: Response) => {
    // Register the phone call to get call id
    const phoneCallResponse = await client.call.registerPhoneCall({
      agent_id: 'oBeDLoLOeuAbiuaMFXRtDOLriTJ5tSxD',
      from_number: "+12137771234", // optional
      to_number: "+12137771235", // optional
      direction: "inbound", // optional
    });

    // Start phone call websocket
    const voiceResponse = new VoiceResponse();
    const dial = voiceResponse.dial();
    dial.sip(
      `sip:${phoneCallResponse.call_id}@sip.retellai.com`,
    );
    res.set("Content-Type", "text/xml");
    res.send(voiceResponse.toString());
  },
);

```

### FAQ

<AccordionGroup>
  <Accordion title="How to transfer call/end call using dial to SIP method?">
    You will need to handle the transfer call/end call logic yourself, meaning that you will need to write a custom function which internally
    interacts with your telephony provider to transfer the call/end the call.
  </Accordion>

  <Accordion title="Will I be able to press digit using dial to SIP method?">
    Yes.
  </Accordion>

  <Accordion title="Two entries created for a single dial?">
    Check your code that interacts with telephony provider. Some provider like Twilio can call the webhook multiple times for a single call.
    In those cases, you will need to dedupe the call.
  </Accordion>

  <Accordion title="Can I use telephony provider's other features like AMD detection using this method?">
    Yes, using this method basically gives you full control over telephony functionalities, so you should be able to use telephony provider's
    other features.
  </Accordion>
</AccordionGroup>

## Security Center

Retell is built on enterprise-grade security to protect your voice communications. We recommend using **TLS 1.2 or higher** to secure all SIP signaling channels between your infrastructure and Retell's SIP servers.

To establish a secure TLS connection with Retell's SIP server, you may need to add Retell's server-side root CA to your trust store. You can download it [here](https://www.amazontrust.com/repository/G2-RootCA1.pem).

### Mutual TLS (mTLS)

Mutual TLS (mTLS) is an extension of the standard TLS protocol that provides two-way authentication between a client and a server. Unlike standard TLS — where only the server presents a certificate to prove its identity to the client — mTLS requires **both sides** to present and verify certificates. This ensures that only trusted, authenticated parties can establish a connection, significantly reducing the risk of man-in-the-middle attacks and unauthorized access.

In a typical TLS handshake:

1. The server presents its certificate to the client.
2. The client verifies the server's certificate and the connection is established.

With mTLS, an additional step is added:

1. The server presents its certificate to the client.
2. The **client also presents its certificate** to the server.
3. Both parties verify each other's certificates before the connection is established.

This mutual verification makes mTLS particularly valuable for SIP-based voice infrastructure, where securing signaling and media traffic between trusted systems is critical.

**Retell supports Mutual TLS (mTLS)** for SIP connections, enabling enterprise-grade security for your telephony integrations. If you require mTLS for your setup, please reach out to [support@retellai.com](mailto:support@retellai.com) to get it configured for your account.

Retell uses a client certificate issued by AWS Private Certificate Authority (PCA). To validate Retell's TLS client certificate, you **must** add the [root certificate](https://retell-trust-store.s3.us-west-2.amazonaws.com/pca/client-root-ca.pem) to your SIP server's trusted certificate store — without this, your server will reject incoming connections from Retell.

## Video Tutorials

### Elastic SIP Trunking

<div style={{ marginBottom: "8px", fontWeight: "bold" }}>
  Twilio
</div>

<div style={{ marginBottom: "24px" }}>
  <iframe width="360" height="200" src="https://www.youtube.com/embed/JC3PV_R-Z1Y?si=JCV66VZwhtutLrWG" title="Twilio SIP Trunking Integration with Retell" frameBorder="0" allow="accelerometer; autoplay; clipboard-write; encrypted-media; gyroscope; picture-in-picture" allowFullScreen />
</div>

<div style={{ marginBottom: "8px", fontWeight: "bold" }}>
  Telnyx
</div>

<div style={{ marginBottom: "24px" }}>
  <iframe width="360" height="200" src="https://www.loom.com/embed/780f5eafbb0d4e4cbe7d6f68bd371a4f?autoplay=false&hide_owner=true&hide_share=true&hide_title=true" title="Vonage SIP Trunking Integration with Retell" frameBorder="0" allow="accelerometer; clipboard-write; encrypted-media; gyroscope; picture-in-picture" allowFullScreen />
</div>

<div style={{ marginBottom: "8px", fontWeight: "bold" }}>
  Vonage
</div>

<div style={{ marginBottom: "24px" }}>
  <iframe width="360" height="200" src="https://www.loom.com/embed/4cb20655656c4415896b2a0e6e8fa1fa?autoplay=false&hide_owner=true&hide_share=true&hide_title=true" title="Vonage SIP Trunking Integration with Retell" frameBorder="0" allowFullScreen style={{ border: "none" }} />
</div>

### Dial to SIP URI

<div style={{ marginBottom: "24px" }}>
  <iframe width="360" height="200" src="https://www.youtube.com/embed/1_BdZA8o1h0?si=IGuaBg8pArbrLNzA" title="Custom Telephony Integration Demo" frameBorder="0" allow="accelerometer; autoplay; clipboard-write; encrypted-media; gyroscope; picture-in-picture" allowFullScreen />
</div>

## Telephony Partners

In certain scenarios, you might need to manipulate the SIP call flow, or have some custom needs that are not directly supported by Retell, you can check out our telephony partners to see if they can help you.

* [Jambonz](https://www.retellai.com/app-partner/jambonz): Jambonz is a SIP server that supports static IP address and can be used to connect to Retell's SIP server to manipulate the SIP call flow. Dedicated slack support channel is available for Retell users.
* [Cloudonix](https://www.retellai.com/app-partner/cloudonix): Cloudonix is a CPaaS that can be connected to Retell's SIP server to manipulate the SIP call flow.