---
title: "Voice providers"
slug: "voice-providers"
description: "Learn to implement and manage BYOC in Voiso by adding voice providers and setting up destination-based routing rules. Manage Voice Providers and BYOC Routing Rules."
tags: ["bring your own carrier", "BYOC", "call routing", "call routing rules", "carrier", "fallback routing", "Inbound", "P-Preferred-Identity", "PPI", "SIP", "SIP connection", "SIP trunking", "voice", "voice provider"]
updated: 2026-06-02T14:29:00Z
published: 2026-06-02T14:29:00Z
canonical: "docs.voiso.com/voice-providers"
---

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

# Voice providers

**Who should read this article**: Administrators

Learn to implement and manage BYOC in your contact center by adding voice providers and setting up destination-based routing rules.

## Introduction

The **Voice providers** feature allows you to connect your organization's existing telephony provider to your contact center. This is commonly known as **Bring Your Own Carrier (BYOC)**. Instead of using the built-in calling infrastructure, you can route calls through a carrier already in use within your organization.

You might need to use this feature if your organization:

- Has an agreement with a specific telecom provider,
- Operates in regions where a local carrier is required, or
- Needs to route calls differently depending on destination.

With **Voice Providers**, you can define how your contact center handles outbound and inbound calls at the carrier level. You assign routing rules based on **country** and **destination code** (such as regional prefixes), and you specify which provider should be used when no rule applies.

The connection between your contact center and your carrier is established through **SIP trunking**. This is a technical setup that your IT or telecom team typically manages, but you are responsible for entering the provider's connection details into your contact center and assigning how calls should be routed.

In the sections that follow, you’ll configure:

- Connection details provided by the application for your carrier
- Connection details required from your carrier
- Call routing rules based on destination
- The default provider for calls that don’t match a specific rule

## The Voice providers page

To add and manage voice call providers, in the Voiso navigation bar, navigate to **Administration > Voice providers** to display the **Voice providers** page. The page contains a list of voice call providers for your contact center.

![Voice Providers Voice Providers Tab](https://cdn.document360.io/44ae307a-3737-4fbf-98e0-9b888deb90dc/Images/Documentation/Voice%20Providers%20Voice%20Providers%20Tab.png)

The **Voice providers** page includes the following tabs:

- [Voice providers](/v1/docs/voice-providers#voice-providers) – a list of all active voice call providers
- [Call routing rules](/v1/docs/voice-providers#call-routing-rules) – a list of rules for call routing to voice providers based on country and destination codes

### Voice providers tab

The **Voice providers** tab lists the names and basic technical details of each of your connected voice call providers (carriers). The Voice providers table displays the following information for each provider:

| Head | Description |
| --- | --- |
| Provider name | The name assigned to identify the voice provider in Voiso. |
| Authentication method | The method your contact center uses to authenticate SIP requests from the provider: either **SIP account credentials** or **Static IPs**. |
| IP address | The IP address from which your contact center expects to receive SIP traffic from this provider. |
| Port | The network port used by the provider for SIP signalling. When TLS transport protocol is selected, Voiso automatically uses `5061`; however, you may modify this value to conform with your operations. |

Click **Add voice provider** to add a new carrier. Refer to [Add Voice provider](/v1/docs/voice-providers#add-voice-provider).

To delete a carrier, hover your mouse pointer over the provider row, then click **Delete** (trash can icon).

### Call routing rules tab

The **Call routing rules** tab allows you to control how outbound calls are routed through your configured voice providers. You can assign specific providers to route calls to particular countries or regional destination codes. This is useful if you want to optimize call quality, manage costs, or comply with provider-specific routing requirements.

Note

You must have more than one voice provider configured before you can create custom routing rules. If only one provider is available, all traffic is routed through that provider by default.

The **Call routing rules** table lists the routing rules you have added. Each entry includes the number prefix and the name of the assigned voice provider.

Refer to [Add call routing rule](/v1/docs/voice-providers#add-call-routing-rule) for information about adding a call routing rule.

#### Fallback routing

A built-in fallback route is added automatically once at least one provider has been set up. This route is labeled **Fallback** and is used when a call does not match any other defined routing rules.

Tip

In some contact centers, only a third-party provider is used, and Voiso is not involved in call delivery. In such cases, that primary provider is assigned the fallback routing.

Note

The fallback route cannot be deleted. Additionally, any provider associated with a routing rule — fallback or otherwise — cannot be removed until the rule is reassigned or deleted.

## Voice providers

### Add voice provider

To add a new voice call provider to your contact center, on the **Voice providers** tab, click **Add voice provider** to display the **Add voice provider** page. Configure the settings for your voice call provider, then click **Save** to add the provider to your contact center.

#### General section

Two authentication methods are supported for SIP traffic: SIP account credentials and Static IPs. From the **Authentication method** menu, choose the method required by your voice provider. Select **SIP account credentials** if your provider requires username and password authentication. Select **Static IPs** if your provider authenticates calls based on the source IP address.

Different sections are displayed on the **Add voice provider** page, depending on the method you choose.

#### SIP account credentials method

If your carrier requires SIP account credentials to authenticate for SIP traffic, the **SIP Access details section** is displayed on the **Add voice provider** page. Use the fields in this section to configure the required credentials.

**SIP Access details section**

To configure the SIP access details for your carrier, enter the following information in the **SIP Access details** section, then click **Save**.

![Voice Providers SIP Access Details Section](https://cdn.document360.io/44ae307a-3737-4fbf-98e0-9b888deb90dc/Images/Documentation/Voice%20Providers%20SIP%20Access%20Details%20Section.png)

| Field | Description |
| --- | --- |
| Provider name | A designation for your carrier to be displayed in the Voice providers table. |
| Domain name | The domain of the SIP account address (e.g., my.name@mydomain.abc). If not specified, the SBC IP proxy domain is used instead. |
| Transport Protocol | The network protocol used for SIP signalling between your contact center and your voice provider. Supported protocols are UDP (connectionless), TCP (connection-oriented), and TLS (encrypted). |
| Authorization realm | Specify the authentication realm if provided by your voice provider. If left blank, Voiso will attempt to obtain it automatically during SIP registration. |
| SBC IP proxy and port | The IP address and port of the SIP proxy provided by your voice provider. Voiso sends and receives SIP traffic through this address. When TLS transport protocol is selected, Voiso automatically uses port `5061`; however, you may modify this value to conform with your operations. |
| Username | The User ID of the SIP account, without the domain name. |
| SIP registration expiry (sec) | The interval, in seconds, after which your contact center re-registers the SIP account with the provider. |
| Password | The password associated with the SIP account. |
| P-Preferred-Identity header | Select this option if the `P-Preferred-Identity` (PPI) SIP header is required by your provider. |
| P-Asserted-Identity header | Select this option if your provider requires the `P-Asserted-Identity` (PAI) header to be included in SIP messages for call authentication and caller identification. |
| Use E.164 compliant numbers | Add a leading "+" to phone numbers in SIP headers if required by your voice provider to ensure E.164-compliant formatting. |
| SRTP | (For TLS only) Select this option if **Secure RTP** (SRTP) is required by your voice provider to encrypt audio streams or to meet industry-specific compliance requirements. Use with TLS to secure both signalling and media when connecting to your provider. |
| Use From header as Contact in INVITE | Select this option to copy the caller identity from the **From** header into the **Contact header** for outbound SIP INVITEs. Some BYOC providers or SBC configurations require this header alignment for validation and anti spoofing policy checks. Enable this option only if your provider instructs you to do so or if call setup fails due to a From and Contact mismatch. |

#### Static IPs method

If your carrier uses IP-based authentication for SIP traffic, the **Voiso SIP connection details** and **Voice provider SIP connection details** sections are displayed on the **Add voice provider** page. Use these sections to provide the SIP connection details that your contact center will share with your provider, and to enter the SIP connection details required by your provider.

Important

Ensure that the IP addresses provided by Voiso are allow-listed by your provider.

![Voice Providers Static IPs Method](https://cdn.document360.io/44ae307a-3737-4fbf-98e0-9b888deb90dc/Images/Documentation/Voice%20Providers%20Static%20IPs%20Method.png)

**Voiso SIP connection details section**

The **Voiso SIP connection details** section is automatically populated with information you must share with your voice provider. Your provider uses these settings to configure their system for SIP connectivity with your contact center. The table below describes the fields in this section.

| Field | Description |
| --- | --- |
| Name | Voiso. This value cannot be changed. |
| Inbound call SIP address | The SIP address format (SIP URI format) required by your contact center for receiving calls from your provider. |
| IP address and port | The IP address and port your contact center uses to receive SIP traffic. Your provider must send calls to this address. When TLS transport protocol is selected, Voiso automatically uses port `5061`; however, you may modify this value to conform with your operations. |

**Voice provider SIP connection details section**

You must fill in the fields in the **Voice provider SIP connection details** section using information provided by your voice carrier. These details allow your contact center to establish a connection with your provider's system for outbound calls.

| Field | Description |
| --- | --- |
| Provider name | A designation for your carrier to be displayed in the Voice providers table. |
| Transport Protocol | The network protocol used for SIP signalling between your contact center and your voice provider. Supported protocols are UDP (connectionless), TCP (connection-oriented), and TLS (encrypted). |
| IP address and port | The IP address and port your voice provider uses to receive SIP traffic. Your contact center will send outbound calls to this address. When TLS transport protocol is selected, Voiso automatically uses port `5061`; however, you may modify this value to conform with your operations. |
| Prefix to add | Your contact center automatically adds this prefix to all dialed numbers before routing outbound calls to your voice provider. Leave blank unless your provider requires a prefix. |
| P-Preferred-Identity header | Select this option if the `P-Preferred-Identity` (PPI) SIP header is required by your provider. |
| P-Asserted-Identity header | Select this option if your provider requires the `P-Asserted-Identity` (PAI) header to be included in SIP messages for call authentication and caller identification. |
| Use E.164 compliant numbers | Add a leading "+" to phone numbers in SIP headers if required by your voice provider to ensure E.164-compliant formatting. |
| SRTP | (For TLS only) Select this option if **Secure RTP** (SRTP) is required by your voice provider to encrypt audio streams or to meet industry-specific compliance requirements. Use with TLS to secure both signalling and media when connecting to your provider. |
| Use From header as Contact in INVITE | Select this option to copy the caller identity from the **From** header into the **Contact header** for outbound SIP INVITEs. Some BYOC providers or SBC configurations require this header alignment for validation and anti spoofing policy checks. Enable this option only if your provider instructs you to do so or if call setup fails due to a From and Contact mismatch. |

### Edit voice provider

Click a provider name to view or edit its configuration on the **Edit voice provider** page. The fields on the **Edit voice provider** page are the same as those on the **Add voice provider** page. Make updates to the information in the required field or fields, and then click **Save**.

Note

The default provider is read-only and may not be modified or deleted.

### Delete voice provider

You can delete any voice provider that you have added. The default voice provider may not be deleted. To delete a voice provider, select the **Voice providers** tab, then click the trash can icon on the right side of the provider's row in the table.

Note

If a call routing rule has been assigned to a voice provider, you cannot delete the provider. You must first [edit the call routing rule](/v1/docs/voice-providers#edit-call-routing-rule) to unassign the voice provider.

## Call routing rules

### Add call routing rule

On the **Call routing rules** tab, click **Add call routing rule** to open the configuration page.

![Voice Provider Add Call Routing Page](https://cdn.document360.io/44ae307a-3737-4fbf-98e0-9b888deb90dc/Images/Documentation/Voice%20Provider%20Add%20Call%20Routing%20Page.png)

In the **General** section, you must define the following fields:

- **Country code** – Select the destination country for this rule.
- **Destination code** – (Optional) Enter a regional or national prefix used to route calls more precisely within the selected country.
- **Voice provider** – Select the carrier to handle calls that match this rule.

Click **Save** to add your call routing rule. Once configured, your routing rules appear in the table on the **Call routing rules** tab.

### Edit call routing rule

To edit a call routing rule, select the **Call routing rules** tab, then click the name of the call routing rule to display the **Edit call routing rule** page. The fields on the **Edit call routing rule** page are the same as those on the **[Add call routing rule](/v1/docs/voice-providers#add-call-routing-rule)** page. Modify the required fields, then click **Save** to update the rule.

### Delete call routing rule

You can delete any call routing rule that you have added. To delete a call routing rule, select the **Call routing rules** tab, then click the trash can icon on the right side of the call routing rule row in the table.

Alternatively, click the name of a call routing rule to display the **Edit call routing rule** page for the rule. Click the trash can icon on the top right of the page.

Note

When you delete a call routing rule, calls to or from that destination will default to the [Fallback call routing rule](/v1/docs/voice-providers#fallback-call-routing-rule).

### Fallback call routing rule

The **Fallback** call routing rule may not be deleted. This rule is used when no other call routing rule has been added or when a call to or from a destination is not covered by a custom routing rule.

If you have more than one voice provider, you can specify which voice provider is assigned to the fallback rule. [Edit the fallback calling rule](/v1/docs/voice-providers#edit-call-routing-rule), use the **Voice provider** menu to select the a voice provider, then click **Save** to update the rule.

Session Initiation Protocol. A signaling protocol used to initiate, maintain, and terminate communication sessions over the internet, including voice, video, and messaging applications.