SIP trunking

Session Initiation Protocol (SIP) trunking is a digital communications technology that enables voice calls to be transmitted over IP networks instead of traditional circuit-switched telephone lines. Rather than relying on physical wiring, SIP trunking uses packet-based transmission to carry voice data over the internet.

This approach reduces costs associated with long-distance and international calls and provides scalability by allowing call capacity to be adjusted as needed. It also offers flexibility in managing telephony systems by enabling the addition or removal of channels without requiring physical infrastructure changes.

Businesses typically use SIP trunking for:

• Cost savings: SIP trunking reduces expenses on long-distance and international calls.
• Scalability: SIP trunking enables you to add or remove lines as needed.
• Business continuity: SIP trunking provides redundancy and failover options so that calls continue even during an outage.

To use SIP trunking, you need:

• An IP-based Private Branch Exchange (PBX) system, which manages internal and external calls, or an SBC (Session Border Controller) that acts as a firewall between your internal network and the internet.
• A reliable internet connection: since SIP trunking relies on the internet to make and receive calls, a stable and fast internet connection is crucial.

Use cases with Infobip SIP trunking

The following table describes common use cases:

Supported SIP trunk types and characteristics

Infobip supports SIP trunking to multiple environments.

SIP trunk channels and related billing plans

When creating a SIP trunk, you need to define the number of channels to be allocated. A channel represents a single concurrent call, so a 10-channel trunk means 10 concurrent calls can take place at any moment in time, whether inbound and/or outbound. Calls submitted to the trunk when the trunk has reached its channel capacity will be rejected.

You can choose between two different channel plans: Metered or Unlimited.

Metered channel plan

With the metered plan, the per-channel price is the same regardless of the call destination or the SIP trunk location (Infobip data center).

The voice traffic (between Infobip and the telco operators) is billed by usage, regardless of the destination or origination.

Unlimited channel plan for US domestic traffic

With the unlimited channel plan, the outbound US domestic voice traffic is not charged by usage (subject to fair use policy). Any traffic to or from other countries will be billed by usage.

Technical requirements

SIP methods

The following SIP methods are supported:

• INVITE and reINVITE
• ACK
• BYE
• CANCEL
• OPTIONS

Transport

The following transport mechanisms are supported:

• UDP (User Datagram Protocol): a connectionless transport mechanism used to transmit voice data between endpoints. It is a lightweight, fast protocol that does not require handshaking or acknowledgment of received packets, making it suitable for real-time applications such as voice calls. However, it does not provide encryption or authentication.

• TLS/SRTP: secure transport mechanisms that use encryption and authentication to protect against eavesdropping, tampering, and other security threats. TLS (Transport Layer Security) encrypts SIP signaling traffic between the PBX and the Infobip infrastructure, while SRTP (Secure Real-time Transport Protocol) encrypts the voice traffic itself. Both require a handshake and verification process, which introduces some latency and overhead but provides higher security and privacy.

Codecs and transcoding

SIP trunking redundancy

Infobip provides multiple levels of redundancy for SIP trunks and routing of DID to SIP. At a high level, these capabilities fall into three categories.

Service address

A service address, also referred to as the Place of Primary Use (PPU), is the physical location where a SIP trunking service is primarily used or consumed. It corresponds to the location of the SIP-terminating equipment (SBC or PBX) connected to Infobip data centers.

The service address (PPU) is independent of the location of the service provider infrastructure (for example, data centers). SIP-terminating equipment connects to the provider network, but taxation and regulatory obligations are determined based on the service address.

It is an important concept for regulatory and taxation purposes in the telecommunications industry, particularly in jurisdictions such as the United States, for the following reasons:

• Taxation: Telecommunications services in the United States are subject to taxes and fees imposed at the federal, state, and local levels. Applicable tax rates vary based on the location where the service is used. Defining the service address or PPU enables accurate tax determination and supports compliance with tax regulations.
• Regulatory compliance: The telecommunications industry in the United States is regulated by federal and state authorities. Compliance with these regulations is essential to ensure fair competition, consumer protection, and adherence to industry standards. Service address information is used by service providers to determine the applicable regulatory jurisdiction and ensure compliance with regulatory requirements.
• Jurisdictional boundaries: Telecommunications services may be subject to different regulatory requirements when crossing state or local boundaries. Defining the service address ensures that services are governed according to the appropriate jurisdiction.

For SIP trunking services with a PPU in the United States, a service address must be associated with the trunk. For services with a PPU outside the United States, associating a service address is not mandatory but is recommended for documentation and administrative purposes.

Managing service addresses

You can manage service addresses using the Infobip web interface. Log in to your account and go to Channels and Numbers > Channels > Voice and WebRTC > SIP Trunking > Service Addresses.

Setting up a SIP trunk

For step-by-step instructions on creating and configuring a SIP trunk using the Infobip web interface or API, see Set up a SIP trunk.

For Microsoft Operator Connect configuration, see Set up Microsoft Operator Connect.

Understanding SIP trunk status

Infobip SBC locations

When you create a SIP trunk, Infobip returns the SBC address(es) you need to configure on your environment. The format depends on whether your account already has existing SIP trunks:

• If you had no existing SIP trunks in your account on April 13, 2026 (first trunk ever created): you receive a unique FQDN for each trunk you create, along with a list of IP subnets to allowlist on your firewall.
• If you already had one or more SIP trunks in your account on April 13, 2026: you continue to receive static SBC IP addresses, including for any new trunks you create.

You can check whether you have existing SIP trunks in the SIP trunking section of your Infobip account.

SBC addresses for accounts with existing SIP trunks

The following table lists the SBC IP addresses by data center.

A migration path from IP-based to FQDN-based trunks will be made available later in 2026.

SBC addresses for new accounts (FQDN-based)

When you create your first SIP trunk, the system generates a unique FQDN for that trunk. The FQDN follows this format:

{unique-id}.{region}.sip.voice.infobip.com

For example: a1b2c3d4-e5f6-7890-abcd-ef1234567890.eu.sip.voice.infobip.com

The FQDN is returned at trunk creation and is specific to that trunk. Each trunk gets its own FQDN.

In addition, you must allowlist the returned IP subnets on your infrastructure to allow traffic from the Infobip SBCs.

SIP trunk creation limitations

By default, a single Infobip account can have:

• Up to 10 trunks per account.
• Up to 10 trunks per Infobip data center.
• Up to 3 IP addresses per trunk for inbound traffic.
• Up to 3 IP addresses per trunk for outbound traffic.
• SIP trunks in up to 2 different Infobip data centers.

Setting up call routing

In this context, you rent one or several numbers from Infobip and want calls received on these numbers to be forwarded to your telephony equipment over your newly created SIP trunk(s). Infobip call routing is the Infobip product with which you can implement such scenario.

Specific notes for provider trunks

The following information is relevant to the specific provider trunks only.

Freshworks trunks

Freshworks provider trunks allow you to use Freshworks services (call center Freshcaller (opens in a new tab)) while leveraging Infobip as the underlying voice connectivity provider for both inbound and outbound calls.

Freshcaller uses Twilio as its communication engine. To use Infobip for voice connectivity with Freshcaller, you must:

• Configure a Bring Your Own Carrier (BYOC) connection with Twilio
• Have a properly configured Freshworks account

Follow the official Freshworks documentation (opens in a new tab) for setup instructions.

To set up an Infobip provider trunk for Freshworks, you must provide these two specific parameters:

When defining multiple Freshworks trunks in the same Infobip data center, consider the following specific behaviors:

• Channels
  • On inbound traffic (origination): The channel count defined per SIP trunk is applied as a channel upper limit on each individual trunk.
  • On outbound traffic (termination): The total channel count for all defined Freshworks trunks is applied as an upper group limit across all Freshworks trunks.
• Reports and Logs
  • On inbound traffic (origination): Reports and logs show the SIP trunk ID and SIP trunk name used for sending traffic to Freshworks.
  • On outbound traffic (termination): Reports and logs show the ID and name of the oldest provisioned Freshworks trunk. This is because traffic from Freshworks does not include a trunk identifier.

Genesys PureCloud trunks

See the Genesys PureCloud Provider SIP Trunk Configuration Guide (opens in a new tab) to help you set up a Genesys PureCloud trunk and configure it in Genesys PureCloud.

You can determine the appropriate region where the SIP trunk needs to be created based on your Genesys PureCloud web interface login URL:

Cisco Webex trunks

See the Cisco Webex Provider SIP Trunk Configuration Guide (opens in a new tab) to help you set up a Cisco Webex trunk and perform the Cisco Webex BYOC Enterprise configuration.

OpenAI Realtime SIP trunks

To connect Infobip voice to the OpenAI Realtime API over SIP, provide your OpenAI Project ID.

Connect inbound caller to your AI agent

Use Infobip Call Routing to connect inbound callers to your OpenAI Realtime project.

To set this up, follow these steps:

1. Create a new route in Call Routing and set your OpenAI trunk as the destination.
2. Configure the route based on how users will call your agent:
   • PHONE: Ensure you have at least one Infobip DID. For detailed steps, see Call Routing documentation.
   • WHATSAPP: Make sure you have at least one WhatsApp sender enabled for WhatsApp Voice. The setup is similar to phone, but inbound configuration is done from the WhatsApp Voice tab of your sender in the Numbers application on the web interface.
   • WEBRTC: See the filter-based route execution documentation. In summary:
     • Your WebRTC client (using Infobip's WebRTC JS or mobile SDK) must place an applicationCall to CALL_ROUTING.
     • Your route in Call Routing must have a filter criteria defined of type WEBRTC.

Enable your AI agent to call recipients

Outbound calls from OpenAI to Infobip over the SIP trunk are not supported directly. To enable your AI agent to call recipients, build a voice application using the Calls API platform.

This application:

• Creates the outbound call to the recipient(s), either as an individual call (opens in a new tab) or bulk calls (opens in a new tab).
• When the call is answered, connect it to an outbound SIP call to your AI agent by:
  • Using the create Dialog (opens in a new tab) method.
  • Setting the childCallRequest to use the SIP endpoint type and referencing your OpenAI trunk's sipTrunkId.