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:

Use case category	Use case	Description
Global reach and connectivity	Outbound (termination)	Use your SIP infrastructure to send call requests to the Infobip platform for termination on PSTN destinations across the world. Infobip offers the widest reach of connectivity on the planet, with more than 200 countries in its global reach and 9 geographically dispersed data center locations accepting your SIP trunk connections in self-service.
	Inbound (origination)	Rent local DID numbers from Infobip, and calls received on these numbers are forwarded to your SIP infrastructure.
Programmable SIP	SIP trunk management	Although SIP trunks can be fully managed from the Infobip web interface, you can integrate and automate the management of your trunks using the API (creation, update, and deletion).
	Your voice application	When using the Calls API platform to develop a voice application that implements your exact scenario, your application can process inbound calls coming from your SIP trunks and create outbound calls to SIP endpoints.

Supported SIP trunk types and characteristics

Infobip supports SIP trunking to multiple environments.

SIP trunk provider	Use case	Type	Configuration	Authentication
INFOBIP	Connect to your own on-premise or cloud hosted PBX or SBC.	Static	Requires manual configuration of the trunk settings such as IP addresses, port number, codecs, and DTMF/Fax transcoding.	Uses IP-based authentication where Infobip trusts traffic from your declared IP addresses.
INFOBIP	Connect to your own on-premise or cloud hosted PBX or SBC.	Registered	Configured dynamically through the registration process between your PBX/SBC and the Infobip servers. Requires manual configuration of the trunk settings codecs and DTMF/Fax transcoding.	Uses username and password authentication to validate the PBX/SBC identity.
Freshworks	Connect to Freshcaller from Freshworks. Freshcaller uses Twilio as its communication engine. See How to configure BYOC for more details.	N/A	Requires your Twilio Account SID and Twilio SIP domain. Other parameters such as codecs, dtmf and fax transcoding settings are preset.	Twilio Account SID and traffic source.
Genesys PureCloud	Connect to your Genesys PureCloud environment.	N/A	Based on the selected Genesys region, the trunks will be automatically mapped to the adequate Infobip data center.	Genesys region, Inbound SIP termination identifier and Outbound SIP termination FQN.
Cisco Webex	Connect to your Cisco Webex environment.	N/A	Available in the USA only.	Requires your Cisco Customer Organizational ID (UUID).
OpenAI Realtime	Connect to OpenAI Realtime API over SIP for your voice AI agent projects.	N/A	Requires your OpenAI Project ID.	Requires your OpenAI Project ID.
Microsoft Operator Connect	Use Infobip as your voice provider for Microsoft Teams-based telephony services.	N/A	Requires your Microsoft Tenant ID. See Microsoft Operator Connect for configuration and usage details.	Requires your Microsoft Tenant ID.

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

Type	Support
Media	G.711a (PCMA): high-quality audio with low latency (8khz sample rate and 64kbps bit rate)
Media	G.711µ (PCMU): high-quality audio with low latency (8khz sample rate and 64kbps bit rate)
Media	G.729: for networks with limited bandwidth, requires additional processing power (8khz sample rate and 8kbps bit rate. Uses a compression algorithm to reduce the bit rate while maintaining acceptable audio quality)
DTMF	RFC2833: sends DTMF separately from the audio stream using dedicated RTP (Real-Time Transport Protocol) event message. Allows for more precise transmission of the DTMF signals, and better compatibility with various network configurations and PBX systems. Increases the overall bandwidth usage and may require additional setup and configuration.
DTMF	Inband DTMF: DTMF signals are transmitted as part of the audio stream, using the same frequency range as the voice data. Simple and widely supported, but can lead to distortion or clipping of the audio signal, particularly in low-bandwidth or noisy environments.
Fax	T38: separates the fax signal from the audio signal and transmits it as a separate stream using UDP or TCP. Most reliable and error-free.
Fax	Inband: transmits fax data as part of the audio stream using the same frequencies as voice data. Can lead to error and distortion in low-bandwidth or noisy environments.

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.

Redundancy Type	Description
Infobip SBC redundancy	For each SIP Trunk you order, Infobip provisions the trunk on two geographically-redundant SBCs within its core network. Upon failure of the primary SBC, calls automatically route to/from the secondary SBC. This capability is limited to Infobip USA data centers.
Your SBC/PBX redundancy	Static SIP trunks can be defined with multiple destination IPs (fixed public IPs that are exposed by you) and distribute calls according to your chosen policy (such as round-robin or failover). If you have redundant infrastructure in your network, you can also order multiple SIP trunks and source calls from either SIP Trunk.
Infobip Call Routing	If you have redundant infrastructure in your network, you may order multiple SIP trunks. Infobip call routing allows you to define routes that consist of SIP trunks and phone number entries. You can have up to 10 entries in a route. Incoming calls to an Infobip DID can be forwarded to a designated route and thereby trigger a hunting sequence. When building your route in Infobip call routing, the last entry in your route can be a phone number. Upon loss of connectivity to your SIP trunks defined in that route, calls destined to the DID are automatically forwarded to the number you designate. There is no additional charge for using Infobip call routing, but regular per-usage rates apply for the traffic resulting from routing. For more information, see Call routing.

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.

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. For call routing scenarios, see Call routing.