SIP Trunking Guide

Written By Maruf Ansari

Last updated 3 months ago

SIP Trunking creates a direct signaling path between your phone system and your agent, with no login session to keep alive. It's configured in two independent halves: Inbound, for calls arriving at your agent, and Outbound, for calls your agent places out through your phone system.

Part of SIP Connectivity. SIP Trunking is one of two ways to set up SIP Connectivity. If you haven't decided between SIP Trunking and SIP Registration yet, see the SIP Connectivity Overview for a full comparison — in most cases, SIP Trunking is the recommended method.

How this compares to SIP Registration

Aspect	SIP Trunking	SIP Registration
How it connects	A direct, server-to-server connection — by trusted IP address or SIP credentials.	Your agent registers itself to your PBX/provider, using a persistent login that must stay active.
Connection state	Stateless — no ongoing session to maintain.	Stateful — relies on a live registration that renews periodically.
Best fit	Enterprise PBXs, SBCs, and carriers that support trunk-style routing (most modern business phone systems).	Systems that only support device-style registration, e.g. legacy PBXs or extension-based providers with no static IP.

Inbound Configuration (Origination)

This is where you tell the system which calls belong to your agent, and how to trust the traffic arriving from your phone system. There are two ways to authenticate inbound calls — you can use either or both:

IP-based authentication — add your phone system's known SIP signaling IP addresses (or CIDR ranges) to Allowed Addresses. This is the simplest and most widely used method when your provider can give you static IPs — it's the approach most SIP trunking providers and PBXs recommend by default.
Credential-based authentication — if your phone system can't offer static IPs (for example, it sits behind a dynamic or shared IP), set an Auth Username and Auth Password instead. Your phone system will send these credentials with each call so the connection can still be trusted without relying on IP address.

Once inbound is configured, the system generates an Originating URL — a SIP address you give to your phone system as the destination for calls headed to your agent.

Inbound fields

Field	Purpose
Phone Number	The provider number(s) that will be used as caller ID for this trunk. You can add more than one.
Allowed Addresses (optional)	IP addresses or CIDR ranges permitted to send calls in — the IP-based authentication method.
Auth Username (optional)	Username your phone system sends to authenticate inbound calls — the credential-based method.
Auth Password (optional)	Password paired with the Auth Username above.
Originating URL	The SIP destination the system generates for you to give to your phone system.

Inbound Configuration — Auth credentials and the generated Originating URL

Outbound Configuration (Termination)

This half handles the other direction: calls your agent places out through your phone system, for example to reach a customer's number. You point your agent at your phone system's SIP address, and optionally supply credentials if your system requires them to accept calls from you.

Whitelist Kickcall's signaling IPs. Before creating your outbound trunk, add our signaling IP ranges to your phone system's or carrier's allow-list so it accepts SIP traffic from your agent. Please refer to the IP Whitelisting guide for the current list of ranges. Outbound calls may fail to connect until all listed ranges are whitelisted on your side.

Field	Purpose
Phone Number	The number your agent will use as its outbound caller ID.
Hostname or IP	Where outbound SIP INVITEs are sent — your phone system's address (no `sip:` prefix).
Destination Country	Two-letter country code for the country the call terminates in. Used to route calls.
Username / Password (optional)	Credentials your phone system requires to accept outbound calls from your agent, if any.

Outbound Configuration — destination address and optional credentials

Additional Configuration (optional)

These extra settings are collapsed by default and only need attention if your phone system has specific routing or security requirements. In most cases, the defaults work without any changes.

Field	Purpose
Media Encryption	Whether call audio is encrypted in transit. Set this to match what your phone system expects.
Transport	The SIP transport protocol used to reach your phone system. Leave as Auto to have it detected automatically.
Outbound Proxy	An intermediate SIP server your calls should route through first, if your phone system uses one.
Auth Username	A separate authentication identity, if your phone system requires one distinct from the Username above.

Additional Configuration — Media Encryption, Transport, Outbound Proxy, and Auth Username

Frequently Asked Questions

What if my phone system can't give me a static IP for inbound calls?

Use credential-based authentication instead — set an Auth Username and Auth Password in Inbound Configuration. Your phone system sends these with each call so it can still be trusted without a fixed IP.

Do I need to whitelist any IPs for outbound calls?

Yes. Please refer to the IP Whitelisting guide for the current list of ranges to add. Outbound calls may fail to connect until all listed ranges are whitelisted on your side.

Do I need to fill in the Additional Configuration fields?

No, not usually. They're optional and collapsed by default — only adjust them if your phone system has specific requirements around encryption, transport, or routing through an outbound proxy.

Can I connect more than one phone number?

Yes. You can add more than one provider phone number under Inbound Configuration — each one goes through the same setup process described in this guide.

Can my agent still make outbound calls if I only set up Inbound Configuration?

No. Outbound calling requires Outbound Configuration (Termination) to be set up separately. If you only need your agent to receive calls, Inbound Configuration alone is sufficient.

Is SIP Trunking better than SIP Registration?

For most modern business phone systems, yes — SIP Trunking is the recommended default. See the SIP Connectivity Overview for the full comparison and guidance on when SIP Registration is the better fit.

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