Overview

Written By Anand Raj

Last updated 3 months ago

SIP Connectivity lets you connect virtually any business phone system — a PBX, a UCaaS platform, or a SIP trunking provider — to your Kickcall agent using the open SIP (Session Initiation Protocol) standard. If your phone system isn't one of Kickcall's built-in Native Carrier presets, SIP Connectivity is the universal option that works with almost any standards-based system.

A SIP connection is simply a signaling link between two systems that lets calls flow between them. Setting one up means agreeing on two things with your phone system: where to send calls, and how the two sides will trust and authenticate each other.

There are two ways to establish that connection:

SIP Trunking — a direct, server-to-server connection — the standard approach used by most modern PBXs, SBCs, and carriers.
SIP Registration — your agent logs in to your phone system the way a desk phone or softphone would.

Recommended approach. For most business phone systems, SIP Trunking is the preferred method. It's simpler to keep reliable, scales better, and is what most enterprise PBXs and carriers are built to support. Use SIP Registration only if your provider specifically requires your agent to register as a device — this is common with older on-premise PBXs or extension-based providers that can't offer a static IP.

Getting started

Every SIP connection is set up from the same place in your agent, regardless of which method you choose:

Create a new agent, or open an existing one.
Go to Configure → Connect Phone.
Under Choose your telephony provider, select Bring Your Own Carrier, then choose SIP Connectivity. (Use this for any PBX, UCaaS platform, or SIP trunking provider that isn't one of the other built-in presets.)

Choose your connection method — SIP Registration or SIP Trunking — using the toggle at the top of the panel.

Choosing SIP Connectivity as your telephony provider

Only one method at a time. SIP Registration and SIP Trunking are alternatives, not additions — a given phone number on your agent uses one or the other, never both at once. Switching the toggle changes which configuration is active; it doesn't combine them.

Method comparison

Aspect	SIP Trunking	SIP Registration
How it connects	A direct, server-to-server SIP connection between your PBX/provider and your agent — either by trusted IP address or by SIP credentials on each call.	Your agent registers itself to your PBX/provider like a SIP extension or desk phone, using a persistent login (SIP REGISTER) that must stay active.
Authentication	IP whitelisting, or username/password sent with each call	Username, password, and domain used to log in and stay logged in
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
Scalability	Scales cleanly to high call volumes	Best for a single line/extension per registration
Typical setup effort	One-time exchange of IPs or credentials with your provider	One-time exchange of SIP domain/proxy and credentials with your provider
Additional call legs	None — calls are routed directly without consuming an extra SIP trunk or creating an additional call leg.	None — calls are routed directly through the registered SIP endpoint without creating an additional call leg.

Call transfers

Once your SIP connection is up and running, your agent can hand calls off to a teammate, a queue, or another extension on your phone system. There are two ways this can happen: a warm transfer or a cold transfer. Both move the caller to a new destination — the difference is how much of a "hand-off" happens along the way.

Warm Transfer — introducing the caller first

Think of this like a receptionist walking a caller down the hall and introducing them in person before stepping away. Your agent briefly puts the caller on hold, reaches out to the person or team the call is going to, and passes along a short summary — who's calling and why. Once that person is ready and agrees to take the call, the caller is connected and your agent steps out.

If the receiving person is unavailable or unable to take the call, your agent can return to the caller and offer another option instead of the call simply failing. This makes warm transfer the better choice whenever context matters — for example, routing a caller to the right department after the agent has already gathered some details.

Cold Transfer — a direct hand-off

A cold transfer is quicker and more direct — like a receptionist pointing down the hall and letting the caller walk there on their own. Your agent redirects the call straight to the destination number or extension immediately, with no introduction or summary passed along, and steps out of the call right away.

Because there's no check-in step, a cold transfer is best used when the caller already knows exactly where they want to go — for example, dialing a known extension — since there's no fallback if the destination doesn't answer.

Aspect	Warm Transfer	Cold Transfer
What happens	Caller is briefly held while your agent reaches the destination, shares context, and confirms they're ready.	Caller is redirected straight to the destination — no introduction, no summary.
Context passed along	Yes — the receiving person hears a summary before the caller joins.	No — the call is handed off as-is.
If no one answers	Your agent returns to the caller and offers another option.	The call may simply go unanswered — there's no automatic fallback.
Best for	Routing to the right person or department with context.	A quick, direct hand-off to a known extension or number.

Good to know. Both transfer types work over either SIP Trunking or SIP Registration, so your choice of connection method doesn't limit which transfer type your agent can use.

Which should you use?

In almost every case, the choice comes down to what your PBX, UCaaS platform, or carrier actually supports — not personal preference.

Start with SIP Trunking

SIP Trunking is the industry-standard way to connect two business phone systems, and it's what most modern PBXs, session border controllers (SBCs), and carriers are designed around. It has no login session to keep alive, handles higher call volumes more gracefully, and is generally simpler to keep reliable over time. If your provider can give you static signaling IPs, use IP-based authentication — it's the most common and most straightforward setup. If they can't, credential-based authentication gets you the same trunk-style connection without needing a fixed IP.

Use SIP Registration when it's required

Some systems — particularly older on-premise PBXs, or providers that issue you an extension rather than a trunk — only support the registration model, where your agent authenticates as if it were a physical device. If that's how your provider works, SIP Registration is the correct (and only) option, not a fallback to avoid.

Quick decision guide

Choose SIP Trunking if — your phone system gives you static IPs to whitelist, or generates SIP trunk/carrier credentials for you.
Choose SIP Registration if — your phone system or provider issues you a SIP username, password, and domain, and asks you to "register a device" or "add an extension".
Not sure which your provider uses — check with your provider or IT team before configuring — they'll know which model their platform supports.

Provider-specific guides

Frequently Asked Questions

Do I need to choose SIP Trunking or SIP Registration before I start?

Not necessarily. If you're not sure, start with SIP Trunking — it's the recommended default. Switch to SIP Registration only if your provider tells you it only supports device-style registration.

Can I use both SIP Trunking and SIP Registration at the same time?

No. A given phone number on your agent uses one method or the other, never both. The toggle on the Connect Phone panel switches which configuration is active.

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?

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

What's the difference between Warm Transfer and Cold Transfer?

Warm Transfer briefly holds the caller while your agent introduces them to the destination and confirms they're ready. Cold Transfer redirects the caller immediately with no introduction. Both work over either SIP Trunking or SIP Registration.

My phone system issues a username, password, and domain and asks me to "register a device". Which method do I use?

SIP Registration. That's the model your provider is describing, and it's the correct option — not a fallback.

What happens if my SIP Registration credentials change or expire?

Because SIP Registration relies on an active login rather than a stateless trunk, inbound and outbound calls will stop working until you update the credentials and the registration reconnects.

Can I connect more than one phone number?

Yes. For SIP Trunking, you can add more than one provider phone number under Inbound Configuration. Each number connected to your agent goes through the same setup process described in this guide.

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