How to Set Up SIP Trunking for Your Business: Step-by-Step Guide (2026)

Step-by-Step: How to Set Up SIP Trunking

Setting up SIP trunking is pretty straightforward, especially if you divide it into these simple steps. This guide walks you through the full configuration process, from choosing a provider to going live. Here’s what we’ll cover:

1. Choose the right SIP trunking provider

2. Collect your SIP credentials

3. Set up call routing rules

4. Test your trunk

5. Go live and monitor performance

Step 1: Choose the Right SIP Trunking Provider

Your SIP trunking provider is the foundation of your entire voice setup. Even a perfect configuration will fail if the provider’s network is unstable, coverage is limited, or support is lacking.

Here’s what to look for:

• Reliability: Look for providers with a proven track record of uptime and clear service-level agreements (SLAs). 99.99% uptime or better is standard for business-grade solutions.

• Redundancy: Global routing infrastructure, multiple points of presence (PoPs), and failover support ensure your calls are not interrupted if one region has issues.

• Codec Support: At minimum, ensure the provider supports G.711 and G.729. Some support Opus for HD audio — check compatibility with your PBX.

• Security Standards: Look for TLS + SRTP support, IP whitelisting, and DDoS mitigation to keep your calls and network secure.

• Scalability & Coverage: Global DID availability, flexible channel scaling, and wide termination reach are essential for growing businesses.

• Support Quality: You’ll want fast, knowledgeable help during setup and ongoing usage — ideally via live chat, email, and phone.

– The 5 Best SIP Trunking Providers 2026 –

Step 2: Collect Your SIP Credentials and Create the SIP Trunk

Once you’ve selected your SIP trunking provider, the next step is to collect the login and routing details that allow your PBX to communicate with their network. These are typically found in your provider’s dashboard or setup email, and will be used during the creation of a trunk.

Here’s what you’ll need:

• SIP Username / Account ID
  This is your unique identifier used for registering your trunk with the provider.

• SIP Password
  Used alongside your username to authenticate access to the SIP trunk.

• SIP Domain
  This is the server or domain where your PBX sends registration requests

• Outbound Proxy
  Some providers use a separate outbound proxy for routing calls. If required, this will be provided.

• SIP Port
  Typically UDP 5060 or TLS 5061, though it may vary based on the provider and security settings.

• Auth Type
  Some providers support IP-based authentication as an alternative to username/password, commonly used for higher security and static-IP deployments.

The next step is to log into your PBX or cloud-based phone system to begin configuring the trunk. This is where you’ll input the SIP details and connect your phone system to the provider’s network.

Now that you’re inside your PBX platform, it’s time to create the SIP trunk connection using the credentials from your provider. This is the core setup that enables your PBX to register with the SIP network and route calls in and out.

Step 3: Configure Call Routing

Once your SIP trunk is registered, the next step is to define how calls are handled — both outbound and inbound. Proper routing ensures that calls reach the correct destination inside and outside your organization.

Outbound Routing (External Calls)

Outbound call routing determines how your PBX sends calls through the SIP trunk. This typically involves:

• Dial Plans: Define which digits your users dial to reach specific destinations (e.g., international, emergency, extensions).

• Prefix Rules: Strip or add digits based on dialing habits — for example, converting “0” to country codes for international calls.

• Route Priority: You can set primary and backup SIP trunks to failover in case of issues.

Inbound Routing (Receiving Calls)

Inbound routing determines what happens when someone calls your business number (DID):

• DID Mapping: Assign specific phone numbers to extensions, departments, or IVRs.

• IVRs (Auto-Attendants): Route callers based on menu selections (e.g., “Press 1 for sales”).

• Ring Groups or Hunt Groups: Distribute calls to teams based on availability or priority.

• Time Conditions: Route calls differently during business hours, after hours, or on holidays.

Step 4: Test Your SIP Trunk

Before going live, it’s critical to test your SIP trunk thoroughly to ensure everything is working as expected. This phase helps detect misconfigurations early and confirms that your system is ready for real-world use.

Check Registration Status
Confirming that the SIP trunk is showing a status like “Registered,” “Connected,” or “Online.” This indicates that your system has authenticated with the SIP provider. If it shows “unreachable” or “failed,” revisit your credentials, SIP server address, and firewall settings.

Make Inbound and Outbound Test Calls
Confirm that calls connect correctly, audio flows both ways, and DTMF tones (e.g., keypad input during IVRs) function. This confirms that routing and media handling are configured properly.

Measure Call Quality
Good call quality generally means latency under 150ms, jitter under 30ms, and near-zero packet loss. If performance is poor, it may indicate a network issue such as congestion, inadequate bandwidth, or misconfigured QoS rules.

Troubleshoot Common Issues
Calls failing to complete may be due to dial plan errors or misrouted numbers. If your SIP trunk won’t register at all, verify that the SIP domain, port, and authentication details match exactly with what your provider has given you.

Step 5: Go Live and Monitor Performance

Once you’ve validated that the SIP trunk works, you’re ready to deploy it across your business. But the work doesn’t stop there — the first 72 hours are critical.

1. Monitor Logs & Dashboards: Pay attention to whether calls are connecting as expected, if trunks are re-registering too frequently, or if error codes like 403 (forbidden), 408 (timeout), or 503 (service unavailable) appear.

2. Track Quality Metrics: Keep a close eye on key voice performance indicators like jitter, packet loss, and latency. If your PBX supports analytics, monitor average call duration, destination types (local, mobile, international), and channel utilisation trends.

3. Scale Channels as Needed: Many providers allow you to instantly add channels through a user dashboard, eliminating the need for manual provisioning. Staying ahead of growth ensures your callers never hear a busy signal or experience degraded quality during peak hours.