Getting started with SIP Trunking is quick and easy with Telnyx. Follow our setup guide and you'll be talking in minutes.
Getting started with SIP trunking is quick and easy with Telnyx. Follow our setup guide and you'll be talking in minutes.
Cloud communications have become essential for modern business operations. For companies already utilizing cloud communications, transitioning to remote work is smoother than those relying on legacy systems.
A sudden migration to communication products like SIP trunking can seem daunting, but with Telnyx it doesn't have to be. If you haven't made the move yet, let us show you how easy setting up SIP connections with full outbound voice capability can be. You can also follow along with the video tutorial where a Solutions Engineer walks you through the process.
"The most critical factor in SIP trunk implementation isn't just the technical setup—it's choosing a provider with robust network infrastructure and proper security protocols. Organizations that invest time in understanding TLS encryption and proper firewall configuration see significantly fewer issues post-deployment."
— , Senior Analyst, Voice Communications, No Jitter / Informa Tech ()
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Here's a quick reference table for the complete setup process:
| Step | Action | Resource Link |
|---|---|---|
| 1 | Create a Telnyx account | Sign up for Mission Control |
| 2 | Create a SIP connection | SIP connection types guide |
| 3 | Configure your softphone | Zoiper configuration guide |
| 4 | Set up Outbound Voice Profile | Voice API documentation |
| 5 | Purchase a phone number | Number management |
| 6 | Assign number to Caller ID | Outbound options setup |
To get things started, you'll need a Telnyx account. Our self-service Mission Control Portal is where you'll control all of your connections. We've designed our Portal to be intuitive so you can sign-up in minutes. If you need any help setting up your account, our support team is available 24/7/365.
Understanding the fundamentals of SIP trunking architecture will help you make informed decisions during setup. SIP (Session Initiation Protocol) is defined by IETF RFC 3261, which establishes the standard for initiating, maintaining, and terminating multimedia sessions over IP networks.
Once you have your portal account up and running, you'll need to set up a SIP Connection. On the left-hand navigation pane of the portal, select 'SIP Connections'. Click the 'add SIP Connection' button and choose a name for your connection. When you're ready, click 'Create SIP Connection'. This will open a window where you can choose the type of SIP connection you want to set up. Choose from:
If you're setting up a SIP connection with a softphone like in the tutorial, choose the 'Credentials' SIP Connection Type.
Telnyx will automatically generate a username and password for your Credentials SIP Connection. For production use, you should use a strong password with special characters for maximum security. Click the 'Save & Finish Editing' button when you're satisfied with your username and password.
For a deeper understanding of how SIP channels work, review our guide on SIP trunk channels to better plan your capacity needs.
Proper network configuration is essential for reliable SIP trunking. Here are the standard ports and codecs you'll need to configure:
| Protocol | Port | Security | Use Case |
|---|---|---|---|
| SIP (UDP) | 5060 | Unencrypted | Standard signaling |
| SIP (TCP) | 5060 | Unencrypted | Reliable signaling |
| SIP (TLS) | 5061 | Encrypted | Secure signaling |
| Protocol | Port Range | Security | Use Case |
|---|---|---|---|
| RTP | 10000–20000 | Unencrypted | Standard media |
| SRTP | 10000–20000 | Encrypted | Secure media |
| Codec | Bandwidth | Quality | Best For |
|---|---|---|---|
| G.711 (μ-law/A-law) | ~87 kbps | High | Clarity-critical calls |
| G.722 | ~64 kbps | HD Voice | Wideband audio |
| G.729 | ~31 kbps | Good | Bandwidth conservation |
| Opus | 32–96 kbps | Excellent | Flexibility and quality |
Security recommendation: According to the SIP Forum guidelines, always use TLS 1.2 or higher for signaling encryption and SRTP for media encryption in production environments. The IETF's TLS protocol specification (RFC 5246) and SRTP specification (RFC 3711) provide the technical standards for secure communications.
This guide was written using Zoiper, but you can use any compatible softphone.
In Zoiper, look for the 'Accounts' button, click 'Add' and enter your username and domain in the following format:
[email protected]
Then enter your password and click 'Login'.
You'll see that it has registered successfully. It's a good idea to do one last sanity check of username, password, and domain before you get started. You can also change the display name on the softphone if you wish.
Before you can make a test call with your new SIP Connection and softphone, you'll need to set up an Outbound Voice Profile on the Telnyx platform. Navigate to the tab on the left-hand side of the Mission Control Portal.
Click 'Add New Profile' and give it a name, then click the 'Create' button.
You'll then need to connect the SIP Connection to your new Outbound Voice Profile. You can do this by searching for the connection by name, selecting it, and then clicking 'Add Connection/Apps to profile'.
Next, choose your Allowed Destinations. This feature allows you to limit your traffic to certain destinations or regions.
To protect your company from toll fraud, navigate to the Advanced Settings at the bottom of the page. Here you can set a Max Destination Rate and enable a Daily Spend Limit per Connection. According to the Communications Fraud Control Association, toll fraud costs businesses billions annually, making these protections essential.
You can also choose to record calls if needed.
Click 'Save'.
That's your Outbound Voice Profile created!
Before placing an outbound call, there are a couple of final steps: you'll need to purchase a number and then apply that number to the Caller ID. Navigate to the 'Numbers' tab on the menu. From here, you'll be able to see all the numbers you have purchased on the Telnyx platform.
To provision a number, navigate to the 'Numbers' tab and click 'Search & Buy Numbers'. Here you can search and buy numbers based on area code, region, and features. Once you've made your selection, click the 'Search Numbers' button to reveal a list of numbers matching your criteria.
To purchase a number, click the 'Add to Cart' button and navigate to the 'Cart' at the top right-hand corner of the screen.
In the Cart, you can assign a number to a connection in the 'Manage' field. From the dropdown menu, choose the connection you created earlier and the number will be assigned to that connection.
You can also assign the number to a messaging profile if you have one set up.
Click 'Place Order'. Your number has been successfully purchased and should now appear in the 'My Numbers' tab.
To use your newly purchased number for Caller ID, copy the number from the 'My Numbers' tab.
Navigate back to the 'SIP Connections' tab and find your chosen connection. On the right-hand side, select the icon for 'Outbound Options'. From here you can enter the number into the 'Caller ID Override' field. Click 'Save all Changes'. This will set the outbound caller ID from the softphone as your chosen number.
Now you're ready to place a call from your softphone!
If you follow this tutorial, you'll have placed an outbound call with Telnyx SIP Trunking in less than 10 minutes.
Once your basic SIP trunk is operational, consider exploring these advanced topics:
For enterprises evaluating their telecommunications strategy, Gartner's Market Guide for Global SIP Trunking Services provides comprehensive guidance on vendor selection and deployment best practices.
Check out our SIP Trunking pricing, and if you have any questions please get in touch with our team of experts. You can also explore our full product catalog to see how SIP trunking integrates with our Voice API, SMS, and AI-powered communication tools.