Your business’s guide to SIP trunking architecture

If you're a developer or a business owner, you're well aware of the importance of efficient communication. As the foundation of modern connectivity, SIP (Session Initiation Protocol) trunking is key to unlocking streamlined communication for your organization.

By allowing businesses to replace traditional phone lines so voice and other communication data can flow seamlessly over the internet, SIP trunking helps organizations save time and money when connecting with customers and colleagues. But knowing what SIP trunking does is different than knowing how it works.

With a basic understanding of SIP trunking architecture, you can make more informed decisions about how to implement SIP trunking in your organization. In this post, we’ll explain the components of SIP trunking and the tools you need to integrate this technology with your existing communications infrastructure.

The core architecture of SIP trunking

SIP trunking architecture primarily concerns the seamless integration and communication between different network components to facilitate Voice Over IP (VoIP) communications. Several elements come together to create the technology that makes digital communication possible:

• SIP client: Typically an IP phone or softphone, initiating and terminating SIP sessions.
• IP-PBX: The internal telephone switching system, which now functions as a SIP client.
• SIP server: Acts as the intermediary, handling SIP call setups and terminations.
• Media server: Responsible for the transmission of voice, video, and other media streams.
• Border element: Manages the traffic between the internal network (LAN) and the external internet (WAN).

In addition to the software and hardware that makes SIP trunking work, a handful of protocol layers determine how voice calls and data get sent over the internet:

• The application layer is where SIP resides. It manages session initiation, modification, and termination.
• The transport layer is responsible for the transmission of SIP messages. It typically uses TCP or UDP protocols.
• The network layer handles IP addressing and routing, ensuring data packets reach their destination.

Now that you’re aware of the different components that make up the architecture of SIP trunking, let’s take a look at how a call moves through those elements.

Standard SIP trunking call flow

SIP trunking can be seen as a modern alternative or complement to traditional PSTN services. It provides a bridge between the old and new by connecting traditional voice services to modern VoIP systems. Here’s what a typical SIP trunking flow looks like:

1. Call initiation: A SIP client sends an INVITE request to the SIP server to start a session.
2. Call processing: The SIP server processes the INVITE and forwards it to the receiving SIP client.
3. Call establishment: Upon acceptance, a 200 OK response is sent back through the SIP server to the initiating client.
4. Media session: Voice or video data is transmitted directly between clients, bypassing the SIP server.
5. Call termination: Either party can send a BYE request, terminating the session.

The average person making a call doesn’t need to consider all these steps. However, for developers whose responsibility it is to ensure business communication is seamless, that’s a different story. If a call doesn’t make it through even one of these steps, it could be dropped, jittery, or have a ton of lag time. To make sure every call is completed satisfactorily, developers need to make sure their SIP trunking system is fully integrated with the rest of their connectivity infrastructure.

This consideration is especially for organizations switching from the Public Switch Telephone Network (PSTN) to SIP trunking or using SIP trunking as a failover system. It’s critical to understand which pieces of SIP trunking architecture help these two systems play nicely together.

Integrating SIP trunking with existing infrastructure

SIP trunking is a cost-efficient, scalable solution that offers more enhanced communication capabilities than traditional telephony options. If you have existing PSTN infrastructure you need to integrate with SIP trunking architecture, you can use the following tools to experience the benefits of SIP trunking:

VoIP gateways

A VoIP gateway is a device or software that serves as a bridge between traditional telephony networks like the PSTN and VoIP networks. It converts digital and analog voice communications, enabling seamless communication between disparate systems. It also converts between different media formats or codecs used in voice communications, ensuring compatibility between devices that may use different standards for voice encoding.

Using a VoIP gateway ensures interoperability between digital and traditional communication systems. It also helps businesses avoid disruptions to operations during or after a transition to SIP trunking.

Session border controllers (SBCs)

An SBC is a device or software application placed at the boundary of networks to manage and secure VoIP network traffic. It provides security, call routing, and protocol translation between existing telephony systems and SIP trunking services.

SBCs can also help organizations comply with regulatory requirements, such as call recording and emergency call handling, by providing the necessary control and integration capabilities. And by managing sessions efficiently, SBCs optimize network resources, reducing unnecessary load and improving the overall efficiency of the communication system.

With these two tools, your organization can leverage the advanced features of digital communication while still enjoying the familiarity of traditional infrastructure like the PSTN.

Choose Telnyx for high-quality SIP trunking architecture

SIP trunking architecture is a complex but highly efficient framework designed to deliver robust, scalable, and flexible VoIP communication solutions. By integrating SIP trunking with your existing communications infrastructure—or replacing traditional comms technology entirely—your business can access advanced communications features to reach both customers and colleagues.

Telnyx and its SIP trunking services are deeply embedded in the communications and connectivity space. With true carrier status—rather than a third-party reseller of communications technology—we can offer customers access to higher-quality calling features than the typical VoIP provider. By sending call data over our private global network rather than the public internet, customers can experience crystal-clear voice calls across the globe—all under a competitive pay-as-you-go pricing model.