SIP Trunk 101: What is Call Origination?

SIP Trunk 101: What is call origination?

Knowing how voice telephony call origination works is straightforward: someone picks up their device or softphone, enters your phone number, and you pick up the phone. In essence, call or voice origination is an incoming call. Of course, the actual "how" is a bit more complicated.

A quick POTS history

With plain old telephone service (POTS) or more commonly known dial-up circuit architecture, the user must "seize" the line or circuit before a call can be established. Once initiated, the caller hears a dial tone and can begin entering the phone number.

Depending on the distance between the caller and callee, a Central Office containing Class-5 or End Office Switching equipment routes the call to the recipient's Central Office through a series of Class-4 or Tandem switches.

This entire infrastructure is typically referred to as the Public Switched Telephone Network (PSTN). Eventually, the receiver will ring if the recipient's line is not seized—if it is, the call initiator gets a busy signal.

The call origination process

Today, VoIP calls use packet switching technology, which digitizes the voice and sends the data to the recipient. The call origination then is the point where the call started—typically on the PSTN—as it is transferred from the Class 4 Tandem Switch by an Internet Telephone Service Provider (ITSP) over the Internet to a destination point.

More specifically, the media is transferred via a Session Initiation Protocol trunk or SIP trunk—basically digital telephone lines that VoIP providers use to connect your private branch exchange (PBX) or telephone system.

The Session Initiation Protocol (SIP) was standardized by the Internet Engineering Task Force (IETF) in RFC 3261, establishing the application-layer control protocol for creating, modifying, and terminating multimedia sessions. SIP works in concert with other protocols to enable Internet endpoints to discover one another and agree on session parameters.

How SIP call origination works

Here's what happens during a typical call origination flow:

1. Call initiation: A caller on the PSTN dials your business number
2. PSTN to IP conversion: The ITSP receives the call and converts analog signals to digital packets
3. SIP signaling: An INVITE request is sent to your SIP server or PBX
4. Media negotiation: Session Description Protocol (SDP) establishes codec parameters
5. Call delivery: Voice media streams via RTP to your endpoint
6. Call establishment: Upon answer, a 200 OK response completes the session setup

For more detail on this architecture, see our guide on SIP trunking architecture and how the components work together.

SIP trunking vs. traditional PRI: a comparison

Understanding the differences between SIP trunking and legacy Primary Rate Interface (PRI) connections helps clarify why modern businesses are migrating to IP-based communications. According to Mordor Intelligence, the SIP trunking market stands at USD 73.14 billion in 2025 and is projected to reach USD 157.91 billion by 2030.

For a deeper dive into these differences, read our SIP Trunk vs. PRI comparison.

"SIP trunking has moved from early adoption to mainstream business necessity. The shift toward hybrid work models has accelerated adoption, as SIP trunking enables seamless communication for distributed teams without geographic limitations."

— SIP.US Industry Trends Report, 2025

Technical specifications for SIP trunking

When evaluating SIP trunk providers or configuring your own deployment, understanding the technical parameters ensures optimal call quality and compatibility.

Ports, codecs, and SIP response codes

For codec selection guidance, see our complete VoIP codec list with bandwidth, quality, and licensing details.

The Telnyx SIP response codes guide provides detailed explanations of each response code and troubleshooting steps.

Choosing the right call origination provider

The key to choosing the right VoIP origination provider depends on several factors and caveats:

Price

Price is a huge consideration that's typically at the top of the list. Different providers have different price points depending on your needs and the services you're looking for. There are also pricing structure differences—next-generation VoIP carriers provide à la carte pricing versus more traditional providers that will typically make you sign a contract.

According to Business News Daily, SIP trunk pricing typically ranges from $15-30 per channel monthly for unlimited plans, with metered options at $0.01-0.03 per minute. Most businesses achieve positive ROI within their first year of implementation.

Check out Telnyx Elastic SIP pricing for transparent, pay-as-you-go rates.

Functionality

Functionality and interoperability could also make or break your decision when choosing a voice origination carrier. If receiving calls is an immediate need for your organization, how quickly will you be able to integrate your existing PBX with the VoIP provider you're considering?

And what about features and usage? Make sure you try out their platform first before making any final commitments. Check out how intuitive the system is and play around with the features to gain a feel of how useful the platform will be for you.

Telnyx offers:

• SIP Trunking — Deploy elastic SIP trunks on our private, multi-cloud network with pay-as-you-go pricing
• Voice API — Programmatic call control for custom voice applications
• Enterprise integrations with Microsoft Teams, Zoom, and Genesys

For setup guidance, our SIP Trunking Quickstart walks through creating connections, configuring authentication, and assigning DIDs.

Overall quality

Take the time to find out how each voice origination provider differs and which one can best accommodate your needs. 24/7 support, for instance, may be something you absolutely must have, and not everyone offers that service.

And beyond support, look into the actual network the provider offers that will best suit your inbound needs: do they have a private backbone that will take your media off the public internet? Will they have security features that will protect your data?

"The biggest pitfall in SIP deployments isn't the technology, it's inadequate planning. Many times, the network has been assumed adequate but hasn't been evaluated for its ability to support voice traffic. Take into consideration network service SLAs, packet loss, jitter, and latency metrics that will affect real-time voice traffic and user experience."

— Gary Audin, Enterprise Communications Consultant, Telecom Reseller

Key quality considerations include:

• Network infrastructure: Does the provider own their network or rely on third parties?
• Redundancy and failover: What happens if a data center goes down? Telnyx's SIP connection failover explains how automatic retries work.
• Call quality monitoring: Tools like CDR reporting let you analyze metrics like post-dial delay and MOS scores.
• Security: Look for TLS encryption for signaling and SRTP for media, plus fraud detection capabilities.

Industry solutions

Different industries have unique requirements for call origination and voice communications:

• Healthcare — HIPAA-compliant voice AI for patient scheduling and appointment reminders
• Contact centers — Omnichannel carrier-grade communications with IVR and call routing

Getting started with SIP trunking

Ready to modernize your voice infrastructure? Here's a straightforward path:

1. Assess your current setup: Inventory your existing circuits, DIDs, and concurrent call requirements
2. Calculate bandwidth needs: Use Telnyx's VoIP bandwidth guidelines based on your expected call volume
3. Choose your connection type: Credential-based, IP-based, or FQDN authentication via SIP connection settings
4. Configure your PBX: Follow our Bring Your Own Carrier guides for platforms like Asterisk, 3CX, FreePBX, and more
5. Test and optimize: Use the Mission Control Portal debugging tools to verify call quality

For step-by-step instructions, see our complete guide on how to set up a SIP trunk.

Additional resources

Telnyx guides

• SIP vs. VoIP: Key differences explained
• Understanding SIP trunk channels
• What is PBX and how can it help scale your business?

External references

• RFC 3261 — SIP: Session Initiation Protocol (IETF)
• SIP Trunking Market Size & Forecast (Mordor Intelligence)
• Session Initiation Protocol — Wikipedia
• VoIP Codec comparison and selection (VirtualPBX)
• SIP Migration best practices (No Jitter)

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Related articles:

• PSTN termination: Understanding the basics
• VoIP origination: How numbers work
• Inbound SIP trunking for your business

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