SIP Trunking Explained: What It Is, How It Works and When to Use It
Quick Answer: SIP trunking explained simply: a SIP trunk is a virtual phone line that connects your business phone system (PBX) to the public telephone network over an internet or private data connection instead of copper wires. A provider gives you phone numbers and a set number of simultaneous call paths, and your PBX uses the SIP protocol to set up each call through that connection. You keep your own phone system and features, while the provider handles reaching every landline and mobile number in the world.
Key Points
- A traditional PRI line carries exactly 23 voice channels, while a SIP trunk can be sized to the number of simultaneous calls you actually need.
- One standard G.711 call uses about 87 kbps each way on Ethernet, so ten calls at once need under 1 Mbps of steady upload and download.
- AT&T plans to retire most of its copper network outside California by the end of 2029, which is pushing many PRI users toward SIP.
- The old Asterisk command “sip show peers” no longer exists in Asterisk 21 and later, because the legacy SIP channel driver was removed in favor of PJSIP.
- A session border controller (SBC) is the usual security and translation point between your PBX and a SIP trunk provider.
What is SIP trunking?
SIP trunking is a service that delivers phone lines to a business phone system as data, using the Session Initiation Protocol. SIP is the abbreviation behind the name: it is the IETF standard, defined in RFC 3261, that starts, manages and ends voice and video sessions.
The “trunk” part comes from old telephone terminology, where a trunk was a shared bundle of lines connecting a company switchboard to the phone company. A SIP trunk keeps that idea but replaces the physical bundle with a logical connection. When people ask for the SIP trunk provider meaning, they mean an internet telephony service provider (ITSP) that sells these trunks, assigns DID (direct inward dial) numbers, and connects your calls to the public switched telephone network (PSTN).
How does SIP trunking work?
SIP trunking works by letting your PBX send call requests to a provider’s SIP servers, which then route each call to the right destination. The audio travels separately as RTP packets once the call is set up.
Here is the flow for an outbound call:
- A user dials a number on a desk phone or softphone connected to the PBX.
- The PBX checks its outbound route and sends a SIP INVITE message to the trunk provider.
- The provider authenticates the request, either by username and password registration or by recognizing your static IP address.
- The provider routes the call to the PSTN or to another IP network.
- Both sides agree on a codec, and voice packets flow until someone hangs up and a SIP BYE ends the session.
A simple picture of the setup looks like this:
Desk phones and softphones
↓
Your PBX (Asterisk, FreePBX, Cisco, Avaya or Teams Phone)
↓
Session border controller and firewall
↓
Internet or private data link
↓
SIP trunk provider
↓
Public telephone network: landlines and mobilesInbound calls run the same path in reverse, ending at the right extension.
SIP trunk example
Imagine a 40 person accounting firm that rarely has more than eight people on outside calls at once. With SIP trunking, it buys eight concurrent call paths, ports its 40 existing numbers as DIDs, and points them at its on premises PBX. In tax season it can add more paths through the provider’s portal instead of waiting for a new circuit. In practice, sizing by peak simultaneous calls rather than by headcount is the single biggest lever on SIP trunk cost.
SIP trunking vs VoIP: what is the difference?
VoIP is the broad technology of carrying voice over IP networks, and SIP trunking is one specific VoIP service. Every SIP trunk is VoIP, but plenty of VoIP, such as a WhatsApp call or a hosted cloud phone service, involves no SIP trunk you manage.
The practical difference is ownership. With a hosted VoIP or UCaaS service, the provider runs the phone system for you. With SIP trunking, you run your own PBX and buy only the connection to the outside world. If you want the basics first, our guide on what VoIP is and how it works covers how voice turns into packets.
SIP trunk vs PBX
A PBX and a SIP trunk are not alternatives; they do different jobs. The PBX is the phone system that runs extensions, menus, voicemail and call routing inside your company. The SIP trunk is the external connection that lets that PBX reach outside numbers.
| PBX | SIP trunk | |
|---|---|---|
| Role | Internal phone system | External phone line service |
| Handles | Extensions, IVR menus, transfers, voicemail | Outside calls, DIDs, caller ID, PSTN access |
| Examples | Asterisk, FreePBX, Cisco, Avaya, Teams Phone | Service from a carrier or ITSP |
| You need it when | You want your own call features and control | Your PBX must make or receive outside calls |
SIP trunk vs PRI
SIP trunks and PRI lines both connect a PBX to the phone network, but PRI does it over a dedicated T1 circuit with fixed capacity. In the US, a PRI provides 23 bearer channels for voice plus one channel for signaling, each running at 64 kbps.
| Feature | SIP trunk | PRI (T1) |
|---|---|---|
| Delivery | Internet or private IP link | Dedicated copper or T1 circuit |
| Capacity | Any number of call paths | Fixed 23 voice channels per circuit |
| Scaling | Change in a portal | Order another circuit |
| Failover | Reroute calls to another site or mobile | Limited to the physical line |
| Main risk | Depends on network quality and security | Hardware cost and copper retirement |
What most articles miss is the timing pressure. FCC rule changes in 2025 and 2026 made it faster for carriers to discontinue legacy copper services, so the decision to leave PRI is increasingly made for you. Our guide to essential IT infrastructure for small businesses explains how to plan for old phone lines going away.
SIP trunk configuration basics
A SIP trunk configuration needs the same core details on almost every PBX. Gather these from your provider before you start:
- Server address: the provider’s SIP domain or IP and port, often 5060 for UDP or 5061 for TLS.
- Authentication: registration credentials, or the static IP the provider will whitelist.
- Codecs: usually G.711 (called ulaw in the US), with G.729 or Opus where supported.
- Numbers: your DIDs and the caller ID format the provider expects.
- Routes: inbound routes that map DIDs to extensions, and outbound routes that send calls to the trunk.
- Firewall and NAT: open only the provider’s IP ranges and set RTP port ranges correctly.
- 911 settings: a registered dispatchable address for each location.
Most people discover that one way audio, where only one side can hear, comes from NAT or firewall settings rather than the provider. On Microsoft Teams, the equivalent setup is Direct Routing, which connects a certified SBC to Teams Phone. If you would rather not handle this alone, co managed IT services let your team share the work with outside specialists.
“sip show trunk” and checking trunk status in Asterisk
Older Asterisk guides use commands like “sip show peers” and “sip show registry” to check trunks. Those belong to the legacy SIP channel driver, which Asterisk deprecated in version 17 and removed in version 21. On current systems, use “pjsip show registrations” to see whether your trunk is registered and “pjsip show endpoints” to see its status.
SIP trunks for VICIdial and call centers
VICIdial, the open source contact center suite, runs on Asterisk and connects to carriers through SIP trunks. Predictive dialers place many calls at once, so channel limits, carrier dialing rules and caller ID authentication matter far more than they do for a normal office.
How many SIP trunk channels and how much bandwidth do you need?
You need one channel for every call you want to run at the same time, not one per employee. Check your phone system’s reports for peak concurrent calls over a few busy weeks, then add a margin.
For bandwidth, Cisco’s tables put a G.711 call at 87.2 kbps each way on Ethernet and a compressed G.729 call at 31.2 kbps. Because of this, latency, jitter and packet loss usually decide call quality more than raw speed does.
Pros, cons and common misconceptions
SIP trunking gives you flexible capacity, number portability between sites, and easy failover. However, it is not automatically cheaper or safer. Call quality depends on your network, and an exposed PBX invites toll fraud, where criminals hijack a system to place expensive calls. Lock it down with strong passwords, IP restrictions and the small business IT security best practices you use elsewhere. Also confirm your provider supports STIR/SHAKEN caller ID authentication, which the FCC requires of voice providers to fight spoofed robocalls.
FAQ
What does SIP stand for in SIP trunking?
SIP stands for Session Initiation Protocol. It is the signaling standard that sets up, changes and ends calls between your PBX and the provider.
Is a SIP trunk a physical line?
No. A SIP trunk is a logical connection that runs over your existing internet or private data circuit, so nothing new needs to be wired to your building.
How many calls can one SIP trunk handle?
A SIP trunk handles as many simultaneous calls as you buy channels for, limited only by your bandwidth. Providers usually sell capacity per channel or as unlimited plans with a concurrency cap.
Do I need a PBX to use SIP trunking?
Yes. A SIP trunk connects to a PBX or similar system, such as an on premises IP PBX, a virtual PBX in the cloud, or Microsoft Teams through Direct Routing.
Can I keep my phone numbers when moving to SIP trunking?
Usually yes. US number portability rules let you move existing numbers to a new provider, although the process can take days or weeks.
Is SIP trunking secure?
SIP trunking can be secure when you encrypt signaling with TLS and media with SRTP, restrict access by IP, and use a session border controller. Unsecured systems are a common target for fraud.
What is a session border controller?
A session border controller is a device or software that sits at the edge of your network to secure SIP traffic, handle NAT, and translate between differing SIP implementations.
Does SIP trunking work with Microsoft Teams?
Yes. Microsoft Teams supports SIP trunks through Direct Routing, which requires a Microsoft certified session border controller, or through Operator Connect with participating carriers.
What happens to SIP trunk calls if the internet goes down?
Calls on that connection stop, but most providers let you reroute numbers automatically to mobile phones, another office or a backup trunk. A second internet link adds more resilience.
Who uses SIP trunking most?
Businesses that already own a PBX, contact centers that need many simultaneous calls, and companies replacing PRI lines are the most common users.
Can I get a SIP trunk outside the US?
Yes. Carriers in most countries sell SIP trunks, though rules on numbers and emergency calling differ. If you are comparing a local offer, such as from PTCL in Pakistan, ask about channel limits, codecs and number porting.
What is SIPconnect?
SIPconnect is a technical recommendation from the SIP Forum that standardizes how IP PBXs and service provider networks connect over SIP trunks. Version 2.0 was ratified in 2017.
Conclusion
SIP trunking lets you keep the phone system you know while replacing aging lines with flexible, data based capacity. Start by measuring your peak simultaneous calls, check your network’s quality, and plan the move before your carrier sets the deadline. Our cloud migration checklist is a useful companion for planning the switch. For the technical detail behind it, the IETF’s RFC 3261 specification for SIP defines the protocol itself, and Microsoft’s Direct Routing planning guide shows what connecting a SIP trunk to Teams Phone requires.
