SIP trunking is a service that connects your existing PBX or dialer to the public switched telephone network (PSTN) over an IP connection instead of traditional phone lines. One trunk carries multiple channels of voice calls at the same time. Capacity grows with your bandwidth, not with copper pairs or PRI circuits. The carrier on the far end handles routing, phone numbers, and STIR/SHAKEN signing.
If you run a business phone system, a call center, or a predictive dialer like VICIdial, you have either already moved to SIP trunking or you are about to. The old way — physical T1 lines carrying 23 channels over copper — is being decommissioned by every major carrier in the country. AT&T, Verizon, and Lumen have all published sunset dates for ISDN PRI service.
This guide is written by SIPNEX, an FCC-licensed telecommunications carrier that holds its own STIR/SHAKEN Service Provider certificate. We are not a reseller explaining someone else’s product. We are the carrier that provisions the trunks, signs the calls, and answers the phone when something breaks.
SIP trunking at a glance
| Traditional phone lines (analog) | ISDN PRI | SIP trunk | |
|---|---|---|---|
| Calls per connection | One call per copper pair | 23 calls per T1 circuit | Limited by bandwidth (unlimited channels on SIPNEX) |
| Signaling | Voltage changes and in-band tones | Q.931 messages on the D channel | SIP messages (INVITE, 200 OK, BYE) |
| Voice path | Analog audio on the pair | 64 kbps G.711 per B channel | RTP packets using G.711, G.729, or Opus |
| Adding capacity | Order another line | Order another T1 | Add bandwidth; no hardware order |
| Location | Tied to the building | Tied to the building | Register from any location |
| Billing | Per line per month | Per circuit per month | Per minute of use (some providers add a per-channel fee; SIPNEX does not) |
| STIR/SHAKEN Identity header | Not carried | Not carried | Carried in the SIP INVITE |
What SIP trunking actually is
SIP stands for Session Initiation Protocol, an IETF standard first published as RFC 2543 in 1999 and updated to RFC 3261 in 2002. SIP is a signaling protocol — it sets up, modifies, and tears down communication sessions. It does not carry the actual voice audio. That job belongs to RTP (Real-time Transport Protocol). Think of SIP as the phone ringing and someone answering. RTP is the conversation itself.
A trunk in telecom is the connection between your phone system and the carrier network. Historically this was a physical bundle of copper wires. A T1 line delivered a Primary Rate Interface carrying exactly 23 voice channels and one signaling channel over ISDN. If you needed 24 simultaneous calls, you ordered a second T1. The connection was physical, fixed, and expensive.
SIP trunking replaces that physical connection with a virtual one running over IP. Your PBX, contact center platform, or dialer software registers to the carrier SIP proxy server over the internet, an MPLS circuit, or dedicated fiber. When you place a call, your system sends a SIP INVITE message to the carrier. The carrier authenticates your trunk and validates your caller ID. It signs the call with STIR/SHAKEN attestation and routes it to the public switched telephone network.
The fundamental shift: SIP trunks are not tied to physical infrastructure. You do not order a circuit or wait for a technician. On a carrier like SIPNEX, concurrent channels are unlimited — your capacity is determined by your bandwidth and server resources, not by leased physical lines.
How SIP trunking works
Here is what happens when your system places an outbound call through a SIP trunk:
Your PBX or dialer constructs a SIP INVITE containing the called number in the Request-URI, your caller ID in the From header, and codec preferences in the SDP body. The SDP lists which audio codecs your system supports. The usual candidates are G.711u (uncompressed, highest quality, about 85 kbps per call), G.729 (compressed: 8 kbps of audio, about 32 kbps on the wire with headers), and Opus (modern, adaptive bitrate).
The carrier receives your INVITE and authenticates the trunk. IP-based authentication checks whether the INVITE came from a whitelisted IP address. Digest authentication challenges your system with a 407 response and your system replies with credentials. After authentication, the carrier checks your account standing and validates the calling number against your authorized CID list.
Next comes the step that separates a direct carrier from a reseller. SIPNEX holds its own STIR/SHAKEN Service Provider certificate. We attach a cryptographic Identity header to the SIP INVITE with an attestation level. If the calling number is a DID we provisioned to you and you have verified authority to use it, we sign at A-level — full attestation. Most resellers cannot do this. They pass your call upstream to another carrier, and that carrier signs at B-level because it has no direct relationship with you.
The carrier then routes the call to the terminating carrier via the PSTN or direct IP peering. When the called party answers, RTP media streams are established and voice audio flows as UDP packets between your system and the carrier media gateway.
Inbound calls work in reverse. The carrier receives a call from the PSTN destined for one of your DIDs. It looks up your SIP trunk endpoint and sends a SIP INVITE to your system.
SIP trunking vs PRI
PRI delivers 23 voice channels over a physical T1 line using ISDN signaling at $300 to $500 per month. Need 24 simultaneous calls? Buy a second PRI. The math is rigid and the infrastructure is physical. Our PRI explainer covers the 23B+D design in detail.
SIP trunking eliminates every one of those constraints. On SIPNEX, channels are unlimited. You pay per minute of actual usage. Need 10 concurrent calls today and 500 tomorrow for a campaign? Scale instantly with no hardware order, no technician visit, no waiting.
Beyond channel flexibility, SIP delivers capabilities PRI cannot. Geographic independence: register from any location. DID portability: add local numbers in any area code instantly. Disaster recovery: automatic failover to backup endpoints. And carrier-level features like STIR/SHAKEN attestation that simply do not exist on PRI.
The cost comparison is stark. A single PRI at $400 per month gives you 23 fixed channels. That same $400 at SIPNEX wholesale rates buys approximately 25,000 to 80,000 minutes depending on your volume tier — far more value and zero idle capacity charges. The full migration path is in our SIP vs PRI comparison.
IP trunk, SIP trunk, IP SIP trunk: same thing?
In practice, yes. “IP trunking” is the umbrella term for any trunk that carries calls over an IP network instead of dedicated copper. SIP trunking is IP trunking that uses SIP as its signaling protocol. Because SIP became the dominant standard, the two terms have collapsed into each other.
A provider advertising an “IP SIP trunk” is selling exactly the product this guide describes. The doubled name spells out both the transport (IP) and the signaling (SIP). If a quote lists an IP trunk without naming the protocol, ask. If the answer is anything other than SIP, everything about codecs, attestation, and interoperability above needs re-checking.
Benefits of SIP trunking
Most buyers hear the same benefits of SIP trunking in every sales pitch. Here is the practical meaning of each.
Lower cost per call. On a usage-billed trunk like SIPNEX you pay per minute of connected talk time and nothing for idle channels. Some providers still bill per channel per month, so check the quote. Traditional phone lines bill per line whether anyone is talking or not. Eliminating physical line charges is the obvious saving, but the bigger number comes from usage-based billing. Most businesses vastly overprovision PRI capacity because running out of channels means busy signals. With unlimited SIP channels, you pay for what you use.
Capacity that follows call volume. One trunk carries multiple channels. Add channels when a campaign starts and drop them when it ends. Nothing physical changes. A tax preparation firm needing 200 channels in March and 20 in July pays for exactly that. No contracts locking in peak capacity year-round.
Long distance stops being a line item. On a SIP trunk, a domestic call is rated per minute by its destination. Distance by itself does not enter the formula. The local-versus-long-distance split that traditional phone lines billed separately has largely dissolved as a retail product.
Numbers from anywhere. Provision local DIDs in any US area code and point them at one trunk. A Dallas office can answer a Miami number without a Miami phone line.
Business continuity built in. If your site loses power or internet, inbound calls can reroute to a backup endpoint the carrier holds on file.
Carrier-level features. STIR/SHAKEN attestation, SMS on the same trunk, and carrier-level failover all require IP infrastructure. PRI supports none of them. Call recording controls and programmable routing ride the same SIP infrastructure.
Room for a dialer. Unlimited concurrent channels means your predictive dialer can scale to its full potential. A carrier that caps you at 50 channels is capping your revenue.
What you need before you order a SIP trunking service
A SIP trunking service has few moving parts, but each one must be in place before the first call. Run this list before you sign.
1. A system that speaks SIP. Your existing PBX or dialer must support SIP trunks. 3CX, FreePBX, Asterisk, FusionPBX, Yeastar, Avaya IP Office, Grandstream, and VICIdial all do. An older analog or digital PBX can still use a trunk. A SIP gateway sits between the PBX trunk ports and your network and converts the calls. Our PBX SIP trunking page lists tested platforms.
2. Bandwidth for voice calls. Each G.711 call needs about 85 kbps in each direction. Twenty concurrent calls need about 1.7 Mbps each way. Two hundred need about 17 Mbps. Reserve that capacity for voice. Do not share it with backups or video uploads. The G.711 vs G.729 comparison covers the codec math.
3. A clean network path. Voice tolerates less than data does. Keep one-way latency under 150 ms, jitter under 30 ms, and packet loss under 1 percent. Our VoIP quality guide explains how to measure each one.
4. Firewall rules and no SIP ALG. Open UDP 5060 to your carrier’s proxy, or TCP 5061 for TLS, plus the RTP port range your PBX uses. Then turn off SIP ALG on the router. That one router feature causes more one-way audio than any other setting. See what SIP ALG does and the SIP trunk configuration guide.
5. Phone numbers. Order new DIDs or port your existing numbers. Porting uses a Letter of Authorization and runs in the background while you dial on new numbers.
6. A 911 plan. Every trunk needs a Registered Location for each site. If you run a PBX, Kari’s Law and RAY BAUM’S Act apply to you as the system operator. More on that below.
Sizing a trunk for your call volume
Call volume and concurrent calls are different numbers. Call volume is how many phone calls you place or receive in a day or a month. Concurrent calls is how many are up at the same moment. Trunk sizing follows the second number. Billing follows the first.
One workable method: pull a month of call records from your current system. Find the busiest fifteen-minute window. Count the calls active at once. That peak, plus headroom for growth, is the channel count you need.
Then convert channels to bandwidth:
| Concurrent calls | G.711 (about 85 kbps each) | G.729 (about 32 kbps each) |
|---|---|---|
| 10 | ~0.85 Mbps each direction | ~0.32 Mbps each direction |
| 50 | ~4.3 Mbps | ~1.6 Mbps |
| 200 | ~17 Mbps | ~6.4 Mbps |
| 500 | ~43 Mbps | ~16 Mbps |
Dialers add one more number: calls per second. A predictive dialer front-loads attempts at the start of each pacing cycle. Ask the carrier what CPS it accepts before you sign. Our telecom acronym guide defines CPS alongside ASR and ACD.
On SIPNEX, channels are unlimited. Bandwidth and server capacity are the only ceilings.
SIP trunking vs hosted PBX: which model fits
SIP trunking gives you the pipe. You run the phone system. Hosted PBX (also called cloud PBX) gives you the pipe and the phone system together. The provider runs call routing, voicemail, and extensions in the cloud.
The split usually follows headcount and control. A small office with no telecom staff is better served by hosted PBX. A call center that needs codec control, custom routing, and unlimited channels needs a trunk into its own PBX or dialer. If you already own a PBX that works, a trunk keeps it working. Nothing on the desk changes.
SIPNEX sells both SIP trunking and hosted PBX, so the advice here is not tied to one model. The full decision guide is hosted PBX vs SIP trunking.
Business continuity on a SIP trunk
Traditional phone lines fail with the building. Cut the copper pair and the numbers on it go dark until a technician repairs it. A SIP trunk separates the numbers from the site.
Three layers cover most outages:
- Carrier-side failover. Register a backup endpoint with the carrier. If the primary registration drops, inbound calls reroute to it.
- A second internet path. A cellular or secondary ISP connection keeps the trunk registered when the main line fails.
- A second carrier. Operators running dialers keep a backup trunk configured and tested. The VICIdial community treats two carriers as baseline practice. Our backup carrier guide walks through the setup.
One rule from that guide applies everywhere: complete a test call through the backup path before the day you need it. A failover that has never passed a call is a hope, not a plan.
E911 on a SIP trunk
A SIP trunk does not know where your phones sit. A landline did — the copper terminated at one address. On VoIP, a 911 call is routed by the Registered Location you file with the carrier. Keep one address per site and update it on every move.
If you run a PBX behind the trunk, you are also the multi-line telephone system operator. Under Kari’s Law, 911 must be dialable directly, with no prefix, and someone on site must be notified whenever a 911 call goes out. RAY BAUM’S Act requires dispatchable location: the street address plus the suite, floor, or room. SIPNEX provisions E911 in the 50 US states with every trunk and DID deployment. Our E911 for VoIP guide covers the turn-up checklist.
Choosing a provider: carrier vs reseller
This is where most buyers get it wrong. The market is full of companies calling themselves SIP trunk providers that are actually resellers sitting on top of an actual carrier.
A reseller purchases trunk capacity from an upstream carrier, marks up the rate, and sells it to you. They do not control the network, cannot control attestation level, and cannot provision DIDs directly. When something breaks, they open a ticket with their carrier and wait.
A direct carrier like SIPNEX holds its own FCC license, its own STIR/SHAKEN SP certificate, and files its own FCC Form 499. What this means for you: direct A-level attestation because your customer relationship is with the signing carrier. Twenty-four hour provisioning because we provision directly. Transparent pricing because we set the rates — there is no hidden markup layer.
Ask your current provider one question: do you hold your own STIR/SHAKEN Service Provider certificate, or do you inherit attestation from an upstream carrier? If they hesitate, you have your answer. Our provider evaluation checklist adds six more questions.
SIP trunking costs in 2026
Wholesale per-minute rates range from $0.005 to $0.030 depending on volume commitment and calling destination. Metro areas like Dallas, Houston, New York, and Los Angeles are cheapest. Rural areas cost more due to higher termination fees from incumbent local exchange carriers.
Wholesale carriers typically bill in 6-second increments after a short minimum — SIPNEX runs 12/6 — so a 47-second call bills as 48 seconds. Platform providers commonly bill full 60/60 minutes, and quality routes across the industry are drifting that way — always ask.
At SIPNEX, operators under 100,000 minutes per month pay $0.025 to $0.030 per minute. At 1 to 5 million minutes, rates drop to $0.008 to $0.012. At 10 million plus, rates reach as low as $0.005. No setup fees, no per-channel fees, no contracts.
Terms you will see on a SIP trunk quote
- Channel — one concurrent call on the trunk. Twenty channels means twenty phone calls at once.
- DID — a phone number that routes directly to a specific endpoint. See our DID guide.
- CPS — calls per second, the rate of new call attempts the carrier accepts.
- Codec — how audio is encoded: G.711 for quality, G.729 for low bandwidth, Opus for adaptive bitrate.
- Registration vs IP authentication — credentials your PBX sends on each request, or a fixed public IP the carrier trusts.
- Attestation — the A, B, or C level the signing carrier attaches to each call under STIR/SHAKEN.
- 12/6 billing — a 12-second minimum and 6-second increments after it. This is how SIPNEX bills dialer traffic.
- SBC — session border controller, a SIP-aware device between your PBX and the carrier that normalizes signaling and handles encryption.
The VoIP glossary covers 60 more.
Frequently asked questions
What is the difference between SIP trunking and VoIP?
VoIP is the broad category — any voice communication over IP networks. SIP trunking is a specific implementation within VoIP using the Session Initiation Protocol to connect a PBX or dialer to a carrier. Other VoIP protocols exist (H.323, MGCP, proprietary ones) but SIP is the dominant standard. If someone offers VoIP service, ask what protocol. If it is SIP, you are getting SIP trunking.
How many SIP trunks do I need?
On a carrier like SIPNEX where channels are unlimited, you need one trunk that handles as many simultaneous calls as your bandwidth supports. Each G.711 call uses about 85 kbps. A 100 Mbps connection can theoretically support over 500 concurrent calls. The question is not how many trunks but how much bandwidth you have allocated for voice traffic.
Is SIP trunking secure?
SIP supports TLS for signaling encryption and SRTP for media encryption. SIPNEX supports both. Beyond encryption, security includes IP-based access controls, digest authentication, rate limiting, and anomaly detection. On your end, firewall your SIP ports, use strong credentials, and monitor for unexpected international calls — the most common sign of a compromised trunk.
Can I keep my existing phone numbers?
Yes. FCC regulations guarantee number portability. You submit a Letter of Authorization to SIPNEX, we submit the port request, and your numbers cut over in 7 to 14 business days. You can start dialing on new SIPNEX DIDs immediately while the port completes in the background. There is no gap in service.
How long does setup take?
SIPNEX provisions trunks in 24 hours from signed agreement. We send you SIP proxy credentials, you configure your PBX or VICIdial instance, register, and start dialing. Resellers typically take 5 to 10 business days because they wait on their upstream carrier. The difference is we provision directly.
What happens if my internet goes down?
Configure a failover endpoint with SIPNEX. If your primary SIP registration drops, inbound calls automatically reroute to a backup destination — a second office, cell phone, cloud IVR, or voicemail. For mission-critical operations, run a backup internet connection or host your PBX in a data center with redundant connectivity.
What hardware connects an analog PBX to a SIP trunk?
A SIP gateway. It connects to the PBX’s analog or digital trunk ports on one side and to your IP network on the other. The PBX keeps working as it always did; the gateway converts its calls to SIP and sends them to the carrier. Some PBX models also accept an IP trunk card that does the same job internally. Either path is a common transition step off retired copper and PRI before a full PBX replacement.
Does a SIP trunk bill long distance calls differently from local calls?
On a usage-billed wholesale trunk like SIPNEX, no. A domestic call is rated per minute by its destination, not by distance. Distance by itself is not the pricing factor; the destination’s termination cost is. A metro destination is usually cheaper than a rural one whether the call crosses the street or the country. International destinations carry their own per-country rates. The separate long distance charge that traditional phone lines added on top of the monthly line fee has largely dissolved on trunks; some bundled retail plans still carry it as a plan component, so check the quote.
Is an IP SIP trunk different from a regular SIP trunk?
No. The two names describe the same product. “IP” names the transport and “SIP” names the signaling protocol. A quote for an IP SIP trunk is a quote for a SIP trunk that runs over an IP network, which every SIP trunk does. Confirm the codec list, the authentication method, and the attestation level. The label itself tells you nothing else.
How does a SIP trunk support business continuity?
By separating your numbers from your building. Inbound calls follow the trunk’s registration, not a copper pair, so the backup endpoint the carrier holds on file takes over when the site drops. Outbound continues from any location where your PBX or dialer can register. Add a second internet path and, for dialer operations, a second carrier — then prove the backup path with a test call before the day you need it.
SIPNEX is an FCC-licensed carrier with its own STIR/SHAKEN Service Provider certificate. We provide SIP trunks built for operators who dial at volume — unlimited channels, direct A-level attestation, and support from engineers who run VICIdial. Request a dialer-grade trunk or see our published rates.
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