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.
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, validates your caller ID, 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 — typically G.711u (uncompressed, highest quality, about 85 kbps per call), G.729 (compressed, about 8 kbps), or 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 because they pass your call upstream to another carrier who signs at B-level since that carrier 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, 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.
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.
Why businesses switch
Cost reduction is immediate. 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.
Scalability matches business reality. 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.
Features transform what your phone system can do. STIR/SHAKEN attestation, call recording controls, programmable routing, SMS on the same trunk — all require SIP infrastructure.
For call centers specifically: 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.
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.
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.
Most carriers bill in 6-second increments. A 47-second call bills as 48 seconds. Some carriers still bill in full-minute increments — 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.
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.
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.
SIPNEX
FCC-licensed carrier with its own STIR/SHAKEN SP certificate. Operator-owned. SIP trunks built for operators who dial at volume.