FCC LICENSED CARRIER ·499 FILER ·STIR/SHAKEN CERTIFIED
SIPNEX
Services · SIP Trunking

SIP trunks for operators who dial like they mean it.

Buy direct from the carrier — not a reseller marking up someone else’s trunk. Unlimited channels, A-level attestation signed with our own STIR/SHAKEN certificate, and support from engineers who know what sip.conf is.

Request Dialer-Grade Trunk → See Pricing
  • Who it’s for: dialer and contact-center operators pushing serious outbound volume.
  • What you get: unlimited channels, direct A-level attestation, DIDs live in minutes.
  • How fast: a live trunk within 24 hours of a signed agreement.
Definition

What does a SIP trunk provider do?

A SIP trunk provider connects your phone system to the public telephone network over the internet. It carries your calls, issues your phone numbers, and signs your caller ID. Some providers are licensed carriers that run their own network. Others are resellers who mark up a carrier’s trunk. SIPNEX is the carrier: FCC-licensed, a registered 499 filer, with our own STIR/SHAKEN certificate.

The provider you pick shows up in your numbers, not your contract. Attestation moves answer rates. Channel caps decide how hard your dialer can push. Provisioning speed sets how fast a new campaign gets numbers. So the first question is not the rate — it is whether you are buying from the network or from a middleman.

What you get

Carrier-grade, not reseller-grade.

Unlimited concurrent channels

No artificial caps. Your trunk scales with your instance — not ours. Run 10 or 10,000 concurrent calls.

Direct A-level STIR/SHAKEN

We sign every outbound call with our own SP-KI certificate. Not downstream, not rewrapped. Direct A-level from the originating carrier.

Instant DID provisioning

Provision local, toll-free, and vanity numbers across all North American area codes. New DIDs active in minutes, not days.

VICIdial-native configuration

We provide tested sip.conf templates, codec recommendations, and direct support from engineers who run VICIdial themselves.

24-hour trunk provisioning

From signed agreement to live trunk in one business day. Complex porting takes longer, but your initial setup doesn’t wait.

Transparent per-minute pricing

Published rates. No quote-gated pricing games. You see what you pay before you talk to anyone.

Side by side

Carrier vs reseller: two kinds of SIP trunk provider.

Every SIP trunk provider either runs the network or resells someone else’s. Here is what that difference means, line by line.

CAPABILITY SIPNEX TYPICAL RESELLER
FCC-licensed carrier ✓ Direct Reselling upstream
STIR/SHAKEN attestation A-level (own cert) B-level (inherited)
Concurrent channel limit Unlimited 10–50 typical
DID provisioning speed Minutes 1–5 business days
VICIdial support Native (we run it) Generic / unsupported
Pricing transparency Published rates Quote-gated
Trunk provisioning 24 hours 5–10 business days
Uptime SLA 99.999% 99.9% if offered
Buyer’s checklist

How to choose a SIP trunk provider.

Six checks separate a carrier from a middleman, and none of them require a sales call. A provider that hides its rates behind a quote form is telling you something — compare our published SIP trunk rates first.

  1. Verify carrier status. Look the provider up in the FCC Form 499 filer database and the Robocall Mitigation Database. Both are public, and neither can be faked.
  2. Ask who signs STIR/SHAKEN. Direct A-level attestation requires the provider’s own SP certificate, not one borrowed from an upstream.
  3. Check the concurrent-channel cap. Ask what happens at 50 channels, then at 500. The answer tells you whose network you are really on.
  4. Time the provisioning. Ask how long a new trunk and a new DID take. Minutes point to a carrier; a week points to a ticket queue.
  5. Confirm codec and platform support. G.711, G.729, and Opus should all be on the menu, along with your PBX or dialer platform.
  6. Test the support line. Ask one technical question before you sign. Whoever picks up is your support experience from day one.
What you pay for

How SIP trunk providers bill phone calls.

Every SIP trunk bill has the same parts. A per-minute rate on connected phone calls. A monthly fee per phone number. Pass-through regulatory charges such as USF and E911 fees. Some providers add a monthly fee per concurrent channel or a flat trunk fee on top. SIPNEX does not charge per channel — channels are unlimited — and our rates sit on the pricing page, not behind a quote form.

The billing increment matters as much as the rate. Twilio and Telnyx bill voice at 60/60: a 60-second minimum, then 60-second steps. A 20-second call pays for a full minute. SIPNEX bills dialer traffic at 12/6 — 12 seconds minimum, then in 6-second steps — so the same call bills 24 seconds. True 6/6 is reserved for traffic averaging about 15 seconds ALOC.

Short calls feel the increment hardest, and dialer calls are short. Divide your average call length into each provider’s increment before you trust a rate table. One note for the negotiating table: quality routes across the industry are drifting toward 60/60, so get the increment in writing. The SIP trunk pricing guide walks through every line item, and the Twilio Elastic SIP Trunking pricing teardown runs the same math at a million calls.

ACTUAL CALL LENGTH BILLED ON 60/60 BILLED ON 12/6 BILLED ON 6/6
5 seconds 60 sec 12 sec 6 sec
20 seconds 60 sec 24 sec 24 sec
25 seconds 60 sec 30 sec 30 sec
45 seconds 60 sec 48 sec 48 sec
65 seconds 120 sec 66 sec 66 sec
3 minutes 180 sec 180 sec 180 sec

Increment math only — the per-minute rate is held constant. At 25 seconds, a typical connected dialer call, 60/60 doubles the cost of the identical call. By three minutes the clocks converge, which is why increments barely matter for a PBX full of long conversations.

Beyond the spec sheet

Where a SIP trunk fits in your stack.

A trunk is the voice artery of whatever communication platform you run. Register your PBX system, predictive dialer, or contact center against SIPNEX, and one trunk carries inbound and outbound traffic. DIDs ring straight into your queues. Your agents dial out on caller IDs you control, signed with A-level attestation.

This is elastic SIP trunking in the literal sense: capacity follows your call volumes instead of a channel count you ordered last quarter. Campaigns ramp, seasonal spikes hit, and the trunk flexes in real time. No change orders. No waiting on a carrier to “add lines.” The only limits are your own server resources and bandwidth.

Whatever sits above the trunk — a PBX, a dialer, a unified communications suite — every call to a real phone number leaves over SIP. That PSTN leg, not the app, determines whether your calls connect and how they are labeled.

Call quality is where SIP trunking services actually separate. Codec choice matters — we recommend G.711 for the cleanest audio — but so does who answers when something degrades. Our support team is the same group of engineers who run high-volume dialing operations. So when you compare providers, compare who picks up the phone, not just the spec sheet.

Turn-up

Connecting a cloud-based PBX, an Asterisk box, or a VICIdial cluster.

A SIP trunk authenticates in one of two ways. With IP authentication, the carrier allow-lists your server’s static public IP, and traffic from that address is trusted with no username or password. With registration (digest authentication), your PBX registers to the carrier’s proxy with credentials — the fit for dynamic IPs. Fixed-location servers with a static IP should use IP authentication; it is the simpler and more secure of the two.

Network prep is short. Open SIP signaling — UDP 5060, or 5061 for TLS — and the RTP media range your PBX uses to the carrier’s proxy IPs only. Asterisk’s sample rtp.conf ships 10000–20000. Turn off SIP ALG on the firewall or router. One-way audio after turn-up is almost always a NAT issue, not a trunk fault.

Codec choice sets audio quality and bandwidth. G.711 is uncompressed, high-quality audio at 64 kbps — about 85 kbps per call in each direction once packet headers are counted. G.729 compresses to 8 kbps, roughly 32 kbps per call, and trades clarity for it. Operators who build custom dial plans — routing call traffic by destination or by trunk — set codec preference per trunk. For dialer traffic we recommend G.711u first with G.729 as fallback; the G.711 vs G.729 comparison has the full numbers.

The trunk requirements are the same whether the phone system is a cloud-based PBX, an on-prem Asterisk box, or a VICIdial cluster. Platform-by-platform settings live in the SIP trunk configuration guide and on our SIP trunking for PBX page. If you would rather not run a phone system at all, SIPNEX hosted PBX rides the same carrier network; the trade-offs are laid out in hosted PBX vs SIP trunking.

Capacity planning

Channels, bandwidth, and calls per second.

On SIP, a trunk is a logical connection and a channel is one simultaneous call on it. One trunk carries as many channels as your bandwidth and server allow. You do not buy a second trunk for more capacity, the way a PRI needed another T1 for 23 more channels. Providers that sell channels in blocks are pricing a scarcity the technology does not have — the math is in our SIP trunk sizing guide.

Bandwidth math is simple. At roughly 85 kbps per G.711 call in each direction, 200 concurrent calls need about 17 Mbps each way. A dedicated 100 Mbps connection carries 500 or more. Size for peak concurrency plus headroom, and remember the link has to carry that load upstream as well as down.

Calls per second (CPS) is a separate limit from channels. A predictive dialer front-loads attempts at the top of each pacing cycle, so a low CPS ceiling smears the burst and distorts pacing. Telnyx, for example, documents a default of 20 CPS per source IP or SIP username, with excess attempts rejected with SIP 503. Ask any provider for its CPS ceiling in writing before load day.

Short-duration traffic is the other thing SIP trunk providers watch. Retail and CPaaS voice plans are priced and policed for conversational calls. Thousands of 20-second dialer attempts stand out in their metrics and can trigger review, surcharges, or a suspended account. Put that traffic on a carrier that expects it. SIPNEX bills dialer traffic at 12/6 because short calls are the workload, not a deviation from it. See which SIP providers allow autodialers and what to do when a dialer account is suspended.

Inbound side

Numbers, E911, and porting on a SIP trunk.

A SIP trunk provider is also your number source. Local DIDs, toll-free numbers, and the caller IDs your campaigns present all come from the carrier. Inbound calls to them ring into your PBX or queues over the same trunk. SIPNEX provisions local DID numbers directly and toll-free numbers as a registered RespOrg.

Emergency calling rides the trunk too. VoIP 911 routes on the Registered Location your carrier holds for each site, not on where the device sits, so every physical site gets its own address at turn-up. SIPNEX provides E911 service in the 50 US states. If you run a PBX, two rules sit on your side. Kari’s Law requires direct 911 dialing with no prefix, plus on-site notification. The RAY BAUM’S Act adds the dispatchable-location rule. 911 for VoIP explains both.

Switching providers does not mean new numbers. Numbers port while the old trunk stays live, and you can dial on new SIPNEX DIDs while the port completes. The FCC’s one-business-day rule (Order 09-41) covers only narrowly defined single-line simple ports. Multi-number business ports typically run 7 to 14 business days, and toll-free numbers move by RespOrg transfer. The number porting guide shows where the days go.

Know what you’re buying

Carrier, reseller, or aggregator?

Licensed carrier (own network)

Runs its own network and holds its own FCC authorization and STIR/SHAKEN certificate. Your trunk, your numbers, and your attestation come from one company. SIPNEX operates at this layer — the same network behind our wholesale VoIP carrier service.

Reseller

Buys trunk capacity from a carrier and rebrands it. Pricing carries a markup, channel caps are common, and support tickets travel through a company that cannot touch the network. Attestation is whatever the upstream grants — typically B-level.

BYOC aggregator

A software platform that lets you bring your own carrier trunk into its cloud — the BYOC model. Useful when your team is committed to a platform’s tooling. But the platform still needs a carrier underneath, and its fees stack on the carrier’s rate.

Aggregators also advertise global reach — every country on one rate card — by stitching together other carriers’ routes. SIPNEX runs the opposite model: a US-focused network kept close to the switch that terminates the call. International destinations are quoted on request rather than from a brokered A-Z deck. One consequence buyers miss: calls entering the US through international gateways typically arrive at C-level attestation, whoever originated them.

The well-known SIP trunk providers sort into the same layers. Twilio and Telnyx are CPaaS platforms: self-serve, API-first, billed 60/60. Bandwidth owns a nationwide network and prices through enterprise sales. SIPNEX is carrier-direct with published per-minute rates. Our ranked comparison of SIP trunk providers, the Bandwidth vs Telnyx vs SIPNEX breakdown, and the seven questions for any SIP trunk provider go deeper.

Frequently asked

About SIP trunking with SIPNEX.

What is a SIP trunk?
A SIP trunk is a virtual phone line that connects your phone system (PBX, dialer, contact center) to the public telephone network over the internet using the SIP protocol. Instead of physical copper lines, your calls travel as data packets. The trunk is the connection between your system and the carrier — in this case, SIPNEX.
How many concurrent calls can a SIP trunk handle?
On SIPNEX, there is no artificial concurrent channel limit at the trunk level. Your capacity is determined by your own server resources and internet bandwidth, not by a cap we impose. Most resellers limit you to 10–50 concurrent channels per trunk — we don’t.
What codecs do you support?
G.711 (ulaw/alaw) for highest quality, G.729 for bandwidth-constrained environments, and Opus where supported. For VICIdial deployments, we recommend G.711u as the primary codec with G.729 as a fallback.
Do I need special equipment to use SIP trunks?
You need an IP-capable PBX, dialer, or softswitch with a reliable internet connection. VICIdial, FreePBX, Asterisk, FreeSwitch, 3CX, and most modern platforms support SIP trunking natively. We provide configuration templates for VICIdial and will assist with other platforms.
How fast can I get a SIP trunk from SIPNEX?
24 hours from signed agreement to live trunk, typically. We provision your initial DIDs and trunk credentials within one business day. Number porting from another carrier takes longer (7–14 business days is standard) but you can start dialing on new DIDs immediately while the port completes.
What is STIR/SHAKEN and why does it matter for my SIP trunk?
STIR/SHAKEN is a caller ID authentication framework mandated by the FCC. Your outbound calls are cryptographically signed by the originating carrier with an attestation level (A, B, or C). A-level means verified caller — best answer rates. SIPNEX signs with our own SP-KI certificate, giving you direct A-level attestation. Most resellers can only provide B-level because they don’t hold their own certificate.
Can I switch my SIP trunks to SIPNEX without changing phone numbers?
Yes — your numbers come with you. Local, toll-free, and vanity numbers port across North America while your trunk stays live, and you can start dialing on new SIPNEX-provisioned DIDs immediately while the port completes in the background. Porting logistics and timelines are covered on our DID numbers page.
What kind of support does SIPNEX offer?
Phone and email support from engineers who run VICIdial and high-volume dialing operations themselves. When you call us about a sip.conf issue, the person answering knows what that means without escalation. We don’t have a generic tier-1 helpdesk — your first contact is the person who can fix it.
How can you tell if a SIP trunk provider is a carrier or a reseller?
Ask one question: does the provider sign calls with its own STIR/SHAKEN SP certificate, or does it inherit attestation from an upstream carrier? Resellers inherit. SIPNEX signs with its own certificate, holds direct FCC authorization, and files its own regulatory paperwork — the checks are laid out on our FCC licensing page.
Do I have to move all my traffic to SIPNEX at once?
No. Running trunks from two providers side by side is standard practice — most PBXs and dialers can register multiple trunks and route between them. Operators typically point a slice of outbound traffic at the new trunk, compare answer rates and quality against the incumbent, then shift volume as the numbers come in. Your existing numbers port in the background while you dial on new SIPNEX DIDs.
Do I need a SIP trunk if I already have VoIP phone service?
They solve different problems. A VoIP phone service sells you the whole system — handsets, apps, voicemail, and the carrier connection bundled together. A SIP trunk is only the carrier connection, for operators who run their own PBX, dialer, or contact center platform. If you own the phone system and just need it to reach the telephone network, a SIP trunk is the piece you are shopping for.
Do SIP trunk providers charge per channel or per minute?
Both models exist. Per-channel plans sell a fixed number of concurrent calls for a monthly fee — a holdover from PRI pricing that caps your dialer at the count you bought. Per-minute plans bill connected talk time and let concurrency float. SIPNEX bills per minute with unlimited channels and no per-channel fee; the rates are on our pricing page. Whichever model you compare, check the billing increment: 60/60 versus 12/6 moves the effective rate on short calls more than the headline rate does.
Should my SIP trunk use IP authentication or registration?
Use IP authentication when your PBX or dialer has a static public IP. The carrier allow-lists the address, no credentials cross the network, and there is no registration to keep alive. Use registration (digest authentication) when the IP is dynamic or the system sits behind a NAT you do not control; the PBX registers to the carrier’s proxy with a username and password. SIPNEX supports both, and our VICIdial setup guide shows the sip.conf for each.
Does a SIP trunk from SIPNEX include E911?
Yes. SIPNEX provides E911 in the 50 US states. Each physical site where your endpoints sit is provisioned with a Registered Location before the trunk goes live, and a 911 call routes to the public safety answering point for that address with your callback number. VoIP 911 routes on the address on file, not on device location, so update the record before agents move or a new site opens. The MLTS operator duties under Kari’s Law and RAY BAUM’S Act are covered on our E911 page.
How many concurrent calls can a 100 Mbps connection carry on a SIP trunk?
Roughly 500 or more G.711 calls on a dedicated 100 Mbps connection. Each G.711 call uses about 85 kbps in each direction — 64 kbps of audio plus IP, UDP, and RTP headers — so 200 concurrent calls need about 17 Mbps each way. G.729 cuts that to about 32 kbps per call at a cost in clarity. Size for your peak concurrent calls plus headroom for signaling and everything else on the link, and check the upstream side: voice needs the same bandwidth in both directions.

Related Guides

More Services

SIP Trunking for PBXDID NumbersToll-FreeSMS MessagingCall TrackingHosted PBX

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