Glossary · SIP

What is SIP trunking?

SIP trunking is a method for delivering voice and messaging services over the internet using the Session Initiation Protocol (SIP). It replaces traditional analog phone lines with virtual “trunks” that connect a customer’s on-premises PBX or Session Border Controller (SBC) to a VoIP provider, which then connects to the Public Switched Telephone Network (PSTN). SIP trunking is common in enterprise deployments where the customer wants to keep an existing PBX but gain cloud flexibility and lower costs.

How SIP trunking works

Customer operates an IP-PBX or SBC on-premises or in a private cloud
SIP trunks run between that PBX/SBC and the SIP trunk provider
The SIP provider handles PSTN origination and termination
Calls traverse the trunks as SIP signaling + RTP media streams
Providers allocate DIDs (Direct Inward Dialing numbers) to the customer

The “trunk” terminology comes from analog telephony where physical trunks carried multiple lines between switches. SIP trunks carry multiple concurrent calls over IP.

SIP trunking vs. hosted VoIP / UCaaS

Dimension	SIP Trunking	Hosted VoIP / UCaaS
PBX location	Customer-operated (on-prem or private cloud)	Provider-operated
Setup complexity	Moderate to high (SBC config, routing)	Low (sign up, download app)
Control	Full control over PBX features and routing	Provider controls platform, customer controls config
Infrastructure cost	Customer pays for PBX/SBC	No infrastructure cost
Feature breadth	Depends on customer’s PBX	Full UCaaS stack from provider
Best for	Enterprises with existing PBX investments	Small to mid-market; simplicity buyers

DialPhone delivers hosted VoIP / UCaaS natively. For customers with existing SBC investments or complex routing requirements, DialPhone also supports SIP trunking to feed a customer-operated PBX or to enable Microsoft Teams Direct Routing.

Why use SIP trunking

Preserve existing PBX investment: don’t replace a working on-premises system
Complex routing requirements: survivable branch appliances, regional failover, specialized recording
Regulatory control: keep media paths within a specific jurisdiction
Hybrid deployment: gradually migrate from on-prem PBX to cloud
Microsoft Teams Direct Routing: use a SIP trunk through a customer-operated SBC
Custom integration: telephony integrated with in-house applications

SIP trunking pricing models

Per concurrent call (channel): pay for simultaneous call capacity
Per-user: flat per-user pricing regardless of call volume
Per-minute: metered pricing based on usage
Unlimited per channel: flat per-channel with unlimited usage

DialPhone SIP trunking pricing is available on request and varies by region, committed volume, and included DID count. The hosted VoIP model (per user) covers most use cases and is significantly simpler to operate. See DialPhone pricing →

Technical components

SIP: the signaling protocol that sets up, modifies, and terminates sessions
RTP / SRTP: the media protocol that carries the actual audio (Secure RTP for encryption)
SBC (Session Border Controller): security and interoperability gateway at the edge of the customer’s network
Codec: compression algorithm (G.711, G.722, Opus) that encodes voice into packets
DID: Direct Inward Dialing number, the phone number assigned to each trunk or user
DTMF: touch-tone digits sent via SIP INFO or RFC 2833

Example

A 2,000-employee financial services firm has a Cisco Unified Communications Manager (CUCM) PBX deployed across three data centers. They use SIP trunking from DialPhone to:

Replace their previous TDM (analog) trunks with SIP trunks, cutting $40,000 per year in PSTN line costs
Maintain their existing CUCM dial plan and call routing
Enable Microsoft Teams Direct Routing for a subset of users moving to Teams Phone
Keep sensitive financial calls routed through the on-premises CUCM for FINRA compliance

SIP trunking vs hosted PBX vs UCaaS: which fits your business

The decision is simpler than vendors make it sound. Ask one question first: do you already own a working PBX you want to keep? If yes, SIP trunking is the candidate. If no, stop reading the SIP trunking guides and look at hosted UCaaS.

SIP trunking is bring-your-own-PBX. You own (or already paid for) an IP-PBX like Cisco CUCM, Avaya IP Office, Asterisk, or 3CX. You buy SIP trunks from a provider to give that PBX a path to the PSTN. You pay per channel (concurrent call capacity) and you pay for minutes or unlimited bundles. The PBX, its features, its upgrades, and its security are your problem. Fits enterprises that already sank capex into on-premises gear, regulated firms that need on-prem media paths, and shops with custom call-center routing that hosted platforms won’t expose.

Hosted PBX / UCaaS is vendor-managed. The PBX runs in the provider’s cloud. You pay per seat, you get a softphone and mobile app on day one, and the provider handles updates, encryption, redundancy, and SBC complexity. Fits anyone who doesn’t already own a PBX — which in 2026 is nearly everyone outside large enterprise.

Cost comparison at 50 users over 3 years:

Model	Year 1	Year 2	Year 3	3-year total
SIP trunks (30 channels × $20/mo + $3K/yr PBX maintenance)	$10,200	$10,200	$10,200	$30,600
Hosted UCaaS ($30/seat × 50)	$18,000	$18,000	$18,000	$54,000
Hosted UCaaS ($24/seat × 50, budget tier)	$14,400	$14,400	$14,400	$43,200

SIP looks cheaper on paper. The table hides the PBX itself ($30K-$150K capex amortized), the SBC ($5K-$25K), an in-house telecom admin, toll fraud insurance, and the engineering time for every Teams/CRM integration you’d get free in a UCaaS bundle. Total cost of ownership flips above 50-100 seats for most companies.

Internal links: Hosted PBX, UCaaS, DialPhone business phone.

Top SIP trunking providers in 2026

DialPhone is hosted UCaaS, not a SIP trunking provider. If you want a managed cloud phone system without running a PBX, see DialPhone business phone. If you specifically need BYOPBX SIP trunks, the providers below are the current shortlist.

Twilio Elastic SIP Trunking — pay-as-you-go with no monthly minimums. Outbound US is roughly $0.004/min, inbound $0.0085/min, DIDs $1/month. Strong API control, programmable routing, used by engineering-led teams. Weak on hand-holding — you operate it yourself. Best for tech companies and developer-platforms.
Bandwidth — carrier-grade SIP with their own US tier-1 network. Custom enterprise pricing only; no public price list. Powers Microsoft Teams Phone, RingCentral, Zoom Phone at the carrier layer. Best for high-volume enterprise and CPaaS resellers needing direct network access.
Telnyx — developer-first like Twilio but typically 30-50% cheaper. About $1/channel + per-minute outbound from $0.0025/min. Private IP backbone, MFA built in, good documentation. Best for SaaS platforms embedding voice and mid-market with engineers on staff.
SIP.US — flat $24.95/channel/month with unlimited US/Canada calling. Predictable, no minute counting, free DIDs included. Best for small businesses with steady volume who hate metered bills. Less flexible internationally.
Vonage Business Communications SIP Trunking — bundled with Vonage’s broader UCaaS suite. Per-channel + per-minute models, enterprise SLAs, integrated with Vonage APIs. Best for mid-market already on Vonage who need a BYOPBX side car.
Plivo — developer platform similar to Twilio/Telnyx. Per-minute outbound from $0.0055/min, $0.80/month per DID. Strong global coverage in 195+ countries. Best for international outbound use cases and CPaaS-style apps.
Flowroute (Intrado/West) — enterprise SIP with HD voice, intelligent routing, and direct PSTN interconnects. Now part of Intrado after the West acquisition. Custom contracts, typically per-channel + minute pools. Best for large contact centers and regulated industries needing carrier diversity.

If you’re reading this and you don’t have a PBX, you don’t want any of these. You want hosted UCaaS. See DialPhone business phone.

SIP trunking pricing models in detail

Three pricing structures dominate the market. Pick the one that matches your call pattern.

Per-channel flat rate — typically $15-30 per channel per month with unlimited minutes (US/Canada) bundled. SIP.US, Nextiva SIP, and Vonage default here. Predictable budgeting. Best for inbound-heavy operations like support lines, appointment booking, and steady-state offices. Wastes money if your channels sit idle most of the day.
Per-minute pay-as-you-go — $0.004-0.012 per minute outbound US, $0.0075-0.015/min inbound. Twilio, Telnyx, Plivo default here. No channel commitment. Best for outbound-heavy campaigns, irregular volume, dialer apps, and SaaS embedding voice. Dangerous if you don’t monitor — a toll fraud incident can rack five figures in hours.
Hybrid bundled (channels + minute pool) — committed channels plus a monthly minute allowance with overage rates. Most enterprise SIP contracts use this. Example: 50 channels included + 500,000 minutes/month, overage at $0.008/min. Best for large enterprises with finance teams that want capex-like predictability with usage-based scaling.

Hidden costs that wreck the budget spreadsheet:

DID number rentals: $0.50-$2/month per number. A call center with 200 local presence numbers pays $100-400/month before a single call.
Toll-free inbound: $0.015-$0.025/min, separate from outbound minute pools. High-volume support lines feel this fast.
International outbound: 10x-50x the domestic rate. UK mobile is often $0.10-0.20/min; some African destinations exceed $1/min.
E911 fees: $0.50-$2/month per DID, mandatory in the US, regulatory in Canada (NG911).
Setup and porting fees: $25-$200 one-time per number to port from your old carrier. Multiply by however many numbers you’re moving.
CNAM (caller ID name) dip fees: $0.004-$0.0075 per outbound lookup at some providers.

Get a quote in writing with all line items before signing. “Unlimited” rarely means unlimited.

SIP trunking technical requirements

You need more than a credit card. SIP trunks demand specific infrastructure:

Compatible IP-PBX or SBC — Cisco CUCM, Avaya, Asterisk, FreePBX, 3CX, Mitel, or a standalone SBC like AudioCodes, Oracle (Acme Packet), or Ribbon. The PBX must speak SIP over TCP/UDP/TLS on port 5060 or 5061.
Bandwidth — budget ~100 Kbps per concurrent HD call (G.722 or Opus with overhead). Older G.711 needs ~87 Kbps. A 30-channel trunk at peak utilization needs 3 Mbps symmetric, plus margin. Asymmetric residential broadband is not enough for serious deployments.
QoS configuration on the edge router — mark SIP signaling (DSCP CS3) and RTP media (DSCP EF) and give them priority queueing. Without QoS, jitter and packet loss destroy call quality the moment someone streams Netflix.
Session Border Controller (SBC) — strongly recommended for security. Acts as a SIP-aware firewall, handles NAT traversal, enforces topology hiding, supports encryption (TLS/SRTP), and provides toll fraud monitoring. Generic firewalls don’t understand SIP and break it.
Static public IP or DNS SRV records — most providers require either a static IP for trunk registration, or DNS SRV records like _sip._udp.example.com for failover. Residential dynamic IPs won’t work.
TLS 1.2+ for signaling, SRTP for media — encrypt both. Plain UDP SIP on port 5060 invites scanning bots and fraud attempts within hours of going live.
E911 location registration — every DID needs a registered service address. Some jurisdictions (Kari’s Law, RAY BAUM’s Act in the US) require dispatchable location accuracy or you face fines.
Failover trunk to a secondary provider — never single-source. A second SIP trunk on a different carrier (e.g., Bandwidth primary, Twilio backup) protects against carrier outages, which do happen and last hours.

When to choose SIP trunking instead of hosted UCaaS

SIP wins in five specific scenarios. Outside these, hosted UCaaS is the better call.

You already own a working PBX with sunk capex you can’t redeploy. A 3-year-old Cisco CUCM cluster you paid $200K for isn’t free to throw away. SIP trunks let you keep it alive and cut PSTN line costs.
You need extreme call-center customizations that hosted platforms don’t expose. Custom IVR logic written in Asterisk dialplan, third-party WFM integrations at the PBX layer, or specialized recording for compliance (e.g., bespoke FINRA archival) — hosted UCaaS APIs may not reach deep enough.
Regulated industries with on-premises requirements. Rare in 2026 — most regulators now accept SOC 2 / ISO 27001 cloud — but some defense, government, and healthcare deployments still mandate on-prem media paths. SIP trunks let the PBX stay in your data center.
Very large enterprise, 1,000+ seats. Per-channel SIP economics beat per-seat hosted at scale. 1,000 seats hosted at $30/seat = $360K/year. 200 SIP channels (1:5 ratio) at $20/channel = $48K/year + your PBX. The math flips around 500-800 seats.
Multi-country deployments where local SIP trunks slash international toll. A SIP trunk in Frankfurt with German DIDs routes German calls as local; routing them through US hosted UCaaS pays international rates. Multinationals with high in-country call volume save real money with regional trunks.

Conclusion: SMBs almost always lose on SIP. Operational burden (you’re the on-call PBX admin), fraud risk (one compromise = $50K+ liability), and missing features (no built-in AI transcription, sentiment, coaching) wipe out any per-channel savings. If you’re under ~500 seats and don’t already own a PBX, get hosted UCaaS and move on.

SIP trunking frequently asked questions

What’s the difference between SIP trunking and VoIP?

VoIP (Voice over IP) is the broad category — any voice traffic carried over IP networks. SIP trunking is a specific delivery method within VoIP: it uses the Session Initiation Protocol to connect a customer-owned PBX to a provider’s network for PSTN access. All SIP trunking is VoIP, but not all VoIP is SIP trunking. Hosted UCaaS, WebRTC apps, mobile softphones, and consumer apps like FaceTime Audio are all VoIP without being SIP trunking. SIP trunking specifically targets the BYOPBX enterprise model.

How many SIP channels do I need for my business?

Channels equal concurrent calls, not users. The standard rule is 1 channel per 3-5 users for typical office work, 1 per 2 users for sales/support teams making constant calls, and 1 per 8-10 users for back-office staff who rarely dial. A 50-person company with mixed roles usually needs 15-25 channels. Erlang B calculators give a more precise answer if you know average call duration and busy-hour call volume. Always add 20% headroom for peaks.

Can I use SIP trunking without a PBX?

Technically yes, with a softphone or SIP-capable device registered directly to the trunk — but it’s a bad idea for anything beyond a single user. SIP trunks expect a PBX or SBC to handle routing, authentication, multiple endpoints, and call control. Without one, you get no extensions, no IVR, no hunt groups, no voicemail orchestration. If you don’t have a PBX and don’t want one, you don’t want SIP trunking — you want hosted UCaaS, which bundles the PBX, the trunks, and the apps into one service.

Is SIP trunking secure against toll fraud?

Not by default. Plain SIP on port 5060 is scanned by bots constantly, and a weak PBX password or open dial plan can rack $20,000+ in fraudulent international calls overnight. Mitigations: enforce TLS 1.2+ for signaling and SRTP for media, deploy an SBC with rate limiting and geo-fencing, restrict international dialing by default, require strong passwords plus IP allowlists, enable fraud alerts at the provider, and set hard monthly spend caps. Most major SIP providers (Bandwidth, Telnyx, Twilio) offer fraud-detection add-ons — turn them on.

How does SIP trunking handle E911 for emergency calls?

Every DID on the trunk must have a registered service address in the provider’s E911 database. When a user dials 911, the trunk routes the call to the appropriate Public Safety Answering Point (PSAP) based on that registered address. US regulations (Kari’s Law and RAY BAUM’s Act) additionally require dispatchable location detail (building, floor, room) for multi-line systems, and notification to a central point (e.g., front desk) when 911 is dialed. Your PBX or SBC must support this. Non-compliance carries five-figure fines per violation.

Can I keep my existing phone numbers when switching to SIP trunking?

Yes — porting (LNP, Local Number Portability) lets you move numbers between carriers. The process: confirm the number is portable (most US/Canada landline and mobile numbers are), get a Letter of Authorization (LOA) and recent bill from your current carrier, submit the port request to the new SIP provider, wait 7-21 business days for the FOC (Firm Order Commitment) date. Numbers cut over on the scheduled date with usually under 15 minutes of downtime. Toll-free numbers port faster (3-5 days). International numbers vary wildly by country.

See SIP trunking options

Most customers don’t need SIP trunking, hosted UCaaS is simpler and cheaper. SIP trunking fits specific enterprise scenarios. Talk to sales → for SIP trunking pricing and deployment details.

Hosted PBX vs. cloud phone: what’s the difference
VoIP security and encryption
VoIP — the protocol SIP trunking is built on
DID — the phone numbers delivered over SIP trunks
Hosted PBX — the simpler alternative for most businesses
DialPhone pricing