What Is a SIP Trunk

A SIP trunk is a data connection that carries phone calls into the public telephone network over the internet instead of over a physical cable - and unlike a single classic phone line, it can carry several calls at the same time. For an app, a phone system (PBX) or an AI voice agent, it is the only way to reach a real phone number at all: without a SIP trunk it can only talk over the internet, with one it can answer and place calls like an ordinary phone.

What is a SIP trunk?

"SIP" stands for Session Initiation Protocol - an internet signaling protocol (standard RFC 3261 from 2002) that two parties use to negotiate a call: who is calling whom, what audio format to use, and when the call starts and ends. The audio itself then travels separately, over a different protocol (RTP) - SIP only "schedules the meeting", it does not carry the voice itself.

The word "trunk" is older, from the days of physical phone systems - it described a bundle of physical lines (typically an ISDN PRI with thirty channels) connecting a company's phone system to the telephone network, able to carry many calls at once. A SIP trunk does the same thing without the physical cable: it is a data connection to a provider (a telephone carrier or a platform like volai) that carries any number of concurrent calls - how many you can run at once is a matter of contract and capacity, not the number of sockets in the wall.

How is it different from a single phone line?

A regular SIP account for a softphone (Zoiper, Linphone) logs into a server once (SIP REGISTER) and stays "online" from then on - similar to how a mobile phone registers with a carrier's network. One set of credentials usually only supports one registered device at a time. A SIP trunk, by contrast, is built for volume: it carries many parallel calls without registering separately for each one.

For a developer, that is in practice the difference between "one phone, one number" and "a gateway that dozens of calls to a support line pass through at once". For a non-technical business owner, it is enough to know that a SIP trunk is the invisible layer that lets an app or an AI act like a phone line in the first place - it is not a product you buy on its own, it is a way of connecting to the phone network.

Why does an AI voice agent need a SIP trunk?

A voice AI agent (ElevenLabs Conversational AI, Vapi, Retell and similar platforms) runs in the cloud and understands audio, not the telephone network. To reach a real caller on a real phone number, it needs a bridge between the internet voice interface where the agent lives and the public telephone network where the caller lives. That bridge is exactly what a SIP trunk provides - it carries signaling (who, with whom, when) and audio in both directions.

For inbound calls this works as one-way trust: the telephone network sends a SIP invite to the address the agent platform registered as its SIP trunk, and the platform accepts the call based on the allowed address, with no persistent registration. Outbound calls are usually harder - most agent platforms (ElevenLabs included) cannot permanently register a phone line as a SIP client, they can only send a single authenticated call. The telephone network will not route that call straight to the destination number, so it goes through a short-lived connection instead - at volai this is called a relay.

How volai handles it

Both directions are available on a single volai number:

The inbound direction is set with a single REST API request - the number's routing switches to SIP and gets a target address:

curl -X PATCH "https://volai.cz/v1/numbers/+420601234567" \
  -H "Authorization: Bearer vk_YOUR_KEY" \
  -H "Content-Type: application/json" \
  -d '{"routing": {"mode": "sip", "sipUri": "sip:number@your-server.com"}}'

Only the inbound minute is billed (0.50 CZK/min (~EUR 0.02), prices as of September 2, 2026), no extra agent charge. Outbound calls from your own softphone go through automatically once it is registered, and are billed from your credit at the same rate as an outbound API call (0.92 CZK/min (~EUR 0.04)). Automatic billing for calls dialed this way is still being finished, though, so a call may not show up on your credit right away - we bill it retroactively; the SIP page has the details. Outbound calls from your own AI platform, which cannot register, go through a short-lived relay connection - the full walkthrough is on the Bring your own agent page.

All prices are in Czech koruna (CZK) excluding VAT. Euro amounts are approximate and for orientation only - your card is always charged in CZK.

Both paths share the same limits, and they are easy to forget: a volai line supports one registered device at a time (not unlimited concurrent registrations), and volai does not offer full trunk peering yet - connecting your own PBX as a peer carrier with its own routing for inbound calls from outside; the details are in the Limits section on the SIP page. A volai SIP trunk also carries audio only, no video and no other SIP extensions - that is no loss for plain phone calls (video was never going over a phone call anyway), but it is worth knowing before anyone tries to push anything but audio down the same connection.

A few terms, defined

When a SIP trunk is worth it, and when a REST API is simpler

Your own SIP trunk makes sense when you want full control over signaling - your own PBX, your own voice platform, your own routing logic. The cost is extra technical work: codecs, NAT, firewalls, the one-device-per-line registration limit, and the fact that minutes run outside our API entirely and cannot be blocked at the exact moment of dialing (the limits are documented on the SIP page).

• Want "call this number" or "send an SMS" without touching SIP at all - the REST API or the MCP server does exactly that: one request, or one sentence to your editor, starts a call without you ever knowing SIP exists.
• Want a voice agent without running your own voice platform - the built-in volai agent works with no SIP setup at all; pricing is on the Pricing page.

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