How does SIP trunking work?

One end of the trunk is your PBX. The other is the carrier's voice network. Atlantech Online, for example, delivers SIP trunks to customers that already run their own VoIP PBX, and offers a hosted PBX to customers who would sooner not run one. With a trunk, the PBX keeps its extensions, voicemail and call routing. The carrier supplies the phone numbers and the route to everyone outside your phone system.

The trunk between them is an arrangement more than a cable. Each side is configured to reach the other and with rules for which messages to trust. No wire is set aside for each call. A trunk's capacity is counted in simultaneous calls, and our answer on what a SIP trunk channel is explains that unit.

Suppose someone at extension 214 dials a supplier's mobile number. The PBX sees that the number belongs to no extension of its own and sends the carrier a SIP message called an INVITE. The SIP standard, RFC 3261, walks through a tutorial call with this pattern.

The INVITE says who is calling and which number they want. It also carries a session description, attached the way a document is attached to an email. That description lists the audio formats the PBX can use and the address where the PBX wants to receive audio.

The carrier replies with 100 Trying, which means the request arrived and is being routed. It works out which network serves the supplier's number and passes the call toward it. This happens during setup. The mobile network is involved before the call is answered.

Next a 180 Ringing message comes back. It reports that the far end is alerting the supplier, and the caller at extension 214 hears ringback. When the supplier answers, a 200 OK arrives. It carries the other side's session description, with the audio format chosen and the address where audio should be sent. The PBX confirms with an ACK. RFC 3261 calls this exchange of two descriptions "a simple offer/answer model." Once it completes, both sides know what kind of audio to send and where to send it.

Other outcomes replace parts of this sequence. A busy or declined call returns an error where the 200 OK would have been, and some carriers ask the PBX to prove its identity before they accept the INVITE.

SIP creates the session, can change it mid-call and ends it. It does not carry the voices. Those travel in the Real-time Transport Protocol, or RTP, defined in RFC 3550. Each side chops its audio into small packets and sends them to the address the other side gave during setup.

RTP assumes an imperfect network. The RFC points out that packet networks sometimes lose packets, reorder them and delay them by varying amounts. So each RTP packet carries a sequence number and a timestamp. The receiver uses them to put the audio back in order and play it at the right pace. That is as far as the protocol goes. RTP "does not guarantee delivery or prevent out-of-order delivery," so it cannot make up for a connection that drops too much. Call quality depends on that connection, and also on the equipment at each end and how it is configured. A companion protocol, RTCP, reports how well the audio is arriving. We cover the network side in the connection and bandwidth SIP trunking needs.

The audio can also take its own route. In RFC 3261's tutorial call, "the end-to-end media packets take a different path from the SIP signaling messages." This helps explain a fault that puzzles people the first time they meet it. A call rings, connects and shows as active, yet one person hears nothing. The SIP messages got through, so one possible fault is in how the audio was addressed or delivered. The packets may have been sent to an address they could not reach, or a firewall may have blocked them on the way. Settings on the equipment at either end can matter too. Signaling and audio are checked separately for that reason.

When either person hangs up, that side sends a BYE and the other answers 200 OK. Both stop sending audio.

When someone dials one of your numbers, their network delivers the call to your carrier, because your carrier holds that number. The carrier then sends the INVITE to your PBX. The PBX rings an extension or plays the auto attendant, and the same alerting, answer and audio steps follow.

For this to work, the carrier has to know where your PBX is. One way is a fixed address agreed when the trunk is set up. Another is registration. RFC 3261 describes a device sending REGISTER messages when it starts and at regular intervals afterward, which tells a server where the device can currently be reached. The RFC presents registration as one of the ways a server can learn this. Which method your trunk uses is part of the carrier's setup.

A trunk may have a session border controller, or SBC, somewhere between the PBX and the carrier's network. RFC 5853 describes the functions these devices can perform. An SBC can repair SIP messages from equipment that does not follow the standard closely. It can keep a connection alive to a PBX that sits behind a firewall's address translation. It can also rewrite the session description so the audio passes through a relay the operator controls. Which of these a given SBC performs depends on how it is deployed.

An SBC is not part of every trunk on the customer's side. A carrier may run its own, and a PBX on a private circuit to the carrier may connect without one. Our answer on the equipment SIP trunking needs covers when you would own one.

Nothing in this exchange reserves a circuit. RFC 3550 says RTP "does not address resource reservation," and SIP manages sessions without setting aside network capacity for them. So the carrier's limit on simultaneous calls is a setting. It is not a count of physical lines.

That gives a trunk room to flex. Atlantech's 2023 post mentions a local government customer that adds capacity on its SIP trunks during weather events such as snowstorms and tropical storms. The carrier's limit is one of several ceilings. The PBX, any SBC or gateway and the data connection each have a maximum of their own, and extra calls need room under all of them. Our answer on how many simultaneous calls you need covers how to choose the number.

The trunk only defines the link between your PBX and your carrier. Beyond the carrier, the call may finish on a mobile network, a copper landline or another IP phone system, and our answer on the difference between PSTN and VoIP compares those networks. In the call above, the request had reached the supplier's mobile network before the supplier's phone was alerted.