What is SIP trunking?

A business phone system, or PBX, does two kinds of work. Inside the company it connects people to each other, through extensions, voicemail, ring groups and the menu that answers the main number. It also needs a link to the public telephone network, because that is how a call reaches a customer's mobile or arrives from a supplier's landline. SIP trunking is that outside link, joining the two sides we describe in the difference between PSTN and PBX.

Twilio describes its own trunking product in terms of the same boundary. Its API reference says the service "provides connectivity for IP-based communications infrastructure to connect to the PSTN." The communications infrastructure is the customer's. The connectivity is what the customer buys.

That boundary also explains how a carrier packages its phone service. Atlantech Online's business phone page offers two routes. One is its cloud PBX, where Atlantech runs the phone system. The other is a connection to a system you run: "if you have your own PBX, we can deliver service via SIP, PRI, or analog." Its April 2023 post on SIP trunks says who each connection is for. SIP trunks go to customers that "already have their own VoIP PBX," and PRI and analog lines go to customers with a legacy PBX. So a SIP trunk is the thing to order when you have a phone system that speaks SIP and you want a carrier to connect it.

The system at your end does not have to be a cabinet in a back room. Twilio's trunking requirements ask for a SIP-enabled network element with access to the internet. Its examples are an IP PBX, a SIP call server, a session border controller and an access device that converts SIP to PRI. Any of them can be the equipment the trunk connects to. A PBX that predates SIP can sometimes be connected through a gateway, which is the subject of using SIP trunking with an existing phone system.

RFC 3261, the standard that defines SIP, calls it "an application-layer control protocol that can establish, modify, and terminate multimedia sessions," and gives internet telephone calls as an example. On a phone call, SIP messages request the call, report that the far end is ringing, confirm the answer and end the call when someone hangs up. The same document marks a limit: "SIP does not provide services." It supplies building blocks, and a carrier builds a service from them. So SIP is a protocol, and SIP trunking is a service that uses it.

The voices do not travel inside SIP messages. RFC 3261 names the Real-time Transport Protocol, or RTP, among the protocols typically used alongside SIP to carry real-time data. Each call on a trunk therefore has two parts, the signaling that manages it and the audio stream itself. This matters when something goes wrong, because a call can connect correctly and still have no sound in one direction. We follow one call through both parts in how SIP trunking works.

The word trunk is older than SIP. Atlantech's post describes a trunk as a link that carries signals between the nodes of a communications system. For a business, that has meant the link between the phone system and the carrier, as distinct from the line to one person's phone. On a SIP trunk the link is not a dedicated cable. It is an arrangement between two systems across an IP network, in which each side is configured to reach the other and to know which traffic to trust.

In conversation, "a SIP trunk" can mean the connection, the phone numbers and the call capacity all at once. Twilio and SIP.US each keep those pieces apart.

Twilio's API holds a different record for each of the first two. There is one for the trunk, which is the connection. There is another for the phone numbers associated with it, which are addresses that route calls to you.

SIP.US, which publishes its prices, adds capacity as a third item and sells it in channels. It says you need one channel "for every concurrent incoming or outgoing call." It lists a channel at $24.95 a month on its month-to-month plan and a local number in the lower 48 states at $1.00 a month plus a one-time $1.00 setup fee, before taxes and fees. It charges nothing for the trunk itself. Atlantech's phone page describes its pricing as tailored to the customer and prints no trunk rate. On a trunk from Atlantech, the number of simultaneous calls you need and the count of phone numbers are things to state when you ask for the quote.

Atlantech's two routes show the difference between a trunk and a hosted phone system. With its SIP delivery, you run the PBX and Atlantech connects it. With its cloud PBX, Atlantech runs the phone system, and its page describes that service as including unlimited nationwide calling. Other providers arrange public calling in other ways, so SIP trunking versus cloud PBX is a comparison of who runs the phone system more than of how calls leave it.

A PRI does the same job as a SIP trunk through a different connection. It is an interface on a digital telephone circuit, and that circuit plugs into a PRI port on the PBX, where a SIP trunk arrives over the data network. A PBX can be built for either one, and Atlantech delivers both. The practical differences between them, in SIP trunking versus PRI, come down to capacity and to what your PBX can accept.

VoIP is a category, and SIP trunking belongs to it. The term covers voice carried over data networks with internet protocols, and that includes a desk phone talking to a PBX across the office network with no carrier involved. A SIP trunk is one use of the technology, at the point where a phone system meets the carrier. That is why SIP trunking and VoIP are not alternatives to each other.

A trunk runs over an IP connection, so that connection has to carry the calls. Twilio's requirements ask for enough bandwidth to support peak call traffic. Its planning figure is 100 kbps for each simultaneous call on the G.711 codec, with overhead included. Suppose your busiest moment has 15 outside calls in progress. By Twilio's allowance, the connection needs 1.5 Mbps free for voice at that moment, on top of everything else the office is doing online. Other codecs give other figures, which we work through in the connection and bandwidth SIP trunking needs.

Spare bandwidth is one part of the requirement. RFC 3550, which defines RTP, says the protocol itself "does not guarantee delivery or prevent out-of-order delivery." It gives the receiving end sequence numbers and timing information to rebuild the audio, and it leaves delivery to the network underneath. Packets that are lost or delayed on the way can impair the audio. So the quality of the connection is one of the things that shapes how calls on a trunk sound, along with the equipment at each end and how it is set up.

The trunk also relies on the equipment at your end having power and a working connection. A carrier's network can be running normally while your phones are down, if the PBX has lost power or the office has lost its data connection. An office that has to keep its phones up plans for both, with backup power for the phone system and a second path to the carrier.