How does the PSTN work?

No single company runs the PSTN. It is the combined networks of many telephone carriers, connected so that a number on one can reach a number on any other. We define it more fully in what the PSTN is.

A call starts with a number, and each switch along the way reads that number to choose the next hop. ITU-T Q.700 is the ITU's overview of the signaling system that telephone switches use with one another. It says the called party number carried in the first setup message is what routes the call to its destination. Those numbers follow E.164, the international numbering plan.

The classic layout has two tiers of switch. Atlantech Online, a business phone and fiber provider, described them in a 2016 post about its own voice switch. A Class 5 switch sits at the local telephone company's central office and directly serves subscribers. Class 4 switches, also called tandems, connect those offices to each other. A call to someone on a different Class 5 switch may pass through one or more Class 4 switches on the way.

The switch Atlantech described in 2016 was already software. It was built on Broadsoft, a soft switch platform that routes calls with programs running on servers. So the tiers describe roles in the network, and the equipment that fills those roles has been changing for years.

Switches talk to each other on a channel that is separate from the one carrying voices. Q.700 calls this common channel signaling: one channel carries labelled messages about many voice circuits at once. The system is Signaling System No. 7, or SS7. Its messages travel over signaling links between switches, sometimes through relay nodes called signal transfer points.

The part of SS7 that controls telephone calls is the ISDN User Part, or ISUP. RFC 3398 is the IETF standard for translating between ISUP and SIP, the protocol used in IP telephony, and it lays out the messages in an ordinary call. Five of them carry a call from setup to hang-up.

The voice needs a path, and on the traditional network that path is a circuit reserved while the call is being set up. ITU-T G.711, the telephone network's basic voice coding standard, samples speech 8,000 times a second and uses eight binary digits for each sample. That works out to 64,000 bits per second for each direction of a call.

Carriers bundle those channels onto trunks, and a business PRI line on a North American T-1 shows the pattern at small scale. Atlantech's comparison of SIP trunking and PRI describes that circuit as 23 voice channels, each supporting one call at a time, plus one data channel for call-related functions such as caller ID. Each voice channel is held by one call until it ends, and then it is free for the next one. Capacity on a circuit like this comes in fixed blocks, and Atlantech notes that PRI "can only be purchased in 23 line units." We compare the two handoffs in SIP trunking versus PRI.

The IP version carries the same conversation as packets. The speech is still digitized, but it crosses a data network in small pieces, and SIP messages do the job that ISUP does on the circuit side. No fixed time slot on a trunk is held for the length of the call, though a carrier can still limit how many calls a customer runs at once. We cover that contrast in the difference between PSTN and VoIP.

Today's PSTN is a mix of both methods. In a 2025 notice on IP interconnection, the FCC describes IP voice networks that carry calls from end to end over managed IP cores and session border controllers. When one of those calls has to cross a TDM network, it is "handed off via a media gateway to TDM at the network edge." TDM, short for time-division multiplexing, is the circuit technology described above.

The same notice names legacy telephone lines, business T-1 and PRI circuits, alarm and elevator lines and 911 services as examples of what may still be provisioned on TDM. We look at how much of the old network remains in whether the PSTN is still used.

A gateway does two translations at the boundary, and RFC 3398 describes the pair. A media gateway controller, also called a softswitch, maps the signaling. A SIP INVITE becomes an ISUP IAM on the way out, and the answer message coming back becomes a SIP 200 response. A media gateway handles the audio, with T1 or E1 trunk interfaces on the telephone side and IP interfaces on the other. The customer-premises version of this device is the subject of what a PSTN gateway does.

The signaling can cross over too. RFC 3398 notes that ISUP typically runs over SS7's own transport layer "although it can also run over IP." So a call may follow the ISUP sequence in the table while its messages travel on an IP network.

The move to IP also changes where carriers connect to each other. In the TDM world, the FCC notes, carriers tend to interconnect at many local central offices and tandems. IP networks can span larger regions and exchange traffic at fewer points, such as carrier hotels and internet exchanges.

A business reaches the PSTN through a carrier, and the carrier can hand over the connection in several forms. Atlantech's business phone service offers four: its own cloud PBX, or delivery to a PBX you already have over SIP, PRI or analog lines. An analog line and a PRI are circuit handoffs. A SIP trunk is an IP handoff, and we explain it in how SIP trunking works. A cloud PBX moves the phone system itself into the carrier's network.

Microsoft Teams adds one more layer and follows the same logic. Microsoft's PSTN connectivity options include Operator Connect, in which a certified carrier supplies the phone numbers and the PSTN access. Atlantech describes its Operator Connect path as "PSTN to carrier to Microsoft Operator Connect to Microsoft Teams," with the carrier managing the session border controllers. Under Direct Routing, the customer supplies its own certified controller, and Microsoft's planning guide lists "one or more PSTN trunks connected to the SBC" among the requirements.

Suppose an employee whose company uses Teams, with Atlantech as its Operator Connect carrier, dials the landline of a small insurance office. Suppose also that the office still has an analog line from its local telephone company.

The call runs Atlantech's path in reverse. Teams passes it to Microsoft's platform, which hands it to Atlantech as IP traffic. Atlantech, as the carrier, routes it toward the network that serves the dialed number. Suppose one carrier along the way connects over TDM. A gateway there turns the SIP INVITE into an IAM and takes a 64 kbit/s channel on a trunk. The far switch returns an ACM once it has the full number, and the line rings. When someone picks up, the ANM comes back and the gateway answers the SIP side. Both people can now talk.

The caller's voice leaves the laptop as packets and arrives at the desk phone as an analog signal. When either person hangs up, the release messages clear the trunk channel and the gateway ends the SIP session. All of the conversions happen inside the carriers' networks, so both people hear an ordinary phone call.