The PSTN connects a call in two layers that run side by side. Signaling messages pass from switch to switch to find the dialed number and ring it. A voice path then opens between the two phones and stays up until someone hangs up. On the older parts of the network, that path is a reserved 64 kbit/s circuit. On the newer parts it is a stream of IP packets, and gateways convert between the two where a call passes from one kind of network to the other.
\n","summary":"The PSTN connects a call in two layers. Signaling messages pass between switches to find the dialed number and ring it, and a voice path stays open until someone hangs up. On older parts of the network that path is a reserved 64 kbit/s circuit. On newer parts it is IP packets, and gateways convert between the two where a call passes from one kind of network to the other.","publishedAt":"2026-07-18T15:31:57.455","verifiedAt":"2026-07-18T00:00:00","editorialStatus":"APPROVED","lastReviewedAt":"2026-10-10T03:02:03.474","nextReviewDueAt":"2026-10-18T00:00:00","templateVersion":"v2","aliases":["How does a phone call travel through the PSTN?","How is a call routed on the public switched telephone network?","How does circuit switching work in the telephone network?","What happens when you make a landline call?","How does the PSTN set up and connect a call?","How do telephone exchanges route calls?","How does SS7 set up a phone call?","How does the PSTN carry voice?","PSTN call flow explained","Steps in a PSTN phone call","How does the traditional telephone network connect calls?","How does the landline network work?"],"confidenceScore":92,"confidenceLabel":"High","canonicalUrl":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"contributorOrganizationProfile":null,"contributorPerson":null,"sections":[{"id":"23a7c81b-98b6-4b2f-b4a4-5769cc784318","sectionKey":"number_steers_the_call","sectionType":"markdown_section","heading":"The number steers the call","introMarkdown":"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](https://www.answerstack.io/q/what-is-the-pstn-public-switched-telephone-network).\n\nA call starts with a number, and each switch along the way reads that number to choose the next hop. [ITU-T Q.700](https://www.itu.int/rec/T-REC-Q.700-199303-I/en) 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.\n\nThe 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](https://www.atlantech.net/blog/atlantech-onlines-voice-switch-what-you-should-know). 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.\n\nThe 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.","introHtml":"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.
\nA 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.
\nThe 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.
\nThe 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.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":10},{"id":"e6a35a54-9db6-45bd-be35-8a71018f6d14","sectionKey":"signaling_sets_up_the_call","sectionType":"markdown_section","heading":"Signaling sets up the call before anyone speaks","introMarkdown":"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.\n\nThe part of SS7 that controls telephone calls is the ISDN User Part, or ISUP. [RFC 3398](https://www.rfc-editor.org/rfc/rfc3398.html) 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.","introHtml":"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.
\nThe 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.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":20},{"id":"259aaf85-f501-4138-bab8-bb991fa16e58","sectionKey":"isup_call_messages","sectionType":"table_section","heading":"The five messages in an ordinary call","introMarkdown":null,"introHtml":null,"outroMarkdown":null,"outroHtml":null,"contentJson":{"rows":[{"cells":["Initial Address Message (IAM)","The caller's side starts the call","Carries the called number to the next switch and begins setup"]},{"cells":["Address Complete Message (ACM)","The far switch has enough of the number","Confirms the call can be set up. Ringing typically follows"]},{"cells":["Answer Message (ANM)","The called party picks up","Marks the call as answered, with audio established both ways if it wasn't already"]},{"cells":["Release (REL)","Either side hangs up","Asks to clear the call"]},{"cells":["Release Complete (RLC)","In reply to the release","Confirms the circuit is ready for another call"]}],"columns":["Message","When it is sent","What it does"]},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":30},{"id":"7706ffc7-0c47-42e5-a750-1d6cbbd76171","sectionKey":"voice_path","sectionType":"markdown_section","heading":"The voice path: a reserved circuit or a stream of packets","introMarkdown":"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](https://www.itu.int/rec/T-REC-G.711-198811-I/en), 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.\n\nCarriers 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](https://www.atlantech.net/blog/sip-trunking-vs-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](https://www.answerstack.io/q/what-is-the-difference-between-sip-trunking-and-pri).\n\nThe 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](https://www.answerstack.io/q/what-is-the-difference-between-pstn-and-voip).","introHtml":"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.
\nCarriers 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.
\nThe 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.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":40},{"id":"f4ffabcc-e7d6-4a88-975f-f4a9d56a8a50","sectionKey":"where_circuits_and_packets_meet","sectionType":"markdown_section","heading":"Where circuits and packets meet","introMarkdown":"Today's PSTN is a mix of both methods. In a [2025 notice on IP interconnection](https://www.federalregister.gov/documents/2025/11/26/2025-21324/advancing-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.\n\nThe 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.\n\nA 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](https://www.answerstack.io/q/what-is-a-pstn-gateway-and-what-do-fxs-and-fxo-ports-do).\n\nThe 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.\n\nThe 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.","introHtml":"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.
\nThe 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.
\nA 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.
\nThe 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.
\nThe 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.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":50},{"id":"6dcce511-85de-4edd-86f0-0530e01d2258","sectionKey":"how_a_business_connects","sectionType":"markdown_section","heading":"How a business phone system connects","introMarkdown":"A business reaches the PSTN through a carrier, and the carrier can hand over the connection in several forms. Atlantech's [business phone service](https://www.atlantech.net/solutions/voice/phone) 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](https://www.answerstack.io/q/how-does-sip-trunking-work). A cloud PBX moves the phone system itself into the carrier's network.\n\nMicrosoft Teams adds one more layer and follows the same logic. Microsoft's [PSTN connectivity options](https://learn.microsoft.com/en-us/microsoftteams/pstn-connectivity) include Operator Connect, in which a certified carrier supplies the phone numbers and the PSTN access. Atlantech [describes its Operator Connect path](https://www.atlantech.net/faq/microsoft-teams-operator-connect/how-does-operator-connect-actually-connect-my-phone-numbers-to-teams) 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](https://learn.microsoft.com/en-us/microsoftteams/direct-routing-plan) lists \"one or more PSTN trunks connected to the SBC\" among the requirements.","introHtml":"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.
\nMicrosoft 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.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":60},{"id":"13809628-b2f9-4746-bd34-10d02c50d6de","sectionKey":"one_call_across_both","sectionType":"markdown_section","heading":"One call across both kinds of network","introMarkdown":"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.\n\nThe 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.\n\nThe 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.","introHtml":"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.
\nThe 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.
\nThe 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.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":70}],"citations":[{"title":"Public switched telephone network","url":"https://en.wikipedia.org/wiki/Public_switched_telephone_network","excerpt":"The public switched telephone network (PSTN) is the aggregate of the world's telephone networks that are operated by national, regional, or local telephony operators.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"PSTN definition; circuit switching with ISUP call setup; switching hierarchy of central offices, tandems, and gateways connected by trunks; digital core with G.711 PCM, DS0 64 kbit/s, DS1/T1/E1, SONET/SDH; SS7 controlling calls between exchanges","domain":"en.wikipedia.org","publisherName":"Wikipedia"},{"title":"PSTN (public switched telephone network)","url":"https://www.techtarget.com/searchnetworking/definition/PSTN","excerpt":"PSTN (public switched telephone network) is the world's collection of interconnected voice-oriented public telephone networks via traditional circuit-switched networks.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"Interconnected circuit-switched networks; subscriber lines connect to a local exchange that communicates with trunk, main, and central exchanges; switches act as routing nodes; carriers transitioning to all-IP; contrast with packet-switched VoIP","domain":"techtarget.com","publisherName":"Informa TechTarget"},{"title":"Circuit switching","url":"https://en.wikipedia.org/wiki/Circuit_switching","excerpt":"The circuit guarantees the full bandwidth of the channel and remains connected for the duration of the communication session.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"Circuit switching establishes a dedicated channel before communication and reserves it for the duration of the call even during silence; the PSTN is the classic example; it uses capacity less efficiently than shared packet switching","domain":"en.wikipedia.org","publisherName":"Wikipedia"},{"title":"Telephone exchange","url":"https://en.wikipedia.org/wiki/Telephone_exchange","excerpt":"The exchange provides dial tone at that time to indicate to the user that the exchange is ready to receive dialed digits.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"Exchange senses the off-hook condition and provides dial tone; it processes pulse or DTMF digits and establishes a connection; it maintains and supervises the connection until a party hangs up; central office / Class 5 switch terminates the local loop","domain":"en.wikipedia.org","publisherName":"Wikipedia"},{"title":"Dual-tone multi-frequency signaling","url":"https://en.wikipedia.org/wiki/Dual-tone_multi-frequency_signaling","excerpt":"Dual-tone multi-frequency signaling is a telecommunication signaling system using the voice-frequency band over telephone lines between telephone equipment and other communications devices.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"DTMF sends dialed digits as paired voice-band tones from the telephone to the switching center, which decodes them to route the call; standardized as ITU-T Recommendation Q.23; it replaced rotary pulse dialing","domain":"en.wikipedia.org","publisherName":"Wikipedia"},{"title":"ISDN User Part","url":"https://en.wikipedia.org/wiki/ISDN_User_Part","excerpt":"part of Signaling System No. 7 (SS7), which is used to set up telephone calls in the public switched telephone network (PSTN)","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"ISUP is the SS7 part used to set up PSTN calls; the Initial Address Message seizes a circuit and carries the called number; the Address Complete Message is returned when the far phone rings; the Answer Message is sent on answer when charging normally starts; Release and Release Complete free the cir","domain":"en.wikipedia.org","publisherName":"Wikipedia"},{"title":"ITU-T Q.700: Introduction to CCITT Signalling System No. 7","url":"https://www.itu.int/rec/T-REC-Q.700","excerpt":"Introduction to CCITT Signalling System No. 7","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-18T00:00:00","supportsText":"Signalling System No. 7 is standardized in the ITU-T Q.700 series as the out-of-band signaling system that sets up, routes, and releases PSTN calls; in force","domain":"itu.int","publisherName":"International Telecommunication Union (ITU-T)"},{"title":"ITU-T G.711: Pulse code modulation (PCM) of voice frequencies","url":"https://www.itu.int/rec/T-REC-G.711","excerpt":"Pulse code modulation (PCM) of voice frequencies","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-18T00:00:00","supportsText":"G.711 defines pulse code modulation encoding of voice frequencies, the baseline digital voice channel of the telephone network; in force","domain":"itu.int","publisherName":"International Telecommunication Union (ITU-T)"},{"title":"ITU-T E.164: The international public telecommunication numbering plan","url":"https://www.itu.int/rec/T-REC-E.164","excerpt":"The international public telecommunication numbering plan","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-18T00:00:00","supportsText":"E.164 is the international public telecommunication numbering plan that gives every line a unique, routable number built from a country code and national digits; in force","domain":"itu.int","publisherName":"International Telecommunication Union (ITU-T)"},{"title":"The Plesiochronous Digital Hierarchy","url":"https://www.technologyuk.net/telecommunications/communication-technologies/plesiochronous-digital-hierarchy.shtml","excerpt":"The digitisation of an analogue voice channel into a 64 kbps digital channel (designated as digital signalling level zero or simply DS0) using pulse code modulation (PCM)","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"A digitized analog voice channel is a 64 kbit/s DS0 using PCM; a T1/DS1 carries 24 DS0 channels at 1.544 Mbit/s; an E1 has 32 timeslots (30 for voice) at 2.048 Mbit/s; time-division multiplexing combines voice channels onto a trunk line","domain":"technologyuk.net","publisherName":"TechnologyUK"},{"title":"Atlantech Online's Voice Switch: What You Should Know","url":"https://www.atlantech.net/blog/atlantech-onlines-voice-switch-what-you-should-know","excerpt":"If you place a call to someone who is not on the same class 5 switch as you, your call may be routed through one or more class 4 switches to reach its final destination.","quoteText":null,"sourceRole":"SUPPORTING","verifiedAt":"2026-07-18T00:00:00","supportsText":"Class 5 switches at the local telephone company's central office directly serve subscribers, and Class 4 (tandem) switches interconnect central offices for long-distance calls in the PSTN; a call to a line on a different Class 5 switch may route through one or more Class 4 switches. Corroborated by ","domain":"atlantech.net","publisherName":"Atlantech Online"}],"revisions":[{"revisedAt":"2026-07-18T15:31:57.455+00:00","versionLabel":"Originally published","note":"","reviewedBy":null}],"relatedAnswers":[{"id":"64a64f8a-a710-4a70-b21b-292c1eb247b6","slug":"what-is-the-pstn-public-switched-telephone-network","question":"What is the PSTN (Public Switched Telephone Network)?","publishedAt":"2026-09-23T15:17:01.345","confidenceScore":93,"confidenceLabel":"High","industry":{"id":"ff619d7c-d7d7-485e-a05a-53fba07f33ed","slug":"telecommunications","label":"Telecommunications","description":"Business voice, fiber, UCaaS, and network services"},"topic":{"slug":"pstn","label":"PSTN","description":"The public switched telephone network: how it works, POTS retirement, and the move to VoIP.","schemaKind":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"snippet":"The PSTN is the combined telephone networks of the world's carriers, connected so that a number on one can be dialed from another. No single company owns it. Landline and mobile carriers both belong to it, and a shared numbering system holds it together. It began with copper lines and circuit switches, and parts of it now run on IP.","url":"/q/what-is-the-pstn-public-switched-telephone-network"},{"id":"ef14e83f-e61d-4263-80d2-b759e8ba8101","slug":"what-is-the-difference-between-pstn-and-pbx","question":"What is the difference between PSTN and PBX?","publishedAt":"2026-09-19T15:32:04.773","confidenceScore":93,"confidenceLabel":"High","industry":{"id":"ff619d7c-d7d7-485e-a05a-53fba07f33ed","slug":"telecommunications","label":"Telecommunications","description":"Business voice, fiber, UCaaS, and network services"},"topic":{"slug":"pstn","label":"PSTN","description":"The public switched telephone network: how it works, POTS retirement, and the move to VoIP.","schemaKind":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"snippet":"The PSTN is the public telephone network that connects phone numbers across carriers. A PBX is one organization's own phone system, which gives users extensions and controls how calls are routed. A business normally uses both: the PBX handles calls inside the organization, and a PSTN connection carries calls to and from outside phone numbers.","url":"/q/what-is-the-difference-between-pstn-and-pbx"},{"id":"895bda56-c48a-429f-8f13-53dc29c09b67","slug":"which-legacy-devices-are-still-at-risk-when-pstn-pots-lines-are-retired-or","question":"Which legacy devices are still at risk when PSTN/POTS lines are retired or moved to VoIP?","publishedAt":"2026-08-29T16:01:24.898","confidenceScore":90,"confidenceLabel":"High","industry":{"id":"ff619d7c-d7d7-485e-a05a-53fba07f33ed","slug":"telecommunications","label":"Telecommunications","description":"Business voice, fiber, UCaaS, and network services"},"topic":{"slug":"pstn","label":"PSTN","description":"The public switched telephone network: how it works, POTS retirement, and the move to VoIP.","schemaKind":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"snippet":"Analog devices that use a copper phone line as their call path are at risk: alarm dialers, elevator phones, telecare and medical alert base stations, fax machines, dial-up card terminals, entry intercoms and building monitoring modems. A device whose line is retired or moved to VoIP needs a path that carries its signals and has backup power, checked by a test arranged with the device's vendor or monitoring company.","url":"/q/which-legacy-devices-are-still-at-risk-when-pstn-pots-lines-are-retired-or"},{"id":"3cc4d7a6-6aed-4446-8dd8-13e022c3e1ba","slug":"is-pstn-secure-and-can-pstn-calls-be-encrypted","question":"Is PSTN secure, and can PSTN calls be encrypted?","publishedAt":"2026-07-11T16:01:27.137","confidenceScore":80,"confidenceLabel":"High","industry":{"id":"ff619d7c-d7d7-485e-a05a-53fba07f33ed","slug":"telecommunications","label":"Telecommunications","description":"Business voice, fiber, UCaaS, and network services"},"topic":{"slug":"pstn","label":"PSTN","description":"The public switched telephone network: how it works, POTS retirement, and the move to VoIP.","schemaKind":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"snippet":"The PSTN gives a call no end-to-end encryption. Its security rests on how each carrier on the path protects its own network, which a caller can't see. Individual legs can be encrypted, such as the IP leg between a phone or app and its provider, but each protection runs between two devices that share the keys and ends at the device that decrypts the call to pass it on. End-to-end encryption needs compatible equipment or software at both ends.","url":"/q/is-pstn-secure-and-can-pstn-calls-be-encrypted"}],"contributorStats":{"verifiedAnswers":149,"openDisputes":0},"schemaJson":{"@context":"https://schema.org","@type":"Question","name":"How does the PSTN work?","text":"How does the PSTN work?","url":"https://www.answerstack.io/q/how-does-the-pstn-work","answerCount":1,"datePublished":"2026-07-18T15:31:57.455","about":[{"@type":"Thing","name":"PSTN"},{"@type":"Thing","name":"Telecommunications"}],"acceptedAnswer":{"@type":"Answer","text":"The PSTN connects a call in two layers that run side by side. Signaling messages pass from switch to switch to find the dialed number and ring it. A voice path then opens between the two phones and stays up until someone hangs up. On the older parts of the network, that path is a reserved 64 kbit/s circuit. On the newer parts it is a stream of IP packets, and gateways convert between the two where a call passes from one kind of network to the other.","url":"https://www.answerstack.io/q/how-does-the-pstn-work","upvoteCount":0,"datePublished":"2026-07-18T15:31:57.455","dateModified":"2026-07-18T00:00:00","citation":[{"@type":"CreativeWork","name":"Public switched telephone network","url":"https://en.wikipedia.org/wiki/Public_switched_telephone_network"},{"@type":"CreativeWork","name":"PSTN (public switched telephone network)","url":"https://www.techtarget.com/searchnetworking/definition/PSTN"},{"@type":"CreativeWork","name":"Circuit switching","url":"https://en.wikipedia.org/wiki/Circuit_switching"},{"@type":"CreativeWork","name":"Telephone exchange","url":"https://en.wikipedia.org/wiki/Telephone_exchange"},{"@type":"CreativeWork","name":"Dual-tone multi-frequency signaling","url":"https://en.wikipedia.org/wiki/Dual-tone_multi-frequency_signaling"},{"@type":"CreativeWork","name":"ISDN User Part","url":"https://en.wikipedia.org/wiki/ISDN_User_Part"},{"@type":"CreativeWork","name":"ITU-T Q.700: Introduction to CCITT Signalling System No. 7","url":"https://www.itu.int/rec/T-REC-Q.700"},{"@type":"CreativeWork","name":"ITU-T G.711: Pulse code modulation (PCM) of voice frequencies","url":"https://www.itu.int/rec/T-REC-G.711"},{"@type":"CreativeWork","name":"ITU-T E.164: The international public telecommunication numbering plan","url":"https://www.itu.int/rec/T-REC-E.164"},{"@type":"CreativeWork","name":"The Plesiochronous Digital Hierarchy","url":"https://www.technologyuk.net/telecommunications/communication-technologies/plesiochronous-digital-hierarchy.shtml"},{"@type":"CreativeWork","name":"Atlantech Online's Voice Switch: What You Should Know","url":"https://www.atlantech.net/blog/atlantech-onlines-voice-switch-what-you-should-know"}]}}}