Skip to content
Answer Stack
Open menu

What is the difference between PSTN and VoIP?

✓ Verified Last reviewed by AnswerStack Next review due Dec 14, 2026

Every claim is sourced below

The PSTN is a network, and VoIP is a way of carrying a call. The public switched telephone network is the carriers' networks, connected so that a number on one can be dialed from another. Voice over IP sends a conversation as data packets over an IP network. The two overlap. A VoIP service can connect to the PSTN, and carriers use IP inside the PSTN itself. The practical comparison is narrower: a traditional phone line against a phone service delivered over a data connection.

One describes reach and the other describes transport

The PSTN is defined by who it connects. US rules, in 47 CFR 20.3, describe the public switched network as common carrier switched networks, by wire or radio, that use the North American Numbering Plan. They name local, long-distance and mobile carriers. The definition says nothing about copper or about circuits. A service that gives you a public phone number reaches this network through a carrier.

VoIP is defined by the way a call is carried, and two internet standards show how it is done. RFC 3261 defines SIP, the Session Initiation Protocol, as a signaling protocol for creating, modifying and ending sessions such as internet telephone calls. RFC 3550 defines RTP, which carries the audio in packets. Each packet has a sequence number and a timestamp so the receiver can rebuild the stream.

Because one term describes reach and the other describes transport, a call can be VoIP without touching the PSTN, and it can be both at once.

Microsoft Teams shows the first case. Microsoft's Teams Phone overview says calls between Teams users in the same organization are handled inside Teams by default and never go to the PSTN. To call an outside phone number, the organization needs a PSTN connection from Microsoft or another operator, which supplies the phone numbers and the access. Microsoft treats that connection as separate from the Teams Phone license.

US rules have a name for the second case: interconnected VoIP. 47 CFR 9.3 defines it as a two-way voice service that needs a broadband connection and IP-compatible equipment, and that lets users receive calls from the PSTN and place calls to it. A business phone service that runs over broadband and makes and receives calls on public phone numbers meets that description.

IP inside the public network

Carriers have been moving their own networks to IP as well. In a 2025 notice on IP interconnection, the FCC describes IP-based voice networks that carry VoIP calls from end to end over managed IP cores and session border controllers. When one of those calls has to cross older equipment, it is "handed off via a media gateway to TDM at the network edge." TDM, or time-division multiplexing, is the circuit technology of the traditional telephone network.

So the PSTN today contains both technologies. A call between two public numbers might be carried as packets the whole way, or it might be converted to a circuit somewhere along its route. It is a PSTN call in both cases, because it runs between public numbers across the carriers' interconnected networks.

The practical differences appear when you compare the two ways a phone service can reach you. One is the traditional line, analog or digital, delivered on the telephone company's own circuit. The other is a service that arrives over your data connection. Both give you a public phone number.

Point of comparison Traditional line IP phone service
How the voice travels An analog signal on copper access; reserved 64 kbit/s channels on digital circuits Packets that share an IP network with other traffic
How the call is set up Signaling on the line or circuit; SS7 between switches on the circuit network For example, SIP messages between phones, servers and carriers
What you plug in An analog phone, or a PBX on a carrier circuit An IP phone, an app, or an adapter for an analog device
Power On an analog line, some basic corded phones run on power from the exchange The router and phone on a broadband connection need local power or a battery
Where it works At the address the line is wired to It can follow the user to another connection, depending on the service

What those differences mean in use

On a traditional digital circuit, voice coded under ITU-T G.711 takes 64,000 bits per second, and that capacity is held for the call whether or not anyone is speaking. Nothing else competes for it.

RTP makes no such promise. RFC 3550 says the protocol does not guarantee delivery or prevent packets from arriving out of order, and that it relies on the network underneath for timely delivery. Call quality on an IP service therefore depends on the connection it runs over. That is the reason to look at the internet connection VoIP needs and at how much bandwidth a call uses before a move.

Power is the difference people tend to discover during an outage. The UK government's guidance on digital landlines says an analog landline carries low-voltage power from the exchange, enough to run some basic corded handsets through a local power cut. The digital landlines replacing it run over a broadband connection and can't carry that power, so the phone and the router need a battery or a generator to keep working.

Location works differently too. A traditional line ends at one address. SIP was designed so that a user can keep a single identifier wherever they connect, and an IP phone service can follow a laptop or a desk phone to another site. Emergency calling has to account for that. 47 CFR 9.3, the rule that defines interconnected VoIP, also defines a registered location: the most recent physical location a provider has on file for the user.

How a business ends up with both

A business doesn't have to choose between the PSTN and VoIP. It keeps its public numbers and decides how the connection to its carrier is delivered.

Atlantech Online, a business phone and fiber provider, sets out the options on its business phone service page. A customer can use Atlantech's cloud PBX, or keep its own PBX and take service over SIP, PRI or analog lines. A 2023 Atlantech post on SIP trunking divides customers the same way. SIP trunks go to customers that have their own VoIP PBX, and PRI and analog lines go to customers with a legacy PBX.

In every one of those arrangements the customer's numbers are on the PSTN. The handoff is what differs. PRI and analog lines are circuits, and a SIP trunk or a cloud PBX arrives as IP. The PBX is a third thing again: the private system that handles extensions and call routing inside the company.

Moving the phone system to IP doesn't settle every line in a building. Atlantech's POTS replacement page compares several ways to replace copper lines. That comparison shows SIP trunking at a typical $15 to $25 per line per month for organizations keeping a PBX, and notes that it requires a PBX and is "not a life-safety path." For fire panels and elevator phones it points to its managed appliance, at a typical $15 to $45 per line per month plus hardware. The appliance gives each device an analog line and carries the calls over broadband with cellular backup.

Both options carry calls over IP, and both reach the PSTN. They differ in what they present to the equipment at your end, which is why an alarm panel and a desk phone in the same building can need different services.

Sources

RFC 3261: SIP: Session Initiation Protocol

IETF

Independent Verified Sep 15, 2026 Supports: SIP is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions including internet telephone calls; it covers user location, availability, capabilities, session setup and session management; it provides a registration function for users to upload current lo

“an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls... SIP transparently supports name mapping and redirection services, which supports personal mobility - users can maintain a single externally visible identifier regardless of their network location.”

RFC 3550: RTP: A Transport Protocol for Real-Time Applications

IETF

Independent Verified Sep 15, 2026 Supports: RTP provides end-to-end network transport functions for real-time data such as audio, and the first twelve octets of the RTP header are present in every packet, which is the 12-octet figure used in the bandwidth derivation

“RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data... The first twelve octets are present in every RTP packet.”

RFC 3551: RTP Profile for Audio and Video Conferences with Minimal Control

IETF

Independent Verified Sep 15, 2026 Supports: Default packetization interval of 20 ms; silence suppression is optional ("Discontiguous transmission (silence suppression) MAY be used with any audio payload format") and receivers must assume senders may suppress; G.711 is carried as PCMA and PCMU at eight bits per sample; G.729 uses 10 ms frames

“For packetized audio, the default packetization interval SHOULD have a duration of 20 ms or one frame, whichever is longer... Since the ability to suppress silence is one of the primary motivations for using packets to transmit voice, the RTP header carries both a sequence number and a timestamp... Discontiguous transmission (silence suppression) MAY be used with any audio payload format.”

RFC 3389: Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN)

IETF

Independent Verified Sep 15, 2026 Supports: The comfort noise payload is primarily for codecs that do not support comfort noise themselves, G.711 among them, and the comfort noise analysis and synthesis, the voice activity detection and the discontinuous transmission algorithms are unspecified and implementation-specific, which is why silence

“The CN payload type is primarily for use with audio codecs that do not support comfort noise as part of the codec itself such as ITU-T Recommendations G.711, G.726, G.727, G.728, and G.722... The comfort noise analysis and synthesis as well as the Voice Activity Detection (VAD) and DTX algorithms are unspecified and left implementation-specific.”

RFC 791: Internet Protocol

IETF

Independent Verified Sep 15, 2026 Supports: A typical IPv4 header is 20 octets (minimum IHL of five 32-bit words), the header size used in the per-call bandwidth derivation

“The maximal internet header is 60 octets, and a typical internet header is 20 octets... Note that the minimum value for a correct header is 5.”

RFC 768: User Datagram Protocol

IETF

Independent Verified Sep 15, 2026 Supports: The UDP header is eight octets, stated as the minimum value of the length field, the header size used in the per-call bandwidth derivation

“Length is the length in octets of this user datagram including this header and the data. (This means the minimum value of the length is eight.)”

ITU-T G.711: Pulse code modulation (PCM) of voice frequencies

International Telecommunication Union (ITU-T)

Independent Verified Sep 15, 2026 Supports: G.711 is in force and specifies a nominal sampling rate of 8,000 samples per second with eight binary digits per sample for international circuits, which is the 64 kbit/s encoding shared by the PSTN digital core and VoIP; text read in the free ITU PDF of the Recommendation reached from this page

“The nominal value recommended for the sampling rate is 8000 samples per second... Eight binary digits per sample should be used for international circuits.”

ITU-T G.729: Coding of speech at 8 kbit/s using CS-ACELP

International Telecommunication Union (ITU-T)

Independent Verified Sep 15, 2026 Supports: G.729 codes speech at 8 kbit/s and is in force in its 2012 edition; Annex B is listed as a silence compression scheme for G.729, establishing that silence handling is a separate, optional codec feature

“G.729 (06/12) Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear prediction (CS-ACELP) ... In force. G.729 Annex B: A silence compression scheme for G.729.”

ITU-T Recommendation Q.700 (03/93): Introduction to CCITT Signalling System No. 7

International Telecommunication Union (ITU-T)

Independent Verified Sep 15, 2026 Supports: SS7 is common channel signaling in which a single channel conveys labelled messages relating to a multiplicity of circuits; its listed applications include the PSTN, ISDN, service control points, mobile networks and network operations; the system is optimized for operation over 64 kbit/s digital cha

“Examples of applications supported by SS No. 7 are: - PSTN; - ISDN; - Interaction with Network Databases, Service Control Points for service control; - Mobiles (Public Land Mobile Network); - Operations Administration and Maintenance of Networks. The signalling system is optimized for operation over 64 kbit/s digital channels.”

47 CFR 9.3, Definitions: Interconnected VoIP service

Electronic Code of Federal Regulations (eCFR), Office of the Federal Register

Independent Verified Sep 15, 2026 Supports: The four-part definition of interconnected VoIP service, and the wider 911-only variant that reaches services which only terminate calls to the PSTN

“An interconnected Voice over Internet Protocol (VoIP) service is a service that: (i) Enables real-time, two-way voice communications; (ii) Requires a broadband connection from the user's location; (iii) Requires internet protocol-compatible customer premises equipment (CPE); and (iv) Permits users generally to receive calls that originate on the public switched telephone network and to terminate calls to the public switched telephone network.”

47 CFR 9.4, Obligation to transmit 911 calls

Electronic Code of Federal Regulations (eCFR), Office of the Federal Register

Independent Verified Sep 15, 2026 Supports: Telecommunications carriers are covered by their own 911 transmission rule, separate from the interconnected VoIP rule in section 9.11

“All telecommunications carriers shall transmit all 911 calls to a PSAP, to a designated statewide default answering point, or to an appropriate local emergency authority as set forth in Sec. 9.5.”

47 CFR 9.11, E911 service requirements for interconnected VoIP

Electronic Code of Federal Regulations (eCFR), Office of the Federal Register

Independent Verified Sep 15, 2026 Supports: Paragraph (a) now reads [Reserved] with a source note at 91 FR 1403, January 14, 2026; paragraph (b) carries the E911 duties: transmission of all 911 calls with ANI and location, automated dispatchable location for fixed service, Registered or Alternative Location Information for non-fixed service,

“(a) [Reserved] ... Each interconnected VoIP service provider shall specifically advise every subscriber, both new and existing, prominently and in plain language, of the circumstances under which E911 service may not be available... Such circumstances include, but are not limited to... broadband connection failure, loss of electrical power... [84 FR 66760, Dec. 5, 2019, as amended at 85 FR 78022, Dec. 3, 2020; 91 FR 1403, Jan. 14, 2026]”

47 CFR 20.3, Definitions: Public switched network

Electronic Code of Federal Regulations (eCFR), Office of the Federal Register

Independent Verified Sep 15, 2026 Supports: The FCC's definition of the public switched network covers any common carrier switched network, by wire or radio, including local exchange carriers, interexchange carriers and mobile service providers using the North American Numbering Plan, which is why a fixed-line table cannot describe the whole

“Public switched network. Any common carrier switched network, whether by wire or radio, including local exchange carriers, interexchange carriers, and mobile service providers, that uses the North American Numbering Plan in connection with the provision of switched services.”

Delete, Delete, Delete (FCC direct final rule), 91 FR 1400

Federal Register / Federal Communications Commission

Independent Verified Sep 15, 2026 Supports: Published January 14, 2026 and effective March 16, 2026 without further action unless significant adverse comment was received by February 3, 2026; removes 47 CFR 9.11(a) on the ground that it governed fixed VoIP before 2021 and non-fixed VoIP before 2022 and was superseded by 9.11(b); records that

“Effective March 16, 2026 without further action, unless significant adverse comment is received by February 3, 2026... One such rule provision, 47 CFR 9.20, has sunset by operation of law and ceased to be effective on September 1, 2025... Section 9.11(a) regulates fixed VoIP services prior to 2021 and non-fixed VoIP services prior to 2022, dates which have passed and, in any case, specifies regulations that have been superseded by subsection 9.11(b).”

Data Breach Reporting Requirements, 89 FR 9968

Federal Communications Commission via the Federal Register

Independent Verified Sep 15, 2026 Supports: The Commission's own statement that it has not broadly addressed the statutory classification of interconnected VoIP, that a provider may offer VoIP on a Title II basis voluntarily, and that it used Title I ancillary jurisdiction to extend CPNI rules to interconnected VoIP

“Although the Commission has not broadly addressed the statutory classification of interconnected VoIP as a general matter, it has consistently recognized that a provider may offer VoIP on a Title II basis if it voluntarily ``holds itself out as a telecommunications carrier and complies with appropriate Federal and State requirements.'' But because the Commission generally had not classified interconnected VoIP, the Commission also exercised its Title I ancillary jurisdiction to extend its CPNI rules to interconnected VoIP services.”

Information Collection Being Reviewed by the FCC (section 9.20 backup power), 89 FR 65880

Federal Communications Commission via the Federal Register

Independent Verified Sep 15, 2026 Supports: Describes the section 9.20 backup power obligation as applying to providers of facilities-based fixed, residential voice services that are not line-powered, and lists the required disclosures: availability of backup power sources, service limitations with and without backup power, purchase and repla

“Section 9.20 of the Commission's rules places limited backup power obligations on providers of facilities-based fixed, residential voice services that are not line-powered to ensure that such service providers meet their obligation to provide access to 911 service during a power outage... we require providers to disclose to subscribers the following information: (1) availability of backup power sources; (2) service limitations with and without backup power during a power outage; (3) purchase and replacement options; (4) expected backup power duration.”

UK transition from analogue to digital landlines

GOV.UK

Independent Verified Sep 15, 2026 Supports: UK providers are replacing the technology behind fixed telephone networks, most customers are expected to have switched by the end of January 2027, and a landline provider should offer a solution such as a battery back-up unit so emergency calls still work during a power cut; page last updated 28 Fe

“Communications providers in the UK are replacing the technology they use to provide fixed telephone networks (landlines). For most customers, the upgrade is expected to be complete by January 2027... your landline provider should offer you a solution such as a battery back-up unit. This will mean that you can still make emergency calls during a power cut.”

Time for a big switch-up as PSTN switch-off looms

Openreach

Primary source Verified Sep 15, 2026 Supports: Openreach's own account of its migration program: BT Group will retire the PSTN network by 31 January 2027, Openreach is withdrawing the Wholesale Line Rental products that run over it, and legacy WLR prices rise 20% on 1 April 2026, a further 40% on 1 July 2026 and a final 40% on 1 October 2026

“Openreach's parent, BT Group will retire the PSTN network by 31st January 2027 - and Openreach has been working to withdraw any Openreach products that work over it... April 1, 2026: Prices rise by 20%. July 1, 2026: Prices rise by a further 40%. October 1, 2026: A final 40% increase.”

What is PSTN (public switched telephone network)?

Informa TechTarget

Supporting Verified Sep 15, 2026 Supports: Reference description of the PSTN as interconnected circuit-switched networks that today combine telephone lines, fiber optic cables, switching centers, cellular networks, satellites and cable systems, and of carriers transitioning to all-IP to avoid running two infrastructures; used only for the co

“PSTN (public switched telephone network) is the world's collection of interconnected voice-oriented public telephone networks via traditional circuit-switched networks... It combines multiple worldwide telephone networks, including the following: Telephone lines. Fiber optic cables. Switching centers. Cellular networks. Satellites. Cable systems.”

Revision history

1 revision since publication
Originally published