SIP trunking is a service that connects a business phone system to the public telephone network over an IP connection. A carrier sells it, and it takes the place of the telephone lines that used to do that job. Your phone system keeps handling extensions, voicemail and call routing. The trunk gives it a way to call outside phone numbers and to take calls from them. The name comes from the Session Initiation Protocol, or SIP, which the two ends use to set calls up and end them.
\n","summary":"SIP trunking is a carrier service that connects a business phone system to the public telephone network over an IP connection. The PBX keeps handling extensions and routing, SIP messages set calls up, and RTP carries the audio. Phone numbers and call capacity are counted separately from the trunk.","publishedAt":"2026-07-18T14:40:06.195","verifiedAt":"2026-09-14T00:00:00","editorialStatus":"APPROVED","lastReviewedAt":"2026-10-10T03:37:25.503","nextReviewDueAt":"2026-12-14T00:00:00","templateVersion":"v2","aliases":["What is a SIP trunk?","How does SIP trunking work?","SIP trunking definition","What does SIP trunking mean?","Is SIP trunking the same as VoIP?","SIP trunk vs PRI"],"confidenceScore":86,"confidenceLabel":"High","canonicalUrl":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"contributorOrganizationProfile":null,"contributorPerson":null,"sections":[{"id":"a9946118-1298-449f-b44b-2a2acd2268d8","sectionKey":"where_trunk_sits","sectionType":"markdown_section","heading":"Where the trunk sits","introMarkdown":"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](https://www.answerstack.io/q/what-is-the-pstn-public-switched-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](https://www.answerstack.io/q/what-is-the-difference-between-pstn-and-pbx).\n\nTwilio describes its own trunking product in terms of the same boundary. Its [API reference](https://www.twilio.com/docs/sip-trunking/api) 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.\n\nThat boundary also explains how a carrier packages its phone service. Atlantech Online's [business phone page](https://www.atlantech.net/solutions/voice/phone) 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](https://www.atlantech.net/blog/sip-trunking-business-phone-service) 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.\n\nThe system at your end does not have to be a cabinet in a back room. Twilio's [trunking requirements](https://www.twilio.com/docs/sip-trunking) 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.","introHtml":"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.
\nTwilio 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.
\nThat 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.
\nThe 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.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":10},{"id":"fc0e7aa7-400c-488a-92e3-7c7f36ac6c0a","sectionKey":"name_meaning","sectionType":"markdown_section","heading":"What the name describes","introMarkdown":"[RFC 3261](https://www.rfc-editor.org/rfc/rfc3261.html), 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.\n\nThe 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](https://www.answerstack.io/q/how-does-sip-trunking-work).\n\nThe 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.","introHtml":"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.
\nThe 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.
\nThe 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.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":20},{"id":"addb231d-3ee4-434e-bf39-11f122bfe977","sectionKey":"counted_separately","sectionType":"markdown_section","heading":"What is counted separately from the trunk","introMarkdown":"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.\n\nTwilio'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.\n\n[SIP.US](https://www.sip.us/pricing/), which publishes its prices, adds capacity as a third item and sells it in [channels](https://www.answerstack.io/q/what-is-a-sip-trunk-channel). 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](https://www.answerstack.io/q/how-many-simultaneous-calls-do-i-need-or-get-on-a-sip-trunk) and the count of phone numbers are things to state when you ask for the quote.","introHtml":"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.
\nTwilio'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.
\nSIP.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.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":30},{"id":"c6c6c6c6-2e02-427d-ac31-b852fb1594c2","sectionKey":"terms_table","sectionType":"table_section","heading":"The terms around a SIP trunk","introMarkdown":null,"introHtml":null,"outroMarkdown":null,"outroHtml":null,"contentJson":{"rows":[{"cells":["PBX","The phone system that runs extensions, voicemail and call routing"]},{"cells":["SIP trunking","The carrier service that connects that system to the public telephone network over IP"]},{"cells":["SIP","The protocol that sets calls up, changes them and ends them"]},{"cells":["RTP","The protocol that carries the audio during a call"]},{"cells":["Channel","One call in progress, where a carrier sells capacity that way"]},{"cells":["Phone number","An address that routes a call to your trunk"]}],"columns":["Term","What it is"]},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":40},{"id":"96a87340-a9b1-46d4-bfb1-439286f1b9b4","sectionKey":"compared_services","sectionType":"markdown_section","heading":"How it differs from cloud PBX, PRI and VoIP","introMarkdown":"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](https://www.answerstack.io/q/what-is-the-difference-between-sip-trunking-and-cloud-pbx) is a comparison of who runs the phone system more than of how calls leave it.\n\nA 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](https://www.answerstack.io/q/what-is-the-difference-between-sip-trunking-and-pri), come down to capacity and to what your PBX can accept.\n\nVoIP 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](https://www.answerstack.io/q/what-is-the-difference-between-sip-trunking-and-voip) are not alternatives to each other.","introHtml":"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.
\nA 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.
\nVoIP 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.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":50},{"id":"772d64a6-72bc-47e3-9d99-19d684d7fcfd","sectionKey":"dependencies","sectionType":"markdown_section","heading":"What a SIP trunk depends on","introMarkdown":"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.\n\nSpare bandwidth is one part of the requirement. [RFC 3550](https://datatracker.ietf.org/doc/html/rfc3550), 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.\n\nThe 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.","introHtml":"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.
\nSpare 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.
\nThe 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.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":60}],"citations":[{"title":"RFC 3261: SIP: Session Initiation Protocol","url":"https://datatracker.ietf.org/doc/html/rfc3261","excerpt":null,"quoteText":"an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"SIP is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions including internet telephone calls; registration via REGISTER to a registrar that writes bindings into the location service consulted by a proxy","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"RFC 3550: RTP: A Transport Protocol for Real-Time Applications","url":"https://datatracker.ietf.org/doc/html/rfc3550","excerpt":null,"quoteText":"The first twelve octets are present in every RTP packet, while the list of CSRC identifiers is present only when inserted by a mixer.","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"RTP carries the real-time audio of a SIP call separately from signaling, with sequence numbers and timestamps; the RTP fixed header is 12 octets, used as an input to the per-call bandwidth calculation","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"RFC 5853: Requirements from Session Initiation Protocol (SIP) Session Border Control (SBC) Deployments","url":"https://datatracker.ietf.org/doc/html/rfc5853","excerpt":null,"quoteText":"SBCs often modify certain SIP headers and message bodies that proxies are not allowed to modify. Consequently, they are, by definition, B2BUAs (Back-to-Back User Agents).","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"SBCs modify headers and bodies proxies may not, and are therefore B2BUAs by definition; SBC functions group into perimeter defense, functions not available in the endpoints, and traffic management; the media path is independent of the signaling path","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"RFC 768: User Datagram Protocol","url":"https://datatracker.ietf.org/doc/html/rfc768","excerpt":null,"quoteText":"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.)","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"The UDP header is 8 octets (source port, destination port, length, checksum), one of the three header sizes in the per-call bandwidth calculation","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"RFC 791: Internet Protocol","url":"https://datatracker.ietf.org/doc/html/rfc791","excerpt":null,"quoteText":"Internet Header Length is the length of the internet header in 32 bit words, and thus points to the beginning of the data. Note that the minimum value for a correct header is 5.","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"The minimum correct IPv4 header is 5 32-bit words, that is 20 octets, the third header size in the per-call bandwidth calculation","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"ITU-T G.711: Pulse code modulation (PCM) of voice frequencies","url":"https://www.itu.int/rec/T-REC-G.711/","excerpt":null,"quoteText":"The nominal value recommended for the sampling rate is 8000 samples per second. ... Eight binary digits per sample should be used for international circuits.","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"G.711 specifies a nominal sampling rate of 8000 samples per second and eight binary digits per sample, which is the 64,000 bit/s payload used in the bandwidth worked example","domain":"itu.int","publisherName":"International Telecommunication Union (ITU-T)"},{"title":"ITU-T G.729: Coding of speech at 8 kbit/s using CS-ACELP","url":"https://www.itu.int/rec/T-REC-G.729/","excerpt":null,"quoteText":"Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear prediction (CS-ACELP)","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"G.729 codes speech at 8 kbit/s, the payload rate used in the compressed-codec side of the bandwidth worked example; status in force","domain":"itu.int","publisherName":"International Telecommunication Union (ITU-T)"},{"title":"ITU-T Recommendation I.412: ISDN user-network interfaces - Interface structures and access capabilities","url":"https://www.itu.int/rec/T-REC-I.412/","excerpt":null,"quoteText":"At the 1544 kbit/s primary rate the interface structure is 23 B + D. At the 2048 kbit/s primary rate the interface structure is 30 B + D.","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"The primary rate interface structure is 23 B + D at 1544 kbit/s and 30 B + D at 2048 kbit/s, with the D-channel at 64 kbit/s; this replaces the vendor-blog source previously used for the PRI channel count","domain":"itu.int","publisherName":"International Telecommunication Union"},{"title":"47 CFR 9.3 Definitions","url":"https://www.ecfr.gov/current/title-47/section-9.3","excerpt":null,"quoteText":"(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.","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"The four-part definition of interconnected VoIP service, plus the narrower 911-only variant covering services that only terminate calls to the PSTN","domain":"ecfr.gov","publisherName":"Office of the Federal Register (eCFR)"},{"title":"47 CFR 9.11: E911 Service (interconnected VoIP)","url":"https://www.ecfr.gov/current/title-47/section-9.11","excerpt":null,"quoteText":"(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","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"Current text as of September 15, 2026: paragraph (a) is reserved and paragraph (b) carries the interconnected VoIP E911 obligations (transmit all 911 calls with ANI and location, dispatchable location rules for fixed and non-fixed service, subscriber advisory and acknowledgment)","domain":"ecfr.gov","publisherName":"Electronic Code of Federal Regulations (National Archives / GPO)"},{"title":"Delete, Delete, Delete (FCC direct final rule), 91 FR 1400","url":"https://www.federalregister.gov/documents/2026/01/14/2026-00612/delete-delete-delete","excerpt":null,"quoteText":"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). ... 10. Section 9.11 is amended by removing and reserving paragraph (a).","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"Effective March 16, 2026, the rule removed and reserved 47 CFR 9.11(a) and 9.10(n) and revised 9.19; it states that 9.11(a) governed pre-2021 fixed and pre-2022 non-fixed VoIP and had been superseded by 9.11(b). Establishes that section 9.11 as a whole was not repealed.","domain":"federalregister.gov","publisherName":"Federal Register / Federal Communications Commission"},{"title":"47 CFR 9.16: General obligations, direct 911 dialing, notification, and dispatchable location","url":"https://www.ecfr.gov/current/title-47/section-9.16","excerpt":null,"quoteText":"a user may directly initiate a call to 911 from any station equipped with dialing facilities, without dialing any additional digit, code, prefix, or post-fix, including any trunk-access code such as the digit 9","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"Kari's Law direct 911 dialing without a trunk-access code, MLTS notification, and RAY BAUM's Act dispatchable location obligations bind manufacturers, importers, sellers, lessors, installers, managers, and operators of multi-line telephone systems","domain":"ecfr.gov","publisherName":"Electronic Code of Federal Regulations (National Archives / GPO)"},{"title":"47 CFR 9.19 Provision of reliable 911 service","url":"https://www.ecfr.gov/current/title-47/section-9.19","excerpt":null,"quoteText":"Any entity that provides covered 911 services, which are 911, E911, or NG911 services for which a failure would impede the real-time routing, delivery, or transfer of 911 traffic.","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"Defines covered 911 service provider as an entity whose failure would impede the real-time routing, delivery, or transfer of 911 traffic, which scopes this rule away from ordinary SIP trunk providers","domain":"ecfr.gov","publisherName":"eCFR, U.S. Government Publishing Office"},{"title":"47 CFR 64.6300 Definitions (caller ID authentication)","url":"https://www.ecfr.gov/current/title-47/section-64.6300","excerpt":null,"quoteText":"Means any service that is interconnected with the public switched telephone network and that furnishes voice communications to an end user using resources from the North American Numbering Plan","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"Definition of voice service: any service interconnected with the PSTN that furnishes voice communications to an end user using North American Numbering Plan resources, which is the category that makes a SIP trunk provider subject to 64.6301","domain":"ecfr.gov","publisherName":"eCFR, U.S. Government Publishing Office"},{"title":"47 CFR 64.6301 Caller ID authentication","url":"https://www.ecfr.gov/current/title-47/section-64.6301","excerpt":null,"quoteText":"a voice service provider shall fully implement the STIR/SHAKEN authentication framework in its internet Protocol networks.","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"STIR/SHAKEN implementation duties for voice service providers: SPC token and certificate, authentication of originated SIP calls, verification of terminated SIP calls, and the conditions for using a third-party authentication service","domain":"ecfr.gov","publisherName":"eCFR, U.S. Government Publishing Office"},{"title":"47 CFR 64.2003 Definitions (Subpart U, privacy of customer information)","url":"https://www.ecfr.gov/current/title-47/section-64.2003","excerpt":null,"quoteText":"For the purposes of this subpart, the term “telecommunications carrier” or “carrier” shall include an entity that provides interconnected VoIP service, as that term is defined in section 9.3 of these rules.","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"For CPNI purposes the term telecommunications carrier includes an entity that provides interconnected VoIP service as defined in 47 CFR 9.3","domain":"ecfr.gov","publisherName":"eCFR, U.S. Government Publishing Office"},{"title":"SIP Trunking Pricing","url":"https://www.sip.us/pricing/","excerpt":null,"quoteText":"$24.95 per channel per/month | $19.99 per channel per month. Includes local and long distance calls to the US48 states and Canada (excluding NWT) at no additional charge.","sourceRole":"PRIMARY","verifiedAt":"2026-09-15T00:00:00","supportsText":"SIP.US published prices read September 15, 2026: SIP channel $24.95 per channel per month month-to-month and $19.99 per channel per month on yearly plans, no set-up fee, including local and long distance to the US 48 states and Canada excluding NWT; US48 local DID $1.00 per month; nomadic e911 enhan","domain":"sip.us","publisherName":"SIP.US"},{"title":"Elastic SIP Trunking Pricing","url":"https://www.twilio.com/en-us/sip-trunking/pricing","excerpt":null,"quoteText":"United States - 48 States (Zone 1) $0.0100 / min; Local origination $0.0034 / min; Local number $1.1500 / mo; Emergency Calling $0.7500 / mo.","sourceRole":"PRIMARY","verifiedAt":"2026-09-15T00:00:00","supportsText":"Twilio published prices read September 15, 2026: termination United States 48 states (Zone 1) $0.0100 per minute, local origination $0.0034 per minute, local phone number $1.1500 per month, emergency calling $0.7500 per month per number; rates shown as starting points","domain":"twilio.com","publisherName":"Twilio"},{"title":"Elastic SIP Trunking","url":"https://www.twilio.com/docs/sip-trunking","excerpt":null,"quoteText":"You must configure a minimum of either an ACL or credential authentication. ... It is highly recommended that you configure User Credentials. IP ACL's alone does not protect against certain types of attacks.","sourceRole":"PRIMARY","verifiedAt":"2026-09-15T00:00:00","supportsText":"Elastic SIP Trunking supports IP access control lists and credential-list (username and password) authentication, requires at least one, and states that IP ACLs alone do not protect against certain types of attacks","domain":"twilio.com","publisherName":"Twilio"},{"title":"Flowroute pricing details","url":"https://www.flowroute.com/pricing/","excerpt":null,"quoteText":"Pay only for the minutes you use. With no limit on simultaneous calls, you can instantly scale to occasional or seasonal bursts in traffic.","sourceRole":"PRIMARY","verifiedAt":"2026-09-15T00:00:00","supportsText":"Flowroute published prices read September 15, 2026: domestic inbound $0.005 per minute starting at, outbound US lower 48 and Canada $0.00833 per minute starting at, DID $1.00 per number starting at, mandatory E911 fee $1.50 per month per E911-enabled US DID; the metered plan states no limit on simul","domain":"flowroute.com","publisherName":"Flowroute"},{"title":"Telecommunications fraud increased 12% in 2023 equating to an estimated $38.95 billion lost to fraud","url":"https://cfca.org/telecommunications-fraud-increased-12-in-2023-equating-to-an-estimated-38-95-billion-lost-to-fraud/","excerpt":null,"quoteText":"a 12% increase in fraud loss reported in 2023 as compared to 2021 equating to an estimated $38.95 billion lost in 2023 representing 2.5% of telecommunications revenues.","sourceRole":"SUPPORTING","verifiedAt":"2026-09-15T00:00:00","supportsText":"Industry association survey estimate: $38.95 billion in global telecom fraud losses in 2023, up 12 percent from 2021, about 2.5 percent of telecommunications revenues, with PBX fraud among five methods accounting for 51 percent of fraud issues. Labeled in the answer as a self-reported survey estimat","domain":"cfca.org","publisherName":"Communications Fraud Control Association (CFCA)"}],"revisions":[{"revisedAt":"2026-07-18T14:40:06.195+00:00","versionLabel":"Originally published","note":"","reviewedBy":null}],"relatedAnswers":[{"id":"77b3581e-fb3d-43da-aefa-a3ebd533d57d","slug":"how-many-simultaneous-calls-do-i-need-or-get-on-a-sip-trunk","question":"How many simultaneous calls do I need or get on a SIP trunk?","publishedAt":"2026-07-18T15:16:53.422","confidenceScore":82,"confidenceLabel":"High","industry":{"id":"ff619d7c-d7d7-485e-a05a-53fba07f33ed","slug":"telecommunications","label":"Telecommunications","description":"Business voice, fiber, UCaaS, and network services"},"topic":{"slug":"sip-trunking","label":"SIP Trunking","description":"SIP trunking for business voice: how it works, channels, bandwidth, and PBX connectivity.","schemaKind":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"snippet":"To size a SIP trunk, measure the calls in your busiest hour, convert that traffic to call paths with a blocking target, then check the result against the provider's plan, your equipment and your connection. What you get is the lowest of those limits.","url":"/q/how-many-simultaneous-calls-do-i-need-or-get-on-a-sip-trunk"},{"id":"bc3ff0a8-b843-46c0-a953-41aa557e1110","slug":"what-is-the-difference-between-sip-trunking-and-cloud-pbx","question":"What is the difference between SIP trunking and cloud PBX?","publishedAt":"2026-07-18T15:01:34.706","confidenceScore":84,"confidenceLabel":"High","industry":{"id":"ff619d7c-d7d7-485e-a05a-53fba07f33ed","slug":"telecommunications","label":"Telecommunications","description":"Business voice, fiber, UCaaS, and network services"},"topic":{"slug":"sip-trunking","label":"SIP Trunking","description":"SIP trunking for business voice: how it works, channels, bandwidth, and PBX connectivity.","schemaKind":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"snippet":"SIP trunking connects a phone system you run to the public telephone network. A cloud PBX is the phone system itself, run by a provider and billed per user.","url":"/q/what-is-the-difference-between-sip-trunking-and-cloud-pbx"},{"id":"97ab21fb-1842-45fa-90d8-e34e67a64794","slug":"what-is-the-difference-between-sip-trunking-and-voip","question":"What is the difference between SIP trunking and VoIP?","publishedAt":"2026-07-18T15:01:30.724","confidenceScore":86,"confidenceLabel":"High","industry":{"id":"ff619d7c-d7d7-485e-a05a-53fba07f33ed","slug":"telecommunications","label":"Telecommunications","description":"Business voice, fiber, UCaaS, and network services"},"topic":{"slug":"sip-trunking","label":"SIP Trunking","description":"SIP trunking for business voice: how it works, channels, bandwidth, and PBX connectivity.","schemaKind":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"snippet":"VoIP is a technology category: voice carried over data networks with internet protocols. SIP trunking is one carrier service built on it, connecting a phone system you run to the public telephone network. A SIP trunk is VoIP. The real choice appears when VoIP means a hosted phone service, and it turns on who runs the phone system.","url":"/q/what-is-the-difference-between-sip-trunking-and-voip"},{"id":"fa8568a0-bdf2-4a59-945f-3ebb2492ff83","slug":"how-does-sip-trunking-work","question":"How does SIP trunking work?","publishedAt":"2026-07-18T15:01:28.862","confidenceScore":92,"confidenceLabel":"High","industry":{"id":"ff619d7c-d7d7-485e-a05a-53fba07f33ed","slug":"telecommunications","label":"Telecommunications","description":"Business voice, fiber, UCaaS, and network services"},"topic":{"slug":"sip-trunking","label":"SIP Trunking","description":"SIP trunking for business voice: how it works, channels, bandwidth, and PBX connectivity.","schemaKind":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"snippet":"A SIP trunk links your PBX to a carrier over an IP network. SIP messages such as INVITE, 180 Ringing, 200 OK, ACK and BYE set up and end each call, and the audio travels separately as RTP packets. The carrier passes calls to and from other networks, and call capacity is a limit rather than a count of physical lines.","url":"/q/how-does-sip-trunking-work"}],"contributorStats":{"verifiedAnswers":149,"openDisputes":0},"schemaJson":{"@context":"https://schema.org","@type":"Question","name":"What is SIP trunking?","text":"What is SIP trunking?","url":"https://www.answerstack.io/q/what-is-sip-trunking","answerCount":1,"datePublished":"2026-07-18T14:40:06.195","about":[{"@type":"Thing","name":"SIP Trunking"},{"@type":"Thing","name":"Telecommunications"}],"acceptedAnswer":{"@type":"Answer","text":"SIP trunking is a service that connects a business phone system to the public telephone network over an IP connection. A carrier sells it, and it takes the place of the telephone lines that used to do that job. Your phone system keeps handling extensions, voicemail and call routing. The trunk gives it a way to call outside phone numbers and to take calls from them. The name comes from the Session Initiation Protocol, or SIP, which the two ends use to set calls up and end them.","url":"https://www.answerstack.io/q/what-is-sip-trunking","upvoteCount":0,"datePublished":"2026-07-18T14:40:06.195","dateModified":"2026-09-14T00:00:00","citation":[{"@type":"CreativeWork","name":"RFC 3261: SIP: Session Initiation Protocol","url":"https://datatracker.ietf.org/doc/html/rfc3261"},{"@type":"CreativeWork","name":"RFC 3550: RTP: A Transport Protocol for Real-Time Applications","url":"https://datatracker.ietf.org/doc/html/rfc3550"},{"@type":"CreativeWork","name":"RFC 5853: Requirements from Session Initiation Protocol (SIP) Session Border Control (SBC) Deployments","url":"https://datatracker.ietf.org/doc/html/rfc5853"},{"@type":"CreativeWork","name":"RFC 768: User Datagram Protocol","url":"https://datatracker.ietf.org/doc/html/rfc768"},{"@type":"CreativeWork","name":"RFC 791: Internet Protocol","url":"https://datatracker.ietf.org/doc/html/rfc791"},{"@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 G.729: Coding of speech at 8 kbit/s using CS-ACELP","url":"https://www.itu.int/rec/T-REC-G.729/"},{"@type":"CreativeWork","name":"ITU-T Recommendation I.412: ISDN user-network interfaces - Interface structures and access capabilities","url":"https://www.itu.int/rec/T-REC-I.412/"},{"@type":"CreativeWork","name":"47 CFR 9.3 Definitions","url":"https://www.ecfr.gov/current/title-47/section-9.3"},{"@type":"CreativeWork","name":"47 CFR 9.11: E911 Service (interconnected VoIP)","url":"https://www.ecfr.gov/current/title-47/section-9.11"},{"@type":"CreativeWork","name":"Delete, Delete, Delete (FCC direct final rule), 91 FR 1400","url":"https://www.federalregister.gov/documents/2026/01/14/2026-00612/delete-delete-delete"},{"@type":"CreativeWork","name":"47 CFR 9.16: General obligations, direct 911 dialing, notification, and dispatchable location","url":"https://www.ecfr.gov/current/title-47/section-9.16"},{"@type":"CreativeWork","name":"47 CFR 9.19 Provision of reliable 911 service","url":"https://www.ecfr.gov/current/title-47/section-9.19"},{"@type":"CreativeWork","name":"47 CFR 64.6300 Definitions (caller ID authentication)","url":"https://www.ecfr.gov/current/title-47/section-64.6300"},{"@type":"CreativeWork","name":"47 CFR 64.6301 Caller ID authentication","url":"https://www.ecfr.gov/current/title-47/section-64.6301"},{"@type":"CreativeWork","name":"47 CFR 64.2003 Definitions (Subpart U, privacy of customer information)","url":"https://www.ecfr.gov/current/title-47/section-64.2003"},{"@type":"CreativeWork","name":"SIP Trunking Pricing","url":"https://www.sip.us/pricing/"},{"@type":"CreativeWork","name":"Elastic SIP Trunking Pricing","url":"https://www.twilio.com/en-us/sip-trunking/pricing"},{"@type":"CreativeWork","name":"Elastic SIP Trunking","url":"https://www.twilio.com/docs/sip-trunking"},{"@type":"CreativeWork","name":"Flowroute pricing details","url":"https://www.flowroute.com/pricing/"},{"@type":"CreativeWork","name":"Telecommunications fraud increased 12% in 2023 equating to an estimated $38.95 billion lost to fraud","url":"https://cfca.org/telecommunications-fraud-increased-12-in-2023-equating-to-an-estimated-38-95-billion-lost-to-fraud/"}]}}}