VoIP is a technology: voice carried over data networks using internet protocols. SIP trunking is one service built on it, in which a carrier connects a business phone system to the public telephone network over IP. A SIP trunk is VoIP, so the two are not alternatives. The question turns into a choice only when "VoIP" is shorthand for a hosted phone service, and then the difference is who runs the phone system.
\n","summary":"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.","publishedAt":"2026-07-18T15:01:30.724","verifiedAt":"2026-09-14T00:00:00","editorialStatus":"APPROVED","lastReviewedAt":"2026-10-10T03:37:25.503","nextReviewDueAt":"2026-12-14T00:00:00","templateVersion":"v2","aliases":["SIP trunking vs VoIP","Is SIP trunking the same as VoIP?","Is a SIP trunk VoIP?","What is the difference between SIP and VoIP?","Is VoIP the same as SIP trunking?","Difference between SIP trunking and hosted VoIP","SIP trunk or VoIP: which is which?"],"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":"94427967-dacb-4bf7-b80c-617d60a1f5f4","sectionKey":"terms_levels","sectionType":"markdown_section","heading":"Four terms at different levels","introMarkdown":"The [NIST glossary](https://csrc.nist.gov/glossary/term/voip) takes its definition of VoIP from CNSSI 4009, a federal glossary. It reads \"a group of technologies used to provide voice/video/collaboration communications over a data network using internet protocols.\" The definition names no product and no single protocol. An IP desk phone calling a colleague through an IP PBX fits it, and so does a call one carrier hands to another over IP. We go further into the category in [what VoIP is](https://www.answerstack.io/q/what-is-voip).\n\nSIP sits one level down, as a protocol used inside that group of technologies. [RFC 3261](https://www.rfc-editor.org/rfc/rfc3261.html), the standard that defines it, describes a protocol that can establish, modify and terminate sessions such as internet telephone calls. The same document says \"SIP does not provide services.\" Somebody has to build a service from it.\n\n[SIP trunking](https://www.answerstack.io/q/what-is-sip-trunking) is one such service. Twilio's [API reference](https://www.twilio.com/docs/sip-trunking/api) describes its trunking product as connectivity that lets IP-based communications infrastructure reach the PSTN, the public switched telephone network. A hosted phone service is another service built with the same technology. When someone offers you \"VoIP\" as an alternative to SIP trunks, the word has to mean a hosted service for the comparison to make sense.","introHtml":"The NIST glossary takes its definition of VoIP from CNSSI 4009, a federal glossary. It reads "a group of technologies used to provide voice/video/collaboration communications over a data network using internet protocols." The definition names no product and no single protocol. An IP desk phone calling a colleague through an IP PBX fits it, and so does a call one carrier hands to another over IP. We go further into the category in what VoIP is.
\nSIP sits one level down, as a protocol used inside that group of technologies. RFC 3261, the standard that defines it, describes a protocol that can establish, modify and terminate sessions such as internet telephone calls. The same document says "SIP does not provide services." Somebody has to build a service from it.
\nSIP trunking is one such service. Twilio's API reference describes its trunking product as connectivity that lets IP-based communications infrastructure reach the PSTN, the public switched telephone network. A hosted phone service is another service built with the same technology. When someone offers you "VoIP" as an alternative to SIP trunks, the word has to mean a hosted service for the comparison to make sense.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":10},{"id":"4e35f3eb-9d8a-4173-9d4c-b6c1db418790","sectionKey":"terms_table","sectionType":"table_section","heading":"What each term names","introMarkdown":null,"introHtml":null,"outroMarkdown":null,"outroHtml":null,"contentJson":{"rows":[{"cells":["VoIP","Technology category","Voice carried over data networks with internet protocols"]},{"cells":["SIP","Protocol","The messages that set a call up, change it and end it"]},{"cells":["SIP trunking","Carrier service","A connection from a phone system you run to the public telephone network"]},{"cells":["Hosted phone service","Provider service","A phone system the provider runs for you"]}],"columns":["Term","Kind of thing","What it names"]},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":20},{"id":"974fd785-f726-4822-a131-8d18cd2ed745","sectionKey":"category_scope","sectionType":"markdown_section","heading":"Why a SIP trunk is VoIP, and what else is","introMarkdown":"A SIP trunk carries calls between your phone system and a carrier across an IP network. SIP messages manage each call and a separate stream of packets carries the audio, as we show in [how SIP trunking works](https://www.answerstack.io/q/how-does-sip-trunking-work). That puts the trunk inside the NIST definition.\n\nThe category reaches well past trunks, though. A call between two IP phones on an IP PBX is VoIP, and the PBX can complete it without sending anything to the carrier. VoIP does not have to use SIP, either. [RFC 3435](https://www.rfc-editor.org/rfc/rfc3435.html) defines MGCP, a different protocol in which a call agent controls the gateways that turn telephone-circuit audio into packets. And VoIP does not have to cross the public internet. NIST's definition says \"a data network,\" and Twilio's [trunking documentation](https://www.twilio.com/docs/sip-trunking) lists a private interconnect beside the public internet as a way to connect.\n\nSo \"VoIP\" on a quote or a spec sheet tells you calls travel over IP. It does not tell you which service you are being offered or who operates it.","introHtml":"A SIP trunk carries calls between your phone system and a carrier across an IP network. SIP messages manage each call and a separate stream of packets carries the audio, as we show in how SIP trunking works. That puts the trunk inside the NIST definition.
\nThe category reaches well past trunks, though. A call between two IP phones on an IP PBX is VoIP, and the PBX can complete it without sending anything to the carrier. VoIP does not have to use SIP, either. RFC 3435 defines MGCP, a different protocol in which a call agent controls the gateways that turn telephone-circuit audio into packets. And VoIP does not have to cross the public internet. NIST's definition says "a data network," and Twilio's trunking documentation lists a private interconnect beside the public internet as a way to connect.
\nSo "VoIP" on a quote or a spec sheet tells you calls travel over IP. It does not tell you which service you are being offered or who operates it.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":30},{"id":"9914bd0e-38ce-4215-8a09-11b93e3234c6","sectionKey":"hosted_service","sectionType":"markdown_section","heading":"When VoIP means a hosted phone service","introMarkdown":"Atlantech Online's [business phone page](https://www.atlantech.net/solutions/voice/phone) puts the two purchases side by side. You can connect to its cloud PBX, or, if you have your own PBX, it can deliver service by SIP, PRI or analog. The cloud PBX is VoIP, and so is the own-PBX route when the delivery is SIP. They differ in where the phone system lives.\n\nOn the first route, Atlantech runs the phone system. Its [UCaaS page](https://www.atlantech.net/solutions/voice/unified-communications) says the hosted phone system is in Atlantech's cloud in place of a PBX in your office. The phone page lists auto attendants, call distribution groups, a receptionist console and voicemail transcription, and says an administrator can add users, devices, phone numbers and call queues. It also describes unlimited nationwide calling, so public calling is part of the package. A buyer who asks for \"VoIP\" and means \"a phone system I don't have to run\" is describing this route.\n\nOn the second route, you run the PBX and Atlantech connects it. Its [April 2023 post on SIP trunks](https://www.atlantech.net/blog/sip-trunking-business-phone-service) says SIP trunks are for customers that already have their own VoIP PBX. The dial plan, the voicemail and the call routing stay in a system you control, and you take on its upkeep. That is the trade in [SIP trunking versus cloud PBX](https://www.answerstack.io/q/what-is-the-difference-between-sip-trunking-and-cloud-pbx).\n\nNeither route is as tidy as its label suggests. A hosted service still leaves equipment at your site. People need phones or apps, and the office network has to carry the calls. Atlantech lists headsets, handsets it supplies and customer-owned devices that work with its voice switch. A trunk, for its part, does not require a PBX in your building. Twilio's requirements name an IP PBX, a SIP call server and a session border controller as examples of what a trunk can connect to. A system of that kind can run on a server somewhere else, provided it can reach the carrier.","introHtml":"Atlantech Online's business phone page puts the two purchases side by side. You can connect to its cloud PBX, or, if you have your own PBX, it can deliver service by SIP, PRI or analog. The cloud PBX is VoIP, and so is the own-PBX route when the delivery is SIP. They differ in where the phone system lives.
\nOn the first route, Atlantech runs the phone system. Its UCaaS page says the hosted phone system is in Atlantech's cloud in place of a PBX in your office. The phone page lists auto attendants, call distribution groups, a receptionist console and voicemail transcription, and says an administrator can add users, devices, phone numbers and call queues. It also describes unlimited nationwide calling, so public calling is part of the package. A buyer who asks for "VoIP" and means "a phone system I don't have to run" is describing this route.
\nOn the second route, you run the PBX and Atlantech connects it. Its April 2023 post on SIP trunks says SIP trunks are for customers that already have their own VoIP PBX. The dial plan, the voicemail and the call routing stay in a system you control, and you take on its upkeep. That is the trade in SIP trunking versus cloud PBX.
\nNeither route is as tidy as its label suggests. A hosted service still leaves equipment at your site. People need phones or apps, and the office network has to carry the calls. Atlantech lists headsets, handsets it supplies and customer-owned devices that work with its voice switch. A trunk, for its part, does not require a PBX in your building. Twilio's requirements name an IP PBX, a SIP call server and a session border controller as examples of what a trunk can connect to. A system of that kind can run on a server somewhere else, provided it can reach the carrier.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":40},{"id":"efb7d675-8e81-4524-949a-48ce0619158c","sectionKey":"office_example","sectionType":"markdown_section","heading":"One office, both ways","introMarkdown":"Suppose a 40-person office runs its own IP PBX with IP desk phones. Each phone connects to the PBX over the office network. When one employee calls another's extension, the PBX connects them itself, and that call is VoIP from end to end. When someone dials a customer's mobile, the PBX sends the call out through its SIP trunk, and the carrier passes it to the mobile network. VoIP is in use on both sides of the PBX. SIP trunking describes only the side that faces the carrier.\n\nNow suppose the same office moves to a hosted service that bundles public calling, as Atlantech's does. The desk phones register with the provider's system over the office's data connection. A call between two extensions is set up by that system, and a call to the customer's mobile leaves from the provider's platform. In this arrangement the office has no PBX of its own and no separate trunk to order, because public calling comes with the service. Its calls are still VoIP.","introHtml":"Suppose a 40-person office runs its own IP PBX with IP desk phones. Each phone connects to the PBX over the office network. When one employee calls another's extension, the PBX connects them itself, and that call is VoIP from end to end. When someone dials a customer's mobile, the PBX sends the call out through its SIP trunk, and the carrier passes it to the mobile network. VoIP is in use on both sides of the PBX. SIP trunking describes only the side that faces the carrier.
\nNow suppose the same office moves to a hosted service that bundles public calling, as Atlantech's does. The desk phones register with the provider's system over the office's data connection. A call between two extensions is set up by that system, and a call to the customer's mobile leaves from the provider's platform. In this arrangement the office has no PBX of its own and no separate trunk to order, because public calling comes with the service. Its calls are still VoIP.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":50},{"id":"34fc91a0-ea08-40d4-b975-50c833bb73e4","sectionKey":"quote_units","sectionType":"markdown_section","heading":"What changes on the quote","introMarkdown":"The unit of the price follows from the service. A trunk is priced by the connection's capacity or use. [SIP.US](https://www.sip.us/pricing/), which publishes its rates, charges $24.95 a month for each [channel](https://www.answerstack.io/q/what-is-a-sip-trunk-channel) on its month-to-month plan, and a channel is one call in progress. Suppose the 40-person office decides to buy eight channels. They come to $199.60 a month before taxes, fees and phone numbers. Headcount does not enter into the sum. Eight is an assumed order here. The right count comes from [sizing the trunk for your busiest hour](https://www.answerstack.io/q/how-many-simultaneous-calls-do-i-need-or-get-on-a-sip-trunk).\n\nA hosted service is priced for the phone system as well as the calling, and Atlantech's is priced by the user. Its UCaaS page advertises a Business Line starting at $12 per user per month and a Premium Line starting at $18. Those are starting rates, and the page prints no contract term, taxes or equipment charges beside them. At the Business Line rate, 40 users would come to $480 a month before any other charges, if every user is on that line and the quote comes in at the starting rate. So the same office counts 40 users on one route and eight calls on the other. Atlantech's phone page describes the pricing for its trunk route as tailored and prints no rate. We compare more per-user rates in [what UCaaS costs per user](https://www.answerstack.io/q/how-much-does-ucaas-cost-per-user).\n\nThe two totals are not comparable as they stand, because the trunk price buys only the connection. The PBX behind it has its own costs, in licenses, hardware or hosting, and in the time someone spends looking after it. Put those beside the trunk before you set it against a hosted quote.","introHtml":"The unit of the price follows from the service. A trunk is priced by the connection's capacity or use. SIP.US, which publishes its rates, charges $24.95 a month for each channel on its month-to-month plan, and a channel is one call in progress. Suppose the 40-person office decides to buy eight channels. They come to $199.60 a month before taxes, fees and phone numbers. Headcount does not enter into the sum. Eight is an assumed order here. The right count comes from sizing the trunk for your busiest hour.
\nA hosted service is priced for the phone system as well as the calling, and Atlantech's is priced by the user. Its UCaaS page advertises a Business Line starting at $12 per user per month and a Premium Line starting at $18. Those are starting rates, and the page prints no contract term, taxes or equipment charges beside them. At the Business Line rate, 40 users would come to $480 a month before any other charges, if every user is on that line and the quote comes in at the starting rate. So the same office counts 40 users on one route and eight calls on the other. Atlantech's phone page describes the pricing for its trunk route as tailored and prints no rate. We compare more per-user rates in what UCaaS costs per user.
\nThe two totals are not comparable as they stand, because the trunk price buys only the connection. The PBX behind it has its own costs, in licenses, hardware or hosting, and in the time someone spends looking after it. Put those beside the trunk before you set it against a hosted quote.
\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":"This document describes Session Initiation Protocol (SIP), 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.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"SIP is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants, including internet telephone calls, multimedia distribution, and multimedia conferences; SIP is a standard, not a product or a service.","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":"RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"RTP provides end-to-end transport for real-time data such as audio, separate from the signaling protocol, which is why call media and call setup are distinct layers in both a SIP trunk and a hosted VoIP service.","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"ITU-T Recommendation H.323: Packet-based multimedia communications systems","url":"https://www.itu.int/rec/T-REC-H.323","excerpt":"H.323 (03/22) Packet-based multimedia communications systems, In force","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"H.323 is an in-force ITU-T standard (03/22 version listed as in force) for packet-based multimedia communications systems, establishing that voice over IP is not tied to SIP.","domain":"itu.int","publisherName":"ITU-T"},{"title":"RFC 3435: Media Gateway Control Protocol (MGCP) Version 1.0","url":"https://datatracker.ietf.org/doc/html/rfc3435","excerpt":"The decomposed multimedia gateway consists of a Call Agent, which contains the call control \"intelligence\", and a media gateway which contains the media functions, e.g., conversion from TDM voice to Voice over IP.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"MGCP is a second non-SIP control protocol used in IP voice, and it splits a gateway into a Call Agent holding the call control intelligence and a media gateway converting TDM voice to voice over IP.","domain":"datatracker.ietf.org","publisherName":"Internet Engineering Task Force (IETF)"},{"title":"47 CFR 9.3 Definitions","url":"https://www.ecfr.gov/current/title-47/section-9.3","excerpt":"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.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"Current text read September 15, 2026 through the eCFR renderer API: the four-part definition of interconnected VoIP service, and the definition of a multi-line telephone system covering network and premises based systems such as Centrex and VoIP as well as PBX, Hybrid, and Key Telephone Systems.","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":"(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.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"Current text read September 15, 2026 through the eCFR renderer API: paragraph (a) reads [Reserved]; paragraph (b) places the E911 duties on providers of interconnected VoIP services, including transmission of all 911 calls with ANI and location information and the plain-language customer notificatio","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":"The Direct Final Rule would repeal approximately 21 rule provisions and rule parts ... that plainly no longer serve the public interest because they have sunset by operation of law.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"FCC direct final rule, citation 91 FR 1400, published January 14, 2026 and effective March 16, 2026, repealing approximately 21 rule provisions that had sunset by operation of law or become obsolete; this is the action that removed and reserved 47 CFR 9.11(a).","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":"(b) Obligation of installers, managers, or operators. A person engaged in the business of installing, managing, or operating multi-line telephone systems may not install, manage, or operate for use in the United States such a system, unless such system is configured such that 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.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-09-15T00:00:00","supportsText":"Current text read September 15, 2026 through the eCFR renderer API: paragraph (a) binds manufacturers, importers, sellers, and lessors of multi-line telephone systems; paragraph (b) binds persons installing, managing, or operating an MLTS to configure direct 911 dialing without a trunk-access prefix","domain":"ecfr.gov","publisherName":"Electronic Code of Federal Regulations (National Archives / GPO)"},{"title":"SIP Trunking Pricing","url":"https://www.sip.us/pricing/","excerpt":"SIP channels Month to Month $24.95 per channel per/month; Yearly Plans $19.99 per channel per month; No set-up fee. Includes local and long distance calls to the US48 states and Canada (excluding NWT) at no additional charge.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-09-15T00:00:00","supportsText":"Published list prices read September 15, 2026: SIP channels $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 calls to the US 48 states and Canada and excluding local, state, and federal taxes and fees; US48","domain":"sip.us","publisherName":"SIP.US"},{"title":"Elastic SIP Trunking Pricing","url":"https://www.twilio.com/en-us/sip-trunking/pricing","excerpt":"United States - 48 States (Zone 1) $0.0100 / min ... Origination Phone number type Price Local $0.0034 / min","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-09-15T00:00:00","supportsText":"Published list prices read September 15, 2026: termination to United States 48 States (Zone 1) $0.0100 per minute, local origination $0.0034 per minute, toll-free termination $0.0011 per minute; billing is per minute with no channel quantity to purchase.","domain":"twilio.com","publisherName":"Twilio"},{"title":"Elastic SIP Trunking","url":"https://www.twilio.com/docs/sip-trunking","excerpt":"A SIP enabled network element (e.g. Session Border Controller, SIP Call Server, IP-PBX, SIP-PRI IAD, etc.) with access to the internet. ... SIP Trunk Peak Bandwidth = Max Simultaneous Calls x 100 kbps","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-09-15T00:00:00","supportsText":"Product documentation read September 15, 2026: using a SIP trunk requires a SIP-enabled network element such as a session border controller, SIP call server, or IP-PBX with internet access, plus peak bandwidth calculated as maximum simultaneous calls times 100 kbps, which covers the G.711 codec plus","domain":"twilio.com","publisherName":"Twilio"},{"title":"Flowroute pricing details","url":"https://www.flowroute.com/pricing/","excerpt":"Domestic Inbound rate $0.005/minute ... Outbound rate (US lower 48 & Canada) $0.00833/minute ... With no limit on simultaneous calls, you can instantly scale to occasional or seasonal bursts in traffic.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-09-15T00:00:00","supportsText":"Published list prices read September 15, 2026: outbound termination to the US lower 48 and Canada starting at $0.00833 per minute and domestic inbound starting at $0.005 per minute, with no limit on simultaneous calls and no channel purchase.","domain":"flowroute.com","publisherName":"Flowroute"},{"title":"Nextiva plans and pricing","url":"https://www.nextiva.com/pricing","excerpt":"Core Connect your team & customers $15 /user/mo $23 Save 35% ... Core $15/user/mo ... Engage $25/user/mo ... Scale $75/user/mo","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-09-15T00:00:00","supportsText":"Published seat prices read September 15, 2026 on a page with a Monthly and Annually toggle marked Save up to 50%: Core $15 per user per month on the discounted side and $23 on the other, Engage $25 or $50, Scale $75, with call routing and team features listed per user; documents that a hosted VoIP s","domain":"nextiva.com","publisherName":"Nextiva"},{"title":"Google Voice for Google Workspace plans and pricing","url":"https://workspace.google.com/products/voice/","excerpt":"Starter Add professional calling features to your team's Workspace plan $10 / user per month ... For up to 10 users ... Standard Scale your growing team with automated call handling and ring groups $20 / user per month","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-09-15T00:00:00","supportsText":"Published team plan prices read September 15, 2026: Starter $10 per user per month for up to 10 users, Standard $20 per user per month with unlimited users and auto attendants and ring groups, Premier $30 per user per month; documents that call routing features are operated by the provider and sold ","domain":"workspace.google.com","publisherName":"Google (vendor pricing page)"},{"title":"SIP trunking vs. VoIP: What's the difference?","url":"https://www.techtarget.com/searchunifiedcommunications/answer/SIP-trunking-vs-VoIP-Whats-the-difference","excerpt":"When someone makes a VoIP call, SIP will be one of the protocols used to enable calls over the data network. ... But SIP supports other media modes as well, such as video and instant messaging.","quoteText":null,"sourceRole":"SUPPORTING","verifiedAt":"2026-09-15T00:00:00","supportsText":"Trade publication explainer used only for the framing that SIP and VoIP are complementary rather than competing, that SIP is one of the protocols used to enable a VoIP call, and that SIP sessions can carry video and instant messaging as well as voice.","domain":"techtarget.com","publisherName":"TechTarget (Jon Arnold)"}],"revisions":[{"revisedAt":"2026-07-18T15:01:30.724+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":"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"},{"id":"fad66bd0-23bc-421f-944d-f514e3d12488","slug":"what-is-sip-trunking","question":"What is SIP trunking?","publishedAt":"2026-07-18T14:40:06.195","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":"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.","url":"/q/what-is-sip-trunking"}],"contributorStats":{"verifiedAnswers":149,"openDisputes":0},"schemaJson":{"@context":"https://schema.org","@type":"Question","name":"What is the difference between SIP trunking and VoIP?","text":"What is the difference between SIP trunking and VoIP?","url":"https://www.answerstack.io/q/what-is-the-difference-between-sip-trunking-and-voip","answerCount":1,"datePublished":"2026-07-18T15:01:30.724","about":[{"@type":"Thing","name":"SIP Trunking"},{"@type":"Thing","name":"Telecommunications"}],"acceptedAnswer":{"@type":"Answer","text":"VoIP is a technology: voice carried over data networks using internet protocols. SIP trunking is one service built on it, in which a carrier connects a business phone system to the public telephone network over IP. A SIP trunk is VoIP, so the two are not alternatives. The question turns into a choice only when \"VoIP\" is shorthand for a hosted phone service, and then the difference is who runs the phone system.","url":"https://www.answerstack.io/q/what-is-the-difference-between-sip-trunking-and-voip","upvoteCount":0,"datePublished":"2026-07-18T15:01:30.724","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":"ITU-T Recommendation H.323: Packet-based multimedia communications systems","url":"https://www.itu.int/rec/T-REC-H.323"},{"@type":"CreativeWork","name":"RFC 3435: Media Gateway Control Protocol (MGCP) Version 1.0","url":"https://datatracker.ietf.org/doc/html/rfc3435"},{"@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":"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":"Nextiva plans and pricing","url":"https://www.nextiva.com/pricing"},{"@type":"CreativeWork","name":"Google Voice for Google Workspace plans and pricing","url":"https://workspace.google.com/products/voice/"},{"@type":"CreativeWork","name":"SIP trunking vs. VoIP: What's the difference?","url":"https://www.techtarget.com/searchunifiedcommunications/answer/SIP-trunking-vs-VoIP-Whats-the-difference"}]}}}