To use SIP trunking you need a SIP-capable phone system, endpoints for the people placing calls, an internet connection sized for voice, and a session border controller at the network edge, plus a gateway only if you are keeping analog or legacy equipment.[1][9][14] The phone system is usually an IP-PBX, either an appliance or software you run on-site or a hosted PBX the provider operates for you, and it speaks the Session Initiation Protocol to set up and tear down each call.[1][2] The people on the calls use IP desk phones or softphones running on a computer or mobile device, and hardware phones commonly draw power over the same Ethernet cable that carries their traffic.[4] A session border controller sits between your system and the provider to secure the SIP session, reconcile mismatched settings, and act as a back-to-back user agent on both call legs.[7][8] If you still have analog phones, fax machines, or a PRI-era PBX, an analog telephone adapter or VoIP gateway converts that traffic to SIP so it can ride the trunk.[5][6]
\n","summary":"SIP trunking runs on ordinary business equipment rather than dedicated phone-company circuits: a SIP-capable phone system (an on-site IP-PBX or a hosted PBX), IP desk phones or softphones, a session border controller at the edge, and an internet link sized for your busy-hour calls. Analog phones or a legacy PRI system connect through an adapter or gateway. A hosted setup shifts most of this into the provider's data center and leaves little more than phones and a solid network in your building.","publishedAt":"2026-07-18T15:16:57.319","verifiedAt":"2026-07-18T00:00:00","editorialStatus":"APPROVED","lastReviewedAt":"2026-07-18T00:00:00","nextReviewDueAt":"2026-10-18T00:00:00","templateVersion":"v2","aliases":["What hardware do I need for SIP trunking?","SIP trunking equipment list","What do I need to set up SIP trunking?","SIP trunking hardware requirements","Do I need an SBC for SIP trunking?","Do I need a new PBX for SIP trunking?","Can I use my old PBX with SIP trunking?","What equipment does a SIP trunk require?","SIP trunking requirements for business","Do I need special equipment for SIP trunking?","What phones work with SIP trunking?","Does SIP trunking need a gateway?"],"confidenceScore":90,"confidenceLabel":"High","canonicalUrl":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"contributorOrganizationProfile":{"entityId":"ec39deab-44fe-48d8-9029-fefe993ab85a","legalName":null,"description":null,"websiteUrl":null,"imageUrl":null,"slogan":null,"subtitle":null,"facts":[],"coiNote":null,"foundingDate":null,"numberOfEmployeesText":null,"contactPoint":null,"address":null,"headquartersText":null,"organizationType":null},"contributorPerson":{"slug":"answerstack-editorial-team","displayName":"AnswerStack Editorial Team"},"sections":[{"id":"7d1da0b3-621f-4b5b-9cc3-d31dbf73b290","sectionKey":"what_equipment_is_needed","sectionType":"markdown_section","heading":"What equipment does SIP trunking require?","introMarkdown":"SIP trunking runs on standard IT equipment instead of the dedicated telephone circuits an older phone system needed, so the core list is a SIP-capable phone system, endpoints for your people, a session border controller at the network edge, and an internet connection sized for the calls you make.[1][9] A gateway or analog adapter joins that list only when you keep analog phones or a legacy PBX that cannot speak SIP on its own.[5][6] None of it includes a physical trunk line, because the trunk is the service your provider delivers, not a box you install.[1]\n\n### The trunk is a service, not a box\n\nThe SIP trunk itself is not something you plug in. Your provider delivers it as a virtual connection over your internet link, using the Session Initiation Protocol to start, change, and end each call while the audio travels separately as Real-time Transport Protocol packets.[2][3] Because the trunk arrives as software rather than a physical line, the equipment question is really about what sits inside your building to originate calls and hand them to the provider cleanly.[1]\n\n### What decides how much you buy\n\nHow much equipment you actually own depends on one early decision: whether you run your own phone system or use a hosted one. An on-premises setup means you buy and maintain the IP-PBX, the session border controller, and the network gear that carries voice.[9] A hosted or cloud PBX moves most of that into the provider's data center, so the kit in your office can be as little as IP phones and a network that handles voice well.[9][14] Every deployment still comes down to three parts that have to talk to each other: a provider that carries your calls, a phone system that speaks SIP, and something at the border that secures and connects the two.[1][14] The sections below take each piece in turn.","introHtml":"SIP trunking runs on standard IT equipment instead of the dedicated telephone circuits an older phone system needed, so the core list is a SIP-capable phone system, endpoints for your people, a session border controller at the network edge, and an internet connection sized for the calls you make.[1][9] A gateway or analog adapter joins that list only when you keep analog phones or a legacy PBX that cannot speak SIP on its own.[5][6] None of it includes a physical trunk line, because the trunk is the service your provider delivers, not a box you install.[1]
\nThe SIP trunk itself is not something you plug in. Your provider delivers it as a virtual connection over your internet link, using the Session Initiation Protocol to start, change, and end each call while the audio travels separately as Real-time Transport Protocol packets.[2][3] Because the trunk arrives as software rather than a physical line, the equipment question is really about what sits inside your building to originate calls and hand them to the provider cleanly.[1]
\nHow much equipment you actually own depends on one early decision: whether you run your own phone system or use a hosted one. An on-premises setup means you buy and maintain the IP-PBX, the session border controller, and the network gear that carries voice.[9] A hosted or cloud PBX moves most of that into the provider's data center, so the kit in your office can be as little as IP phones and a network that handles voice well.[9][14] Every deployment still comes down to three parts that have to talk to each other: a provider that carries your calls, a phone system that speaks SIP, and something at the border that secures and connects the two.[1][14] The sections below take each piece in turn.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":0},{"id":"6ef610ad-8bca-4710-b2b8-2d0a311b1d2d","sectionKey":"equipment_checklist","sectionType":"table_section","heading":"The SIP trunking equipment checklist","introMarkdown":"Five pieces of equipment cover almost every SIP trunk deployment, and how many you own depends on whether the phone system is yours or the provider's.[9] The table lists them for a quick scan, and each one gets a section below explaining what it does and how to size or choose it.","introHtml":"Five pieces of equipment cover almost every SIP trunk deployment, and how many you own depends on whether the phone system is yours or the provider's.[9] The table lists them for a quick scan, and each one gets a section below explaining what it does and how to size or choose it.
\n","outroMarkdown":"A hosted phone service collapses much of this list, since the provider runs the PBX and often the SBC, leaving you responsible mainly for phones and a network that keeps voice clean.[9] A fully on-premises build reverses that, because every row becomes yours to buy, secure, and maintain.[7][9]","outroHtml":"A hosted phone service collapses much of this list, since the provider runs the PBX and often the SBC, leaving you responsible mainly for phones and a network that keeps voice clean.[9] A fully on-premises build reverses that, because every row becomes yours to buy, secure, and maintain.[7][9]
\n","contentJson":{"rows":[{"cells":["SIP-capable phone system (IP-PBX or hosted PBX)","Routes calls and hands them to the trunk in SIP","Always; on-site if you run it, or in the provider's cloud [1][9]"]},{"cells":["IP phones and softphones","The endpoints people use to place and take calls","Always, though a softphone-only office needs no desk hardware [4]"]},{"cells":["Session border controller (SBC)","Secures and normalizes SIP at the border of your network","Most on-premises builds; usually provider-side when hosted [7][8]"]},{"cells":["VoIP gateway or analog adapter (ATA)","Converts analog or PRI-era traffic into SIP","Only when you keep analog phones or a legacy PBX [5][6]"]},{"cells":["Internet connection and edge network","Carries and prioritizes voice: router, firewall, PoE switch","Always; size it to busy-hour concurrent calls [9][11]"]}],"columns":["Equipment","What it does","When you need it"]},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":1},{"id":"a86d45d8-1aa2-4886-8147-77d62ce13b2e","sectionKey":"phone_system","sectionType":"markdown_section","heading":"A SIP-capable phone system (IP-PBX or hosted PBX)","introMarkdown":"The phone system is the one piece of equipment no SIP trunk can do without, because it originates and receives calls in SIP and hands them to the provider.[1] For most businesses this is an IP-PBX, which comes in three forms: a hardware appliance in a rack, software you run on your own server or a virtual machine, and a hosted PBX the provider operates for you.[9] The first two keep the system in your building, where you control it and carry the maintenance, while a hosted option removes most of that burden because the provider runs the platform and you connect phones to it.\n\nThe detail that trips people up is SIP compatibility. A system built for analog or PRI lines can still connect to a trunk, but it needs a gateway to translate its signaling first, and even a native IP-PBX can format SIP headers or negotiate codecs slightly differently from a given provider.[1][6] Those small mismatches are a common cause of one-way audio or calls that drop after a set time. Checking your PBX vendor's list of certified providers before you commit, or planning for a session border controller to reconcile the differences, saves a troubleshooting cycle after cutover.","introHtml":"The phone system is the one piece of equipment no SIP trunk can do without, because it originates and receives calls in SIP and hands them to the provider.[1] For most businesses this is an IP-PBX, which comes in three forms: a hardware appliance in a rack, software you run on your own server or a virtual machine, and a hosted PBX the provider operates for you.[9] The first two keep the system in your building, where you control it and carry the maintenance, while a hosted option removes most of that burden because the provider runs the platform and you connect phones to it.
\nThe detail that trips people up is SIP compatibility. A system built for analog or PRI lines can still connect to a trunk, but it needs a gateway to translate its signaling first, and even a native IP-PBX can format SIP headers or negotiate codecs slightly differently from a given provider.[1][6] Those small mismatches are a common cause of one-way audio or calls that drop after a set time. Checking your PBX vendor's list of certified providers before you commit, or planning for a session border controller to reconcile the differences, saves a troubleshooting cycle after cutover.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":2},{"id":"abe9d233-63b8-4d80-8244-b308f8d41a46","sectionKey":"endpoints","sectionType":"markdown_section","heading":"IP phones and softphones for the people making calls","introMarkdown":"Endpoints are the devices your team actually speaks into, and with SIP trunking they are IP based rather than analog.[4] A hardware IP phone looks like a normal desk phone but connects to your data network over Ethernet and registers to your phone system, while a softphone is an application that turns a computer or mobile device into the same thing.[4] Many organizations mix the two, giving desk phones to people who take calls all day and softphones to staff who are mobile or hybrid.\n\nA practical point on hardware phones is power. Many draw electricity over the same Ethernet cable that carries their traffic through Power over Ethernet, so a switch that supplies PoE removes the need for a separate power adapter at every desk.[4][9] Endpoints are also where the codec choice shows up, since the phone and the provider have to agree on one for the call. G.711 delivers full narrowband quality at the cost of more bandwidth per call, while a compressed codec such as G.729 trades a little fidelity for lighter network use.[11][12] For a pure softphone deployment you may need no telephone hardware at all, only headsets and the network underneath.","introHtml":"Endpoints are the devices your team actually speaks into, and with SIP trunking they are IP based rather than analog.[4] A hardware IP phone looks like a normal desk phone but connects to your data network over Ethernet and registers to your phone system, while a softphone is an application that turns a computer or mobile device into the same thing.[4] Many organizations mix the two, giving desk phones to people who take calls all day and softphones to staff who are mobile or hybrid.
\nA practical point on hardware phones is power. Many draw electricity over the same Ethernet cable that carries their traffic through Power over Ethernet, so a switch that supplies PoE removes the need for a separate power adapter at every desk.[4][9] Endpoints are also where the codec choice shows up, since the phone and the provider have to agree on one for the call. G.711 delivers full narrowband quality at the cost of more bandwidth per call, while a compressed codec such as G.729 trades a little fidelity for lighter network use.[11][12] For a pure softphone deployment you may need no telephone hardware at all, only headsets and the network underneath.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":3},{"id":"2e3c7482-382d-4c53-b4d5-b890d2da4691","sectionKey":"session_border_controller","sectionType":"markdown_section","heading":"A session border controller at the network edge","introMarkdown":"A session border controller sits between your phone system and the provider, managing every SIP call that crosses the boundary of your network.[8] Most SBCs work as a back-to-back user agent, which means the device fully ends the provider's call leg and starts a fresh one toward your PBX, giving it authority over both sides of the call.[8] From that position it does jobs a plain firewall cannot: the IETF's requirements document groups them into perimeter defense such as access control and topology hiding, functions the endpoints lack such as NAT traversal and protocol repair, and traffic management such as media monitoring and quality of service.[7]\n\nWhether you own one depends on how you deploy. An on-premises phone system almost always needs an SBC, either as a hardware appliance or as software on a virtual machine, because it is what protects an internet-facing phone system and smooths the header and codec differences between your PBX and the carrier.[7][9] With a hosted phone service the provider usually runs the SBC on its side, so a small office may never buy one. Sites with strict security requirements often keep their own regardless, so they set the policies at their own border.","introHtml":"A session border controller sits between your phone system and the provider, managing every SIP call that crosses the boundary of your network.[8] Most SBCs work as a back-to-back user agent, which means the device fully ends the provider's call leg and starts a fresh one toward your PBX, giving it authority over both sides of the call.[8] From that position it does jobs a plain firewall cannot: the IETF's requirements document groups them into perimeter defense such as access control and topology hiding, functions the endpoints lack such as NAT traversal and protocol repair, and traffic management such as media monitoring and quality of service.[7]
\nWhether you own one depends on how you deploy. An on-premises phone system almost always needs an SBC, either as a hardware appliance or as software on a virtual machine, because it is what protects an internet-facing phone system and smooths the header and codec differences between your PBX and the carrier.[7][9] With a hosted phone service the provider usually runs the SBC on its side, so a small office may never buy one. Sites with strict security requirements often keep their own regardless, so they set the policies at their own border.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":4},{"id":"d09bf981-ee94-4044-9362-c80a9a6d1dbb","sectionKey":"gateway_or_ata","sectionType":"markdown_section","heading":"A gateway or adapter if you keep analog or legacy gear","introMarkdown":"A gateway or analog adapter is optional equipment that only matters when you are holding onto phones or a PBX that cannot speak SIP.[5][6] An analog telephone adapter, or ATA, connects traditional analog telephones and fax machines to an IP voice network through one or more foreign exchange station ports, so a fax machine or a lobby phone can keep working on a SIP trunk.[5] A VoIP gateway does the same job at larger scale, converting time-division voice from a PRI or T1 into the Real-time Transport Protocol packets an IP network carries.[6]\n\nThis piece is what lets a business migrate without replacing everything at once. A company with an older PBX built for PRI lines can put a gateway in front of it, connect the SIP trunk, and keep its existing handsets running while it plans a full move to IP.[6] The trade-off is that routing calls through a gateway adds a conversion step and can limit access to newer SIP features, so a gateway suits a transition better than a permanent design. Where every phone is already IP and every user can move to a softphone, you skip this equipment entirely.[4]","introHtml":"A gateway or analog adapter is optional equipment that only matters when you are holding onto phones or a PBX that cannot speak SIP.[5][6] An analog telephone adapter, or ATA, connects traditional analog telephones and fax machines to an IP voice network through one or more foreign exchange station ports, so a fax machine or a lobby phone can keep working on a SIP trunk.[5] A VoIP gateway does the same job at larger scale, converting time-division voice from a PRI or T1 into the Real-time Transport Protocol packets an IP network carries.[6]
\nThis piece is what lets a business migrate without replacing everything at once. A company with an older PBX built for PRI lines can put a gateway in front of it, connect the SIP trunk, and keep its existing handsets running while it plans a full move to IP.[6] The trade-off is that routing calls through a gateway adds a conversion step and can limit access to newer SIP features, so a gateway suits a transition better than a permanent design. Where every phone is already IP and every user can move to a softphone, you skip this equipment entirely.[4]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":5},{"id":"5ead4199-1b97-4aa0-8a1c-aee137a2e823","sectionKey":"internet_and_network","sectionType":"markdown_section","heading":"An internet connection and edge network sized for voice","introMarkdown":"Voice on a SIP trunk shares your internet connection with everything else, so the connection and the gear around it are as much a part of the equipment list as any phone.[9] Each concurrent call needs roughly 85 to 87 kilobits per second in each direction with the common G.711 codec once IP, UDP, and RTP headers are counted, dropping to about 31 kilobits with the more compressed G.729.[11] Multiply your busy-hour concurrent calls by that figure, add headroom for the data traffic on the same link, and you have a floor for the connection to buy.[9]\n\nThe network gear matters as much as raw bandwidth. A business-grade router with quality-of-service settings lets you prioritize voice packets so a large download does not turn calls choppy, and a managed switch, ideally one that supplies Power over Ethernet, connects and powers the phones.[9] Unlike a PRI, where adding a twenty-fourth simultaneous call meant ordering another physical circuit, a SIP trunk scales in software over the connection you already have, so the ceiling is your bandwidth rather than a hardware limit.[10] A provider that reaches the public phone network also has to record the physical location of your equipment for 911, and that registered location has to stay current as devices move.[13]","introHtml":"Voice on a SIP trunk shares your internet connection with everything else, so the connection and the gear around it are as much a part of the equipment list as any phone.[9] Each concurrent call needs roughly 85 to 87 kilobits per second in each direction with the common G.711 codec once IP, UDP, and RTP headers are counted, dropping to about 31 kilobits with the more compressed G.729.[11] Multiply your busy-hour concurrent calls by that figure, add headroom for the data traffic on the same link, and you have a floor for the connection to buy.[9]
\nThe network gear matters as much as raw bandwidth. A business-grade router with quality-of-service settings lets you prioritize voice packets so a large download does not turn calls choppy, and a managed switch, ideally one that supplies Power over Ethernet, connects and powers the phones.[9] Unlike a PRI, where adding a twenty-fourth simultaneous call meant ordering another physical circuit, a SIP trunk scales in software over the connection you already have, so the ceiling is your bandwidth rather than a hardware limit.[10] A provider that reaches the public phone network also has to record the physical location of your equipment for 911, and that registered location has to stay current as devices move.[13]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":6},{"id":"3cb33146-2982-468b-bb64-c2697befa233","sectionKey":"contributor_perspective","sectionType":"markdown_section","heading":"How this answer was researched","introMarkdown":"This entry separates what the standards define from what the market sells. The behavior of SIP and RTP comes from the IETF documents that specify them, and the description of what a session border controller does is taken from the IETF's requirements for SBC deployments rather than any vendor's brochure.[2][3][7] Equipment definitions such as the analog adapter, the media gateway, and the IP phone are drawn from independent reference sources, and the bandwidth and sizing figures are attributed to industry sources because they vary by provider and codec rather than being fixed facts.[4][5][6][11] The 911 location obligation comes from the Code of Federal Regulations as written.[13] Product names, prices, and feature sets in telecom change quickly, so the specifics here reflect what the cited sources said on the verification date. Practitioners who deploy, secure, or support SIP trunks are welcome to suggest corrections, which are checked against primary sources before any update.","introHtml":"This entry separates what the standards define from what the market sells. The behavior of SIP and RTP comes from the IETF documents that specify them, and the description of what a session border controller does is taken from the IETF's requirements for SBC deployments rather than any vendor's brochure.[2][3][7] Equipment definitions such as the analog adapter, the media gateway, and the IP phone are drawn from independent reference sources, and the bandwidth and sizing figures are attributed to industry sources because they vary by provider and codec rather than being fixed facts.[4][5][6][11] The 911 location obligation comes from the Code of Federal Regulations as written.[13] Product names, prices, and feature sets in telecom change quickly, so the specifics here reflect what the cited sources said on the verification date. Practitioners who deploy, secure, or support SIP trunks are welcome to suggest corrections, which are checked against primary sources before any update.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":"This answer was written and reviewed by the AnswerStack Editorial Team, which has no commercial stake in the products, companies, or methods discussed. Every claim is cited inline and verified on the dates shown.","noteHtml":"This answer was written and reviewed by the AnswerStack Editorial Team, which has no commercial stake in the products, companies, or methods discussed. Every claim is cited inline and verified on the dates shown.
\n","sortOrder":7},{"id":"8e557996-2bae-456d-96df-6d5d5d38f8b5","sectionKey":"hosted_vs_onprem","sectionType":"markdown_section","heading":"How the equipment list changes: hosted vs on-premises","introMarkdown":"The single biggest factor in what you buy is whether the phone system lives in your building or in the provider's data center, and each approach carries trade-offs worth weighing before you order hardware.[9]\n\n### On-premises deployments\n\nRunning your own IP-PBX and SBC gives you direct control over configuration, security policy, and how calls route, which suits organizations with complex needs or strict data requirements.[7][9] The cost is that you buy, secure, patch, and eventually replace that equipment, and you carry responsibility for its uptime.\n\n### Hosted and cloud deployments\n\nA hosted or cloud PBX shifts the PBX and usually the SBC to the provider, so your equipment shrinks to phones and a solid network.[9] That lowers the upfront hardware bill and the maintenance load, though you depend on the provider's platform and configuration choices, and deep customization can be harder to arrange.\n\n### What to watch either way\n\nTwo things bite regardless of model. Voice quality depends on the network, so a link without quality-of-service settings or with too little headroom produces choppy calls under load no matter whose PBX runs them.[9][11] And because a trunk connects to the public phone network, 911 obligations follow you: the provider records the location of your equipment, and that record has to stay accurate as people and devices move, or an emergency call can route to the wrong place.[13]","introHtml":"The single biggest factor in what you buy is whether the phone system lives in your building or in the provider's data center, and each approach carries trade-offs worth weighing before you order hardware.[9]
\nRunning your own IP-PBX and SBC gives you direct control over configuration, security policy, and how calls route, which suits organizations with complex needs or strict data requirements.[7][9] The cost is that you buy, secure, patch, and eventually replace that equipment, and you carry responsibility for its uptime.
\nA hosted or cloud PBX shifts the PBX and usually the SBC to the provider, so your equipment shrinks to phones and a solid network.[9] That lowers the upfront hardware bill and the maintenance load, though you depend on the provider's platform and configuration choices, and deep customization can be harder to arrange.
\nTwo things bite regardless of model. Voice quality depends on the network, so a link without quality-of-service settings or with too little headroom produces choppy calls under load no matter whose PBX runs them.[9][11] And because a trunk connects to the public phone network, 911 obligations follow you: the provider records the location of your equipment, and that record has to stay accurate as people and devices move, or an emergency call can route to the wrong place.[13]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":8},{"id":"e894dd4f-7ec7-4221-bdbc-3ddc3869dd05","sectionKey":"what_it_is_not","sectionType":"markdown_section","heading":"What SIP trunking does not require","introMarkdown":"SIP trunking removes several things older phone setups needed, and knowing what you can skip prevents over-buying.[1]\n\n### It does not need a dedicated phone circuit\n\nA SIP trunk arrives over your existing internet connection, so there is no PRI, T1, or copper pair to order from the phone company.[1][10] Capacity that once meant installing another physical circuit is now a software change with your provider.[10]\n\n### It does not always need an SBC you own\n\nAn on-premises system almost always needs a session border controller, but a hosted phone service usually includes that function on the provider's side, so a small office may never rack one of its own.[7][8]\n\n### It does not require replacing your analog phones\n\nAnalog handsets, fax machines, and alarm lines can stay in place behind an adapter or gateway that converts their signaling to SIP.[5][6] Replacing them is a choice about features and tidiness, not a requirement of the trunk.\n\n### It is not the same as a hosted phone system\n\nA SIP trunk connects a phone system you already run to the outside world, while a hosted PBX is the phone system itself delivered as a service.[1] The two are sometimes used together, and confusing them leads to buying equipment you do not need or skipping equipment you do.","introHtml":"SIP trunking removes several things older phone setups needed, and knowing what you can skip prevents over-buying.[1]
\nA SIP trunk arrives over your existing internet connection, so there is no PRI, T1, or copper pair to order from the phone company.[1][10] Capacity that once meant installing another physical circuit is now a software change with your provider.[10]
\nAn on-premises system almost always needs a session border controller, but a hosted phone service usually includes that function on the provider's side, so a small office may never rack one of its own.[7][8]
\nAnalog handsets, fax machines, and alarm lines can stay in place behind an adapter or gateway that converts their signaling to SIP.[5][6] Replacing them is a choice about features and tidiness, not a requirement of the trunk.
\nA SIP trunk connects a phone system you already run to the outside world, while a hosted PBX is the phone system itself delivered as a service.[1] The two are sometimes used together, and confusing them leads to buying equipment you do not need or skipping equipment you do.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":9}],"citations":[{"title":"SIP trunking","url":"https://en.wikipedia.org/wiki/SIP_trunking","excerpt":"SIP trunking is a voice over Internet Protocol (VoIP) technology and streaming media service based on the Session Initiation Protocol (SIP) by which Internet telephony service providers (ITSPs) deliver telephone services and unified communications to customers equipped with SIP-based private branch exchange (IP-PBX) and unified communications facilities.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"Definition of SIP trunking as a VoIP service based on SIP by which ITSPs deliver telephone service to customers equipped with SIP-based IP-PBX; the trunk as a service rather than a physical line; IP-PBX as the required customer phone system","domain":"en.wikipedia.org","publisherName":"Wikipedia"},{"title":"RFC 3261: SIP: Session Initiation Protocol","url":"https://datatracker.ietf.org/doc/html/rfc3261","excerpt":"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":"PRIMARY","verifiedAt":"2026-07-18T00:00:00","supportsText":"SIP as the application-layer signaling protocol the phone system uses to create, modify, and terminate calls; user agents as SIP endpoints","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":"PRIMARY","verifiedAt":"2026-07-18T00:00:00","supportsText":"RTP carries the real-time voice media of a SIP call separately from the SIP signaling","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"VoIP phone","url":"https://en.wikipedia.org/wiki/VoIP_phone","excerpt":"A VoIP phone or IP phone uses voice over IP technologies for placing and transmitting telephone calls over an IP network, such as the Internet. VoIP phones can be simple software-based softphones or purpose-built hardware devices; some VoIP phones receive electricity from Power over Ethernet.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"IP phones and softphones as the endpoints for SIP calls; hardware phones connect over Ethernet and may receive power from Power over Ethernet; softphones run on a computer or mobile device","domain":"en.wikipedia.org","publisherName":"Wikipedia"},{"title":"Analog telephone adapter","url":"https://en.wikipedia.org/wiki/Analog_telephone_adapter","excerpt":"a device for connecting traditional analog telephones, fax machines, and similar customer-premises devices to a digital telephone system or a voice over IP telephone network... one or more foreign exchange station (FXS) telephone ports.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"An analog telephone adapter connects traditional analog telephones and fax machines to an IP voice network through foreign exchange station (FXS) ports","domain":"en.wikipedia.org","publisherName":"Wikipedia"},{"title":"Media gateway","url":"https://en.wikipedia.org/wiki/Media_gateway","excerpt":"VoIP media gateways perform the conversion between Time-division multiplexing (TDM) voice to a media streaming protocol, such as the Real-time Transport Protocol, (RTP).","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"A VoIP media gateway converts time-division (TDM) voice from a PRI or T1 into RTP media packets, letting a legacy PBX connect to a SIP trunk","domain":"en.wikipedia.org","publisherName":"Wikipedia"},{"title":"RFC 5853: Requirements from Session Initiation Protocol (SIP) Session Border Control (SBC) Deployments","url":"https://datatracker.ietf.org/doc/html/rfc5853","excerpt":"a) perimeter defense (access control, topology hiding, and denial-of-service prevention and detection); b) functionality not available in the endpoints (NAT traversal, protocol interworking or repair); and c) traffic management (media monitoring and Quality of Service (QoS)).","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-18T00:00:00","supportsText":"SBC functions grouped as perimeter defense (access control, topology hiding, denial-of-service prevention), functionality not in endpoints (NAT traversal, protocol interworking or repair), and traffic management (media monitoring and QoS)","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"Session border controller","url":"https://en.wikipedia.org/wiki/Session_border_controller","excerpt":"Most SBCs are implemented as back-to-back user agent. A B2BUA is a proxy-like server that splits a SIP transaction in two call legs: on the side facing the user agent client (UAC), it acts as server, on the side facing user agent server (UAS) it acts as a client.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"SBC as a network element protecting SIP-based VoIP networks; most implemented as a back-to-back user agent that splits a SIP transaction into two call legs","domain":"en.wikipedia.org","publisherName":"Wikipedia"},{"title":"SIP Trunking Hardware Requirements: Ensure Your System is Ready","url":"https://didlogic.com/blog/sip-trunking-hardware-requirements/","excerpt":"IP-PBX (hardware appliance, software-based, or cloud-hosted); Session Border Controller (SBC); SIP phones or softphones; business-class router with Quality of Service (QoS); managed PoE switch; VoIP gateway to bridge legacy PBX or analog lines to SIP. G.711 uses about 85 kbps per call including overhead; acceptable ranges: latency under 150 ms, jitter under 30 ms, packet loss under 1%.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"Equipment list for a SIP trunk deployment: IP-PBX (appliance, software, or cloud-hosted), SIP phones or softphones, session border controller, business-class router with QoS, managed PoE switch, and a VoIP gateway for legacy PBX or analog lines; per-call bandwidth and quality thresholds","domain":"didlogic.com","publisherName":"DIDLogic"},{"title":"SIP Trunking vs. PRI: Differences, Benefits, & What To Use","url":"https://www.bandwidth.com/blog/pri_vs_sip/","excerpt":"PRI systems can support 23 channels per physical circuit, so if businesses need additional capacity, then that means they'll need to install and maintain additional circuits and hardware to support. SIP, on the other hand, represents a digital call path and therefore is easier to scale up and down with seasonal demand.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"PRI supports 23 channels per physical circuit and needs added circuits and hardware to scale; a SIP trunk scales up and down virtually over the internet connection without new hardware","domain":"bandwidth.com","publisherName":"Bandwidth"},{"title":"G.711 vs G.729: VoIP Codec Comparison","url":"https://telcobridges.com/learning/sip-trunking/g711-vs-g729/","excerpt":"G.711 generates 50 packets per second, each carrying 160 bytes of audio plus about 54 bytes of RTP/UDP/IP/Ethernet headers, for a total of around 85 to 87 kbps per direction. G.729 generates the same 50 packets per second, each carrying 20 bytes of audio plus the same 54 bytes of headers, for a total of around 31 kbps per direction.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"Per-call bandwidth including IP, UDP, and RTP overhead: G.711 about 85 to 87 kbps per direction, G.729 about 31 kbps per direction; used to size the internet connection to concurrent calls","domain":"telcobridges.com","publisherName":"TelcoBridges"},{"title":"ITU-T G.711: Pulse code modulation (PCM) of voice frequencies","url":"https://www.itu.int/rec/T-REC-G.711/","excerpt":"Pulse code modulation (PCM) of voice frequencies. Status: In force.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-18T00:00:00","supportsText":"G.711 is the ITU-T PCM voice-coding recommendation, the reference narrowband codec cited when sizing per-call bandwidth; status in force","domain":"itu.int","publisherName":"International Telecommunication Union (ITU-T)"},{"title":"47 CFR 9.11: E911 service requirements for interconnected VoIP","url":"https://www.law.cornell.edu/cfr/text/47/9.11","excerpt":"The service provider has obtained from the customer, prior to the initiation of service, the Registered Location (as defined in Sec. 9.3) at which the service will first be used... circumstances under which E911 service may not be available... include... relocation of the end user's IP-compatible CPE, use by the end user of a non-native telephone number, broadband connection failure, loss of electrical power.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-18T00:00:00","supportsText":"Interconnected VoIP providers must obtain the Registered Location of the equipment for E911 and advise subscribers when E911 may be unavailable, including on relocation of the customer premises equipment, use of a non-native number, broadband failure, or power loss","domain":"law.cornell.edu","publisherName":"Legal Information Institute, Cornell Law School"},{"title":"SIP Trunking Compatibility Issues: What You Need to Know","url":"https://www.atlantech.net/blog/sip-trunking-compatibility-issues-what-you-need-to-know","excerpt":"In order to successfully deploy SIP trunking you need: An ITSP or a VoIP provider. IP Private Branch Exchange (PBX). A Border Element or gateway.","quoteText":null,"sourceRole":"SUPPORTING","verifiedAt":"2026-07-18T00:00:00","supportsText":"Plain-language corroboration that a SIP trunk deployment needs three parts that must communicate: a provider (ITSP), an IP-enabled PBX, and a border element or gateway; corroborated by [1] and [9]","domain":"atlantech.net","publisherName":"Atlantech Online"}],"revisions":[],"relatedAnswers":[{"id":"460c9c02-3a75-4382-b125-ea1587c82aa8","slug":"what-kind-of-internet-connection-and-bandwidth-do-i-need-for-sip-trunking","question":"What kind of internet connection and bandwidth do I need for SIP trunking?","publishedAt":"2026-07-18T15:16:59.312","confidenceScore":90,"confidenceLabel":"High","industry":{"id":"ff619d7c-d7d7-485e-a05a-53fba07f33ed","slug":"telecommunications","label":"Telecommunications","description":"Business voice, fiber, UCaaS, and network services"},"topic":{"slug":"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 runs over a standard broadband connection, but its size and quality both matter. Budget about 85 to 90 kbps per simultaneous call on G.711 or roughly 31 kbps on G.729a, multiply by your busy-hour concurrent calls, and add 10 to 20 percent of headroom. The connection also has to hold one-way latency under about 150 ms, jitter under about 30 ms, and packet loss under roughly 1 percent, with upload usually the limiting direction. A symmetric business or fiber link with quality of service is the common recommendation.","url":"/q/what-kind-of-internet-connection-and-bandwidth-do-i-need-for-sip-trunking"},{"id":"5dd7bb8f-a2ec-4293-ad95-35aba460ad29","slug":"can-i-use-sip-trunking-with-my-existing-phone-system","question":"Can I use SIP trunking with my existing phone system?","publishedAt":"2026-07-18T15:16:55.298","confidenceScore":90,"confidenceLabel":"High","industry":{"id":"ff619d7c-d7d7-485e-a05a-53fba07f33ed","slug":"telecommunications","label":"Telecommunications","description":"Business voice, fiber, UCaaS, and network services"},"topic":{"slug":"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":"Whether a SIP trunk fits your current phone system comes down to what that system is. Most IP-PBX platforms from the past decade speak SIP and connect by pasting in provider credentials. A legacy PBX on PRI or T1 connects through a digital gateway that converts SIP to the circuit it expects, and analog phones connect through an FXS adapter. Keeping the system you have avoids a full replacement, but the trunk still depends on your internet link and carries 911 obligations.","url":"/q/can-i-use-sip-trunking-with-my-existing-phone-system"},{"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":88,"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":"Simultaneous calls on a SIP trunk are counted in channels, one call per channel. Size the number you need from busy-hour concurrency, not headcount: a rough office baseline is one channel per three to four employees, a contact center closer to one per agent, and Erlang B gives the precise figure at a set blocking target. The number you can get is capped by bandwidth, codec, and your provider's plan, with each call using about 80 to 115 kbps and channels adjustable in software.","url":"/q/how-many-simultaneous-calls-do-i-need-or-get-on-a-sip-trunk"},{"id":"15b4d605-db80-4b61-876a-d42f6308be0d","slug":"what-is-a-sip-trunk-channel","question":"What is a SIP trunk channel?","publishedAt":"2026-07-18T15:01:36.739","confidenceScore":90,"confidenceLabel":"High","industry":{"id":"ff619d7c-d7d7-485e-a05a-53fba07f33ed","slug":"telecommunications","label":"Telecommunications","description":"Business voice, fiber, UCaaS, and network services"},"topic":{"slug":"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 channel is one simultaneous call path: each active call, inbound or outbound, uses one channel and releases it when the call ends. Channel count, not how many phone numbers you own, sets how many calls run at once. Capacity is virtual and limited mainly by bandwidth, roughly 85 to 115 kbps per channel, so channels scale in software rather than through new circuits. Providers sell either fixed channel bundles or elastic trunks that cap calls per second instead.","url":"/q/what-is-a-sip-trunk-channel"}],"contributorStats":{"verifiedAnswers":224,"openDisputes":0},"schemaJson":{"@context":"https://schema.org","@type":"Question","name":"What equipment do I need to use SIP trunking?","text":"What equipment do I need to use SIP trunking?","url":"https://www.answerstack.io/q/what-equipment-do-i-need-to-use-sip-trunking","answerCount":1,"datePublished":"2026-07-18T15:16:57.319","author":{"@type":"Person","name":"AnswerStack Editorial Team","worksFor":{"@type":"Organization","name":"AnswerStack"},"url":"https://www.answerstack.io/contributors/answer-stack"},"about":[{"@type":"Thing","name":"SIP Trunking"},{"@type":"Thing","name":"Telecommunications"}],"acceptedAnswer":{"@type":"Answer","text":"To use SIP trunking you need a SIP-capable phone system, endpoints for the people placing calls, an internet connection sized for voice, and a session border controller at the network edge, plus a gateway only if you are keeping analog or legacy equipment.[1][9][14] The phone system is usually an IP-PBX, either an appliance or software you run on-site or a hosted PBX the provider operates for you, and it speaks the Session Initiation Protocol to set up and tear down each call.[1][2] The people on the calls use IP desk phones or softphones running on a computer or mobile device, and hardware phones commonly draw power over the same Ethernet cable that carries their traffic.[4] A session border controller sits between your system and the provider to secure the SIP session, reconcile mismatched settings, and act as a back-to-back user agent on both call legs.[7][8] If you still have analog phones, fax machines, or a PRI-era PBX, an analog telephone adapter or VoIP gateway converts that traffic to SIP so it can ride the trunk.[5][6]","url":"https://www.answerstack.io/q/what-equipment-do-i-need-to-use-sip-trunking","upvoteCount":0,"datePublished":"2026-07-18T15:16:57.319","dateModified":"2026-07-18T00:00:00","author":{"@type":"Person","name":"AnswerStack Editorial Team","worksFor":{"@type":"Organization","name":"AnswerStack"},"url":"https://www.answerstack.io/contributors/answer-stack"},"citation":[{"@type":"CreativeWork","name":"SIP trunking","url":"https://en.wikipedia.org/wiki/SIP_trunking"},{"@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":"VoIP phone","url":"https://en.wikipedia.org/wiki/VoIP_phone"},{"@type":"CreativeWork","name":"Analog telephone adapter","url":"https://en.wikipedia.org/wiki/Analog_telephone_adapter"},{"@type":"CreativeWork","name":"Media gateway","url":"https://en.wikipedia.org/wiki/Media_gateway"},{"@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":"Session border controller","url":"https://en.wikipedia.org/wiki/Session_border_controller"},{"@type":"CreativeWork","name":"SIP Trunking Hardware Requirements: Ensure Your System is Ready","url":"https://didlogic.com/blog/sip-trunking-hardware-requirements/"},{"@type":"CreativeWork","name":"SIP Trunking vs. PRI: Differences, Benefits, & What To Use","url":"https://www.bandwidth.com/blog/pri_vs_sip/"},{"@type":"CreativeWork","name":"G.711 vs G.729: VoIP Codec Comparison","url":"https://telcobridges.com/learning/sip-trunking/g711-vs-g729/"},{"@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":"47 CFR 9.11: E911 service requirements for interconnected VoIP","url":"https://www.law.cornell.edu/cfr/text/47/9.11"},{"@type":"CreativeWork","name":"SIP Trunking Compatibility Issues: What You Need to Know","url":"https://www.atlantech.net/blog/sip-trunking-compatibility-issues-what-you-need-to-know"}]}}}