SIP trunking is a way to deliver phone service over an internet connection instead of over dedicated telephone lines, using the Session Initiation Protocol to set up and tear down calls between a business phone system and a provider that connects to the public telephone network.[1][2] The trunk itself is virtual: an internet telephony service provider carries your calls in software, so a company with an IP-PBX no longer needs the physical T1 or PRI circuits that older systems required.[1][6] SIP handles the call signaling while the voice travels separately as RTP media packets, and a session border controller usually sits at the network edge to secure and normalize the traffic.[2][3][5] Because capacity is measured in concurrent call paths rather than physical lines, you can add or remove channels without installing hardware, and a service that reaches the regular phone network falls under the FCC's interconnected VoIP rules for 911 and caller location.[6][10][11]
\n","summary":"A SIP trunk is a virtual voice circuit that carries calls over the internet between a company's phone system and its provider, replacing the physical PRI or T1 lines older setups used. Signaling runs on the Session Initiation Protocol, audio travels as RTP packets, and a session border controller secures the edge. Capacity is counted in concurrent call paths you can scale in software, and providers that reach the public phone network carry 911 and caller-location obligations.","publishedAt":"2026-07-18T14:40:06.195","verifiedAt":"2026-07-17T00:00:00","editorialStatus":"APPROVED","lastReviewedAt":"2026-07-17T00:00:00","nextReviewDueAt":"2026-10-17T00:00:00","templateVersion":"v2","aliases":["What is a SIP trunk?","How does SIP trunking work?","SIP trunking meaning","What does SIP trunking stand for?","SIP trunk vs PRI","What is a SIP trunk provider?","How does a SIP trunk connect to the PSTN?","What is Session Initiation Protocol trunking?","SIP trunking explained","Do I need a session border controller for SIP trunking?","How many channels does a SIP trunk have?","Is SIP trunking the same as VoIP?"],"confidenceScore":91,"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":"915ea305-7191-4ab1-8e72-b741eeb6762a","sectionKey":"how_it_works","sectionType":"markdown_section","heading":"How does SIP trunking work?","introMarkdown":"SIP trunking carries your phone calls over an internet connection between your phone system and a provider that reaches the public telephone network, so you no longer buy the physical trunk lines a traditional setup required.[1][13] The name comes from the Session Initiation Protocol, the standard the industry uses to start, change, and end a call, whether that call is voice, video, or a mix.[2] A trunk in telephone terms is simply the shared link that connects a private phone system to the outside world, and with SIP trunking that link is a software session riding your existing data connection rather than copper pairs or a T1 circuit.[1]\n\n### Signaling and media travel separately\n\nTwo jobs run at the same time on every call. SIP does the signaling, which means it locates the far end, negotiates which audio formats both sides can handle, and then connects or ends the call as the conversation requires.[2] The voice itself does not travel inside SIP. Once both ends agree, the audio flows as Real-time Transport Protocol packets, which timestamp and sequence each slice of sound so the far end can play it back in order.[3] Splitting the two lets a provider route a call intelligently while the audio takes the most direct path.\n\n### Where the provider fits\n\nBetween your building and the public phone network sits an internet telephony service provider, usually shortened to ITSP.[1] Your phone system hands each outbound call to that provider over the SIP trunk, and the provider passes it into the wider phone network so it can reach any mobile or landline number, with inbound calls traveling the reverse route.[4] Most business deployments place a session border controller at that boundary, a device or software instance that ends the provider's session and starts a fresh one toward your PBX, which gives it control over security and call formatting on both sides.[4][5]","introHtml":"SIP trunking carries your phone calls over an internet connection between your phone system and a provider that reaches the public telephone network, so you no longer buy the physical trunk lines a traditional setup required.[1][13] The name comes from the Session Initiation Protocol, the standard the industry uses to start, change, and end a call, whether that call is voice, video, or a mix.[2] A trunk in telephone terms is simply the shared link that connects a private phone system to the outside world, and with SIP trunking that link is a software session riding your existing data connection rather than copper pairs or a T1 circuit.[1]
\nTwo jobs run at the same time on every call. SIP does the signaling, which means it locates the far end, negotiates which audio formats both sides can handle, and then connects or ends the call as the conversation requires.[2] The voice itself does not travel inside SIP. Once both ends agree, the audio flows as Real-time Transport Protocol packets, which timestamp and sequence each slice of sound so the far end can play it back in order.[3] Splitting the two lets a provider route a call intelligently while the audio takes the most direct path.
\nBetween your building and the public phone network sits an internet telephony service provider, usually shortened to ITSP.[1] Your phone system hands each outbound call to that provider over the SIP trunk, and the provider passes it into the wider phone network so it can reach any mobile or landline number, with inbound calls traveling the reverse route.[4] Most business deployments place a session border controller at that boundary, a device or software instance that ends the provider's session and starts a fresh one toward your PBX, which gives it control over security and call formatting on both sides.[4][5]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":0},{"id":"a27076ce-7d0e-4aea-82e8-948b66da39e0","sectionKey":"sip_vs_pri","sectionType":"table_section","heading":"How is a SIP trunk different from a PRI line?","introMarkdown":"A PRI is a fixed bundle of physical channels, while a SIP trunk is capacity delivered in software over your internet connection.[6] A Primary Rate Interface, the digital circuit many businesses used before SIP, carries a set number of voice channels on a physical T1 line, which in North America means 23 usable channels per circuit.[6] Adding a twenty-fourth simultaneous call meant ordering a second circuit and waiting for it to be installed. A SIP trunk counts capacity as concurrent call paths a provider can raise or lower without new wiring, so the ceiling becomes your available bandwidth rather than a hardware limit.[6] The table compares the two on the points that tend to drive the decision.","introHtml":"A PRI is a fixed bundle of physical channels, while a SIP trunk is capacity delivered in software over your internet connection.[6] A Primary Rate Interface, the digital circuit many businesses used before SIP, carries a set number of voice channels on a physical T1 line, which in North America means 23 usable channels per circuit.[6] Adding a twenty-fourth simultaneous call meant ordering a second circuit and waiting for it to be installed. A SIP trunk counts capacity as concurrent call paths a provider can raise or lower without new wiring, so the ceiling becomes your available bandwidth rather than a hardware limit.[6] The table compares the two on the points that tend to drive the decision.
\n","outroMarkdown":"PRI still fits where a site has poor internet or strict isolation needs, because a dedicated circuit does not compete with data traffic for bandwidth. For most organizations with a reliable connection, sizing capacity to real demand instead of paying for a full circuit year-round is why most new deployments choose SIP trunking.[6]","outroHtml":"PRI still fits where a site has poor internet or strict isolation needs, because a dedicated circuit does not compete with data traffic for bandwidth. For most organizations with a reliable connection, sizing capacity to real demand instead of paying for a full circuit year-round is why most new deployments choose SIP trunking.[6]
\n","contentJson":{"rows":[{"cells":["Physical basis","Dedicated digital circuit installed to your building","Virtual session over your existing internet link [1]"]},{"cells":["Channel capacity","23 voice channels per T1 in North America [6]","Concurrent call paths limited mainly by bandwidth [6]"]},{"cells":["Scaling","Order and install another circuit for more capacity [6]","Provider adjusts channels in software, often same day [6]"]},{"cells":["Number handling","Numbers tied to the physical circuit","Numbers provisioned and ported in the provider portal [1]"]},{"cells":["Voice transport","Time-division voice on the circuit","Voice as RTP packets over IP [3]"]}],"columns":["Factor","PRI (T1 circuit)","SIP trunk"]},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":1},{"id":"d76ac815-ebc8-47e9-ada2-e6693b84b3bf","sectionKey":"components_table","sectionType":"table_section","heading":"What you need to run a SIP trunk","introMarkdown":"A working SIP trunk deployment depends on five pieces that fit together: the phone system that speaks SIP, the provider that carries your calls, a session border controller at the edge, an internet connection sized for voice, and the numbers and channels the provider assigns.[1][4] The table lists them at a glance, and the sections after it explain what each one does and how to choose or size it.","introHtml":"A working SIP trunk deployment depends on five pieces that fit together: the phone system that speaks SIP, the provider that carries your calls, a session border controller at the edge, an internet connection sized for voice, and the numbers and channels the provider assigns.[1][4] The table lists them at a glance, and the sections after it explain what each one does and how to choose or size it.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{"rows":[{"cells":["SIP-capable phone system (IP-PBX)","Originates and receives calls in SIP","Confirm the PBX or gateway supports SIP natively [1]"]},{"cells":["SIP trunk provider (ITSP)","Connects your trunk to the public phone network","Compare coverage, pricing model, and E911 support [1]"]},{"cells":["Session border controller","Secures and normalizes SIP at the network edge","Needed in most business deployments [5]"]},{"cells":["Internet connection","Carries signaling and voice packets","Size bandwidth to concurrent calls plus headroom [8]"]},{"cells":["Numbers and channels","Your phone numbers and simultaneous-call capacity","Count busy-hour concurrent calls, not total phones [6]"]}],"columns":["Component","What it does","How to approach it"]},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":2},{"id":"43d4f6ba-acad-4f97-a553-c13105c8404a","sectionKey":"component_pbx","sectionType":"markdown_section","heading":"The SIP-capable phone system (IP-PBX)","introMarkdown":"A SIP trunk needs a phone system that can speak SIP, which for most businesses is an IP-PBX. This is the on-premises or virtual system that routes internal calls, applies dialing rules, and hands outbound calls to the trunk.[1] Older PBX equipment built for analog or PRI lines can still work, but it needs a gateway that converts its signaling into SIP before the call reaches the provider. The point that matters for planning is compatibility, because not every PBX implements SIP the same way, and small differences in how each side formats headers or negotiates codecs are a common cause of one-way audio and dropped calls.[5] Before you commit, check your PBX vendor's list of certified providers, or plan for a session border controller to reconcile the differences, since sorting this out in advance avoids a painful troubleshooting cycle after cutover.","introHtml":"A SIP trunk needs a phone system that can speak SIP, which for most businesses is an IP-PBX. This is the on-premises or virtual system that routes internal calls, applies dialing rules, and hands outbound calls to the trunk.[1] Older PBX equipment built for analog or PRI lines can still work, but it needs a gateway that converts its signaling into SIP before the call reaches the provider. The point that matters for planning is compatibility, because not every PBX implements SIP the same way, and small differences in how each side formats headers or negotiates codecs are a common cause of one-way audio and dropped calls.[5] Before you commit, check your PBX vendor's list of certified providers, or plan for a session border controller to reconcile the differences, since sorting this out in advance avoids a painful troubleshooting cycle after cutover.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":3},{"id":"cbb474bd-fbc1-449a-ad0e-04f2490637fd","sectionKey":"component_provider","sectionType":"markdown_section","heading":"The SIP trunk provider (ITSP)","introMarkdown":"The provider, formally an internet telephony service provider, is the company that carries your calls between the trunk and the public phone network.[1] It assigns your phone numbers, routes calls to and from other carriers, and bills you for capacity or usage. Providers differ in ways that affect both cost and reliability, including the geographic footprint where they hold numbers, whether they price by channel or by the minute, how they handle number porting, and whether emergency calling is bundled or sold separately.[9] Because the provider becomes the path for every call your business makes, it is worth asking how their network is built, whether they operate their own connections to the phone network or resell someone else's, and what their record is on outages. A provider that reaches the public network also carries regulatory duties for 911 and caller identification, so confirming they meet those obligations protects you as much as them.[10][11]","introHtml":"The provider, formally an internet telephony service provider, is the company that carries your calls between the trunk and the public phone network.[1] It assigns your phone numbers, routes calls to and from other carriers, and bills you for capacity or usage. Providers differ in ways that affect both cost and reliability, including the geographic footprint where they hold numbers, whether they price by channel or by the minute, how they handle number porting, and whether emergency calling is bundled or sold separately.[9] Because the provider becomes the path for every call your business makes, it is worth asking how their network is built, whether they operate their own connections to the phone network or resell someone else's, and what their record is on outages. A provider that reaches the public network also carries regulatory duties for 911 and caller identification, so confirming they meet those obligations protects you as much as them.[10][11]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":4},{"id":"cf680d38-98a5-4b62-b5df-ad14c772d281","sectionKey":"component_sbc","sectionType":"markdown_section","heading":"The session border controller","introMarkdown":"A session border controller sits at the edge of your network and manages every SIP call passing between your phone system and the provider.[5] It works as a back-to-back user agent, which means it fully ends the provider's session and starts a new one toward your PBX, giving it authority over the call on both legs.[4] That position lets it do several jobs a plain firewall cannot: it hides your internal network layout, blocks malformed or hostile SIP traffic, translates between codecs when the two sides do not match, and handles the address translation that lets private IP phones reach a public provider.[5] The IETF's requirements document for SBC deployments groups these functions into perimeter defense, connectivity fixes such as NAT traversal, and traffic management like media monitoring and quality of service.[5] For a small setup the provider may supply this function from the cloud, while larger or security-sensitive sites often run their own SBC so they control the policies at their border.","introHtml":"A session border controller sits at the edge of your network and manages every SIP call passing between your phone system and the provider.[5] It works as a back-to-back user agent, which means it fully ends the provider's session and starts a new one toward your PBX, giving it authority over the call on both legs.[4] That position lets it do several jobs a plain firewall cannot: it hides your internal network layout, blocks malformed or hostile SIP traffic, translates between codecs when the two sides do not match, and handles the address translation that lets private IP phones reach a public provider.[5] The IETF's requirements document for SBC deployments groups these functions into perimeter defense, connectivity fixes such as NAT traversal, and traffic management like media monitoring and quality of service.[5] For a small setup the provider may supply this function from the cloud, while larger or security-sensitive sites often run their own SBC so they control the policies at their border.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":5},{"id":"4ad99534-88f5-4804-b55a-eba4f35ae1e6","sectionKey":"component_bandwidth","sectionType":"markdown_section","heading":"An internet connection sized for voice","introMarkdown":"Voice quality on a SIP trunk depends on the internet connection carrying it, so sizing bandwidth to your call volume is part of the setup rather than an afterthought.[8] Each concurrent call needs roughly 85 to 87 kilobits per second in each direction with the common G.711 codec, once the IP, UDP, and RTP headers are counted on top of the 64 kilobit payload.[7][8] A more compressed codec such as G.729 drops that to about 31 kilobits per second per direction at a small cost in fidelity, which helps where bandwidth is tight.[8] Multiply your busy-hour concurrent calls by the per-call figure, add headroom for the data traffic sharing the link, and you have a floor for the connection you need. Voice is sensitive to delay, jitter, and lost packets, so prioritizing call traffic with quality-of-service settings on your network is what keeps audio clean once the link gets busy.[8]","introHtml":"Voice quality on a SIP trunk depends on the internet connection carrying it, so sizing bandwidth to your call volume is part of the setup rather than an afterthought.[8] Each concurrent call needs roughly 85 to 87 kilobits per second in each direction with the common G.711 codec, once the IP, UDP, and RTP headers are counted on top of the 64 kilobit payload.[7][8] A more compressed codec such as G.729 drops that to about 31 kilobits per second per direction at a small cost in fidelity, which helps where bandwidth is tight.[8] Multiply your busy-hour concurrent calls by the per-call figure, add headroom for the data traffic sharing the link, and you have a floor for the connection you need. Voice is sensitive to delay, jitter, and lost packets, so prioritizing call traffic with quality-of-service settings on your network is what keeps audio clean once the link gets busy.[8]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":6},{"id":"55b56dfd-f351-46e9-910c-fb50085df3d9","sectionKey":"component_numbers_channels","sectionType":"markdown_section","heading":"Phone numbers and concurrent call channels","introMarkdown":"A SIP trunk is sized in concurrent call paths, also called channels, which is the number of calls you can run at the same time rather than the number of phones or numbers you own.[6] A company with fifty employees rarely has fifty calls happening at once, so the useful measure is how many simultaneous calls you place at your busiest hour, plus a little headroom.[6] Separate from channels are your direct inward dial numbers, the individual phone numbers that route to specific people or departments, and you can hold far more numbers than channels because most of them sit idle at any given moment.[9] This split is one of the practical advantages of SIP over a PRI, where numbers and capacity were bound to the physical circuit. With a trunk you can carry hundreds of numbers on a handful of channels, and you can raise the channel count for a seasonal peak and lower it again without touching hardware.[6]","introHtml":"A SIP trunk is sized in concurrent call paths, also called channels, which is the number of calls you can run at the same time rather than the number of phones or numbers you own.[6] A company with fifty employees rarely has fifty calls happening at once, so the useful measure is how many simultaneous calls you place at your busiest hour, plus a little headroom.[6] Separate from channels are your direct inward dial numbers, the individual phone numbers that route to specific people or departments, and you can hold far more numbers than channels because most of them sit idle at any given moment.[9] This split is one of the practical advantages of SIP over a PRI, where numbers and capacity were bound to the physical circuit. With a trunk you can carry hundreds of numbers on a handful of channels, and you can raise the channel count for a seasonal peak and lower it again without touching hardware.[6]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":7},{"id":"0a07f624-2e2d-418d-b025-4d0ea636c772","sectionKey":"pricing","sectionType":"markdown_section","heading":"How much does SIP trunking cost and how is it sized?","introMarkdown":"SIP trunk pricing follows two main models, and which one fits depends on how predictable your call volume is.[9] Channel-based plans charge a flat monthly fee per concurrent call path, commonly in the range of 15 to 30 dollars per channel for unlimited domestic calling, which suits businesses with steady, high call volume that want a predictable bill.[9] Metered plans instead charge per minute, often around one to three cents a minute, which works out cheaper for organizations whose lines sit quiet much of the day.[9] On top of either model you pay a small monthly fee for each phone number, commonly about a dollar per number, with enhanced numbers that include emergency calling costing more.[9] A common mistake is buying a channel for every phone, because capacity is measured in simultaneous calls, so most businesses need far fewer channels than handsets, and right-sizing that number is where the savings over a PRI actually come from.[6][9]","introHtml":"SIP trunk pricing follows two main models, and which one fits depends on how predictable your call volume is.[9] Channel-based plans charge a flat monthly fee per concurrent call path, commonly in the range of 15 to 30 dollars per channel for unlimited domestic calling, which suits businesses with steady, high call volume that want a predictable bill.[9] Metered plans instead charge per minute, often around one to three cents a minute, which works out cheaper for organizations whose lines sit quiet much of the day.[9] On top of either model you pay a small monthly fee for each phone number, commonly about a dollar per number, with enhanced numbers that include emergency calling costing more.[9] A common mistake is buying a channel for every phone, because capacity is measured in simultaneous calls, so most businesses need far fewer channels than handsets, and right-sizing that number is where the savings over a PRI actually come from.[6][9]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":8},{"id":"757b6baa-23ca-44e1-9d83-758bc2f6274b","sectionKey":"contributor_perspective","sectionType":"markdown_section","heading":"How this answer was researched","introMarkdown":"This entry draws its protocol claims from the standards themselves and its regulatory claims from the rules as written. The definitions of SIP and RTP come from the IETF documents that specify them, and the description of what a session border controller does is taken from the IETF's requirements document for SBC deployments.[2][3][5] Capacity and cost points are attributed to independent industry sources rather than presented as fixed facts, since pricing and channel practices vary by provider and change over time.[6][8][9] The 911 and interconnected VoIP obligations come from the Code of Federal Regulations, and the fraud figures come from the Communications Fraud Control Association's own survey.[10][11][12] Telecom products and prices move quickly, so the specific numbers here reflect what the cited sources said on the verification date. Practitioners who provision, secure, or regulate SIP trunks are welcome to suggest corrections or additions, which are checked against primary sources before any update.","introHtml":"This entry draws its protocol claims from the standards themselves and its regulatory claims from the rules as written. The definitions of SIP and RTP come from the IETF documents that specify them, and the description of what a session border controller does is taken from the IETF's requirements document for SBC deployments.[2][3][5] Capacity and cost points are attributed to independent industry sources rather than presented as fixed facts, since pricing and channel practices vary by provider and change over time.[6][8][9] The 911 and interconnected VoIP obligations come from the Code of Federal Regulations, and the fraud figures come from the Communications Fraud Control Association's own survey.[10][11][12] Telecom products and prices move quickly, so the specific numbers here reflect what the cited sources said on the verification date. Practitioners who provision, secure, or regulate SIP trunks are welcome to suggest corrections or additions, 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":9},{"id":"b71d6c23-7e09-4094-8a94-af729b0f76e5","sectionKey":"trade_offs","sectionType":"markdown_section","heading":"Trade-offs and limits worth knowing","introMarkdown":"SIP trunking depends on your internet connection and exposes your phone system to the public network, so a few limits deserve planning that a legacy circuit did not require.\n\n### Toll fraud and security\n\nBecause a SIP trunk connects to the internet, an exposed or misconfigured system can be found and abused by attackers who route expensive international calls through it. Telecom fraud is not a small problem: the Communications Fraud Control Association estimated 38.95 billion dollars in global losses in 2023, up 12 percent from two years earlier, with PBX fraud among the top methods reported.[12] Strong credentials, closed unused ports, encryption of signaling and media, and provider-side limits on call destinations and spending are the usual defenses.\n\n### Dependence on internet and power\n\nA SIP trunk stops working if the internet link or local power fails, unlike an analog line that draws power from the phone company. Sites that cannot tolerate downtime pair the trunk with a backup connection and battery power, and configure the provider to reroute calls to mobiles or another site when the primary link goes dark.\n\n### Emergency calling obligations\n\nA provider that lets you call the public phone network is an interconnected VoIP service under FCC rules, which means 911 calls must carry a call-back number and a validated location, and subscribers have to be told plainly when 911 might not work, such as during an outage.[10][11] Because a trunk can serve numbers registered at different addresses, keeping location records accurate for each number is an ongoing responsibility rather than a one-time task.\n\n### Call quality under load\n\nVoice shares the connection with everything else on your network, so a link that is fine for email can still produce choppy calls when it saturates. Prioritizing voice traffic and keeping enough headroom for the number of concurrent calls is what prevents that.[8]","introHtml":"SIP trunking depends on your internet connection and exposes your phone system to the public network, so a few limits deserve planning that a legacy circuit did not require.
\nBecause a SIP trunk connects to the internet, an exposed or misconfigured system can be found and abused by attackers who route expensive international calls through it. Telecom fraud is not a small problem: the Communications Fraud Control Association estimated 38.95 billion dollars in global losses in 2023, up 12 percent from two years earlier, with PBX fraud among the top methods reported.[12] Strong credentials, closed unused ports, encryption of signaling and media, and provider-side limits on call destinations and spending are the usual defenses.
\nA SIP trunk stops working if the internet link or local power fails, unlike an analog line that draws power from the phone company. Sites that cannot tolerate downtime pair the trunk with a backup connection and battery power, and configure the provider to reroute calls to mobiles or another site when the primary link goes dark.
\nA provider that lets you call the public phone network is an interconnected VoIP service under FCC rules, which means 911 calls must carry a call-back number and a validated location, and subscribers have to be told plainly when 911 might not work, such as during an outage.[10][11] Because a trunk can serve numbers registered at different addresses, keeping location records accurate for each number is an ongoing responsibility rather than a one-time task.
\nVoice shares the connection with everything else on your network, so a link that is fine for email can still produce choppy calls when it saturates. Prioritizing voice traffic and keeping enough headroom for the number of concurrent calls is what prevents that.[8]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":10},{"id":"36fbb6bb-ec1d-4918-89cb-41fb0b7db79e","sectionKey":"what_it_is_not","sectionType":"markdown_section","heading":"What SIP trunking is not","introMarkdown":"SIP trunking is one specific way to deliver business phone service, and it gets confused with several neighboring ideas.\n\n### It is not VoIP in general\n\nVoIP is the broad category of carrying calls over IP networks, and SIP trunking is one form of it aimed at connecting a phone system to the public network.[1] A softphone app or a hosted phone system is also VoIP, yet neither is a SIP trunk.\n\n### It is not a hosted or cloud phone system\n\nWith SIP trunking you still run your own phone system, on-site or virtual, and the trunk only connects it outward.[1] A hosted PBX or UCaaS product moves the whole phone system into a provider's data center, so the two address different parts of the problem and are sometimes used together.\n\n### It is not the SIP protocol by itself\n\nSIP is the signaling standard that sets up calls for many applications, including video and messaging.[2] A SIP trunk is a commercial service built on that protocol to carry voice between your system and a carrier, so the protocol is the ingredient and the trunk is the finished service.\n\n### It is not a PRI\n\nA PRI is a physical digital circuit with a fixed channel count, while a SIP trunk delivers the same connectivity as software over your internet link.[6] Both connect your phones to the outside world, but only the SIP trunk scales without new hardware.","introHtml":"SIP trunking is one specific way to deliver business phone service, and it gets confused with several neighboring ideas.
\nVoIP is the broad category of carrying calls over IP networks, and SIP trunking is one form of it aimed at connecting a phone system to the public network.[1] A softphone app or a hosted phone system is also VoIP, yet neither is a SIP trunk.
\nWith SIP trunking you still run your own phone system, on-site or virtual, and the trunk only connects it outward.[1] A hosted PBX or UCaaS product moves the whole phone system into a provider's data center, so the two address different parts of the problem and are sometimes used together.
\nSIP is the signaling standard that sets up calls for many applications, including video and messaging.[2] A SIP trunk is a commercial service built on that protocol to carry voice between your system and a carrier, so the protocol is the ingredient and the trunk is the finished service.
\nA PRI is a physical digital circuit with a fixed channel count, while a SIP trunk delivers the same connectivity as software over your internet link.[6] Both connect your phones to the outside world, but only the SIP trunk scales without new hardware.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":11}],"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).","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-17T00:00:00","supportsText":"Core definition of SIP trunking as a VoIP service based on SIP by which ITSPs deliver telephone service to IP-PBX equipped customers; virtual trunk replacing physical lines; ITSP role and number handling","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":"2026-07-17T00:00:00","supportsText":"SIP as the application-layer signaling protocol that creates, modifies, and terminates sessions such as Internet telephone calls","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.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-17T00:00:00","supportsText":"RTP carries the real-time audio media of a SIP call, with sequencing and timestamps, separate from SIP signaling","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"SIP Trunking: Architecture, Security & SBC Guide for Service Providers","url":"https://telcobridges.com/learning/sip-trunking/","excerpt":"a logical connection between two SIP endpoints, typically between a service provider's SBC and a carrier's SBC, or between an SBC and an IP-PBX, that carries both signaling (call setup, teardown, mid-call events) and media (the actual voice audio) over the same IP transport.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-17T00:00:00","supportsText":"SIP trunk as a logical connection between SIP endpoints (SBC or IP-PBX) carrying signaling and media over IP; SBC as back-to-back user agent that terminates and re-originates the session; replacement for legacy PRI","domain":"telcobridges.com","publisherName":"TelcoBridges"},{"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":"2026-07-17T00:00:00","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":"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.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-17T00:00:00","supportsText":"PRI supports up to 23 concurrent channels per physical circuit and needs added circuits and hardware to scale; SIP channels adjust virtually, limited mainly by bandwidth; pay for what you use","domain":"bandwidth.com","publisherName":"Bandwidth"},{"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":"2026-07-17T00:00:00","supportsText":"G.711 is the ITU-T PCM voice-coding recommendation, the reference narrowband codec used to size per-call bandwidth; status in force","domain":"itu.int","publisherName":"International Telecommunication Union (ITU-T)"},{"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.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-17T00:00:00","supportsText":"Per-call bandwidth including IP, UDP, and RTP overhead: G.711 about 85 to 87 kbps per direction (64 kbps payload), G.729 about 31 kbps per direction; roughly 20 ms packetization; voice sensitive to delay, jitter, loss","domain":"telcobridges.com","publisherName":"TelcoBridges"},{"title":"How Much Does a SIP Trunk Cost? Your 2025 Pricing Guide","url":"https://www.sip.us/blog/latest-news/how-much-does-a-sip-trunk-cost/","excerpt":"SIP trunk pricing typically starts around $19.99 per channel monthly for unlimited plans and generally ranges from $15 to $30 monthly. Metered: you pay for what you use, typically ranging from $0.01 to $0.03 per minute.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-17T00:00:00","supportsText":"Channel pricing about 15 to 30 dollars per channel per month for unlimited domestic plans; metered about 0.01 to 0.03 dollars per minute; DID numbers about a dollar per month, enhanced E911 numbers more; metered vs channel model fit","domain":"sip.us","publisherName":"SIP.US"},{"title":"47 CFR 9.3, Definitions: Interconnected VoIP service","url":"https://www.law.cornell.edu/cfr/text/47/9.3","excerpt":"Enables real-time, two-way voice communications; requires a broadband connection from the user's location; requires internet protocol-compatible customer premises equipment; and 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":"PRIMARY","verifiedAt":"2026-07-17T00:00:00","supportsText":"Definition of interconnected VoIP: real-time two-way voice, broadband connection, IP-compatible customer premises equipment, and calls to and from the PSTN; the category a SIP trunk provider falls under","domain":"law.cornell.edu","publisherName":"Legal Information Institute, Cornell Law School"},{"title":"47 CFR 9.11: E911 service requirements for interconnected VoIP","url":"https://www.law.cornell.edu/cfr/text/47/9.11","excerpt":"Each interconnected VoIP service provider shall specifically advise every subscriber, both new and existing, of the circumstances under which E911 service may not be available... Such circumstances include... broadband connection failure, loss of electrical power.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-17T00:00:00","supportsText":"Interconnected VoIP E911 duties: transmit all 911 calls with call-back number (ANI) and location, and advise every subscriber in plain language of circumstances where E911 may be unavailable, including broadband or power failure and use of a non-native number","domain":"law.cornell.edu","publisherName":"Legal Information Institute, Cornell Law School"},{"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":"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.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-17T00:00:00","supportsText":"Global telecom fraud estimated at $38.95 billion in 2023, up 12 percent from 2021, about 2.5 percent of telecom revenues; PBX fraud listed among the top methods","domain":"cfca.org","publisherName":"Communications Fraud Control Association (CFCA)"},{"title":"SIP Trunking: Efficient Voice Communications","url":"https://www.atlantech.net/blog/sip-trunking-business-phone-service","excerpt":"SIP Trunking replaces the traditional method of the public switched telephone network (PSTN) or PRIs and instead uses a specific packet switched network to send voice traffic to a destination.","quoteText":null,"sourceRole":"SUPPORTING","verifiedAt":"2026-07-17T00:00:00","supportsText":"Plain-language corroboration that SIP trunking replaces the PSTN or PRIs and sends voice over the internet through an IP-enabled PBX; corroborated by [1] and [4]","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":"c1c7b4b5-5d54-4723-8305-65e598b2bcbb","slug":"what-equipment-do-i-need-to-use-sip-trunking","question":"What equipment do I need to use SIP trunking?","publishedAt":"2026-07-18T15:16:57.319","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 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.","url":"/q/what-equipment-do-i-need-to-use-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"}],"contributorStats":{"verifiedAnswers":224,"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","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":"SIP trunking is a way to deliver phone service over an internet connection instead of over dedicated telephone lines, using the Session Initiation Protocol to set up and tear down calls between a business phone system and a provider that connects to the public telephone network.[1][2] The trunk itself is virtual: an internet telephony service provider carries your calls in software, so a company with an IP-PBX no longer needs the physical T1 or PRI circuits that older systems required.[1][6] SIP handles the call signaling while the voice travels separately as RTP media packets, and a session border controller usually sits at the network edge to secure and normalize the traffic.[2][3][5] Because capacity is measured in concurrent call paths rather than physical lines, you can add or remove channels without installing hardware, and a service that reaches the regular phone network falls under the FCC's interconnected VoIP rules for 911 and caller location.[6][10][11]","url":"https://www.answerstack.io/q/what-is-sip-trunking","upvoteCount":0,"datePublished":"2026-07-18T14:40:06.195","dateModified":"2026-07-17T00:00:00","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":"SIP Trunking: Architecture, Security & SBC Guide for Service Providers","url":"https://telcobridges.com/learning/sip-trunking/"},{"@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":"SIP Trunking vs. PRI: Differences, Benefits, & What To Use","url":"https://www.bandwidth.com/blog/pri_vs_sip/"},{"@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":"G.711 vs G.729: VoIP Codec Comparison","url":"https://telcobridges.com/learning/sip-trunking/g711-vs-g729/"},{"@type":"CreativeWork","name":"How Much Does a SIP Trunk Cost? 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