A SIP trunk connects your phone system to a carrier over an IP network in place of telephone circuits. Each call has two parts. The PBX and the carrier exchange short SIP messages that set the call up, and the audio travels as a separate stream of packets until one side hangs up. The carrier passes the call on to whichever network serves the number you dialed.
\n","summary":"A SIP trunk links your PBX to a carrier over an IP network. SIP messages such as INVITE, 180 Ringing, 200 OK, ACK and BYE set up and end each call, and the audio travels separately as RTP packets. The carrier passes calls to and from other networks, and call capacity is a limit rather than a count of physical lines.","publishedAt":"2026-07-18T15:01:28.862","verifiedAt":"2026-07-18T00:00:00","editorialStatus":"APPROVED","lastReviewedAt":"2026-10-10T03:20:20.003","nextReviewDueAt":"2026-10-18T00:00:00","templateVersion":"v2","aliases":["How does a SIP trunk work?","How do SIP trunk calls work?","SIP trunking call flow explained","How does SIP trunking route calls?","What happens during a SIP call?","How does a SIP trunk connect to the PSTN?","How do SIP signaling and RTP media work together?","How does SIP trunk authentication work?","How does an IP-PBX use a SIP trunk?","SIP trunk registration vs IP authentication","How does the SIP INVITE and SDP negotiation work?","How does SIP trunking work step by step"],"confidenceScore":92,"confidenceLabel":"High","canonicalUrl":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"contributorOrganizationProfile":null,"contributorPerson":null,"sections":[{"id":"09158c3d-72be-4c45-bb6e-9f97fc787a9d","sectionKey":"two_ends","sectionType":"markdown_section","heading":"The two ends of the trunk","introMarkdown":"One end of the trunk is your PBX. The other is the carrier's voice network. Atlantech Online, for example, [delivers SIP trunks](https://www.atlantech.net/blog/sip-trunking-business-phone-service) to customers that already run their own VoIP PBX, and offers a hosted PBX to customers who would sooner not run one. With a trunk, the PBX keeps its extensions, voicemail and call routing. The carrier supplies the phone numbers and the route to everyone outside your phone system.\n\nThe trunk between them is an arrangement more than a cable. Each side is configured to reach the other and with rules for which messages to trust. No wire is set aside for each call. A trunk's capacity is counted in simultaneous calls, and our answer on [what a SIP trunk channel is](https://www.answerstack.io/q/what-is-a-sip-trunk-channel) explains that unit.","introHtml":"One end of the trunk is your PBX. The other is the carrier's voice network. Atlantech Online, for example, delivers SIP trunks to customers that already run their own VoIP PBX, and offers a hosted PBX to customers who would sooner not run one. With a trunk, the PBX keeps its extensions, voicemail and call routing. The carrier supplies the phone numbers and the route to everyone outside your phone system.
\nThe trunk between them is an arrangement more than a cable. Each side is configured to reach the other and with rules for which messages to trust. No wire is set aside for each call. A trunk's capacity is counted in simultaneous calls, and our answer on what a SIP trunk channel is explains that unit.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":10},{"id":"b02d6c54-df7f-4e14-88b3-266891a15c1e","sectionKey":"outbound_call","sectionType":"markdown_section","heading":"One outbound call, message by message","introMarkdown":"Suppose someone at extension 214 dials a supplier's mobile number. The PBX sees that the number belongs to no extension of its own and sends the carrier a SIP message called an INVITE. The SIP standard, [RFC 3261](https://www.rfc-editor.org/rfc/rfc3261.html), walks through a tutorial call with this pattern.\n\nThe INVITE says who is calling and which number they want. It also carries a session description, attached the way a document is attached to an email. That description lists the audio formats the PBX can use and the address where the PBX wants to receive audio.\n\nThe carrier replies with 100 Trying, which means the request arrived and is being routed. It works out which network serves the supplier's number and passes the call toward it. This happens during setup. The mobile network is involved before the call is answered.\n\nNext a 180 Ringing message comes back. It reports that the far end is alerting the supplier, and the caller at extension 214 hears ringback. When the supplier answers, a 200 OK arrives. It carries the other side's session description, with the audio format chosen and the address where audio should be sent. The PBX confirms with an ACK. RFC 3261 calls this exchange of two descriptions \"a simple offer/answer model.\" Once it completes, both sides know what kind of audio to send and where to send it.\n\nOther outcomes replace parts of this sequence. A busy or declined call returns an error where the 200 OK would have been, and some carriers ask the PBX to prove its identity before they accept the INVITE.","introHtml":"Suppose someone at extension 214 dials a supplier's mobile number. The PBX sees that the number belongs to no extension of its own and sends the carrier a SIP message called an INVITE. The SIP standard, RFC 3261, walks through a tutorial call with this pattern.
\nThe INVITE says who is calling and which number they want. It also carries a session description, attached the way a document is attached to an email. That description lists the audio formats the PBX can use and the address where the PBX wants to receive audio.
\nThe carrier replies with 100 Trying, which means the request arrived and is being routed. It works out which network serves the supplier's number and passes the call toward it. This happens during setup. The mobile network is involved before the call is answered.
\nNext a 180 Ringing message comes back. It reports that the far end is alerting the supplier, and the caller at extension 214 hears ringback. When the supplier answers, a 200 OK arrives. It carries the other side's session description, with the audio format chosen and the address where audio should be sent. The PBX confirms with an ACK. RFC 3261 calls this exchange of two descriptions "a simple offer/answer model." Once it completes, both sides know what kind of audio to send and where to send it.
\nOther outcomes replace parts of this sequence. A busy or declined call returns an error where the 200 OK would have been, and some carriers ask the PBX to prove its identity before they accept the INVITE.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":20},{"id":"f81f861e-29d6-429d-a6e0-44450b9a5047","sectionKey":"call_messages","sectionType":"table_section","heading":"The messages in that call","introMarkdown":null,"introHtml":null,"outroMarkdown":null,"outroHtml":null,"contentJson":{"rows":[{"cells":["INVITE","PBX","Requests the call and describes the audio the PBX can send and receive"]},{"cells":["100 Trying","Carrier","The request arrived and is being routed"]},{"cells":["180 Ringing","Carrier","The far end is alerting the person called"]},{"cells":["200 OK","Carrier","The call was answered, with the other side's audio description"]},{"cells":["ACK","PBX","Confirms the answer and completes setup"]},{"cells":["BYE","Whichever side hangs up","Ends the call, and the other side replies 200 OK"]}],"columns":["Message","Sent by","What it means"]},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":30},{"id":"6b5c818a-0d91-405f-a19f-517349e14dc2","sectionKey":"audio_path","sectionType":"markdown_section","heading":"How the audio travels","introMarkdown":"SIP creates the session, can change it mid-call and ends it. It does not carry the voices. Those travel in the Real-time Transport Protocol, or RTP, defined in [RFC 3550](https://datatracker.ietf.org/doc/html/rfc3550). Each side chops its audio into small packets and sends them to the address the other side gave during setup.\n\nRTP assumes an imperfect network. The RFC points out that packet networks sometimes lose packets, reorder them and delay them by varying amounts. So each RTP packet carries a sequence number and a timestamp. The receiver uses them to put the audio back in order and play it at the right pace. That is as far as the protocol goes. RTP \"does not guarantee delivery or prevent out-of-order delivery,\" so it cannot make up for a connection that drops too much. Call quality depends on that connection, and also on the equipment at each end and how it is configured. A companion protocol, RTCP, reports how well the audio is arriving. We cover the network side in the connection and bandwidth SIP trunking needs.\n\nThe audio can also take its own route. In RFC 3261's tutorial call, \"the end-to-end media packets take a different path from the SIP signaling messages.\" This helps explain a fault that puzzles people the first time they meet it. A call rings, connects and shows as active, yet one person hears nothing. The SIP messages got through, so one possible fault is in how the audio was addressed or delivered. The packets may have been sent to an address they could not reach, or a firewall may have blocked them on the way. Settings on the equipment at either end can matter too. Signaling and audio are checked separately for that reason.\n\nWhen either person hangs up, that side sends a BYE and the other answers 200 OK. Both stop sending audio.","introHtml":"SIP creates the session, can change it mid-call and ends it. It does not carry the voices. Those travel in the Real-time Transport Protocol, or RTP, defined in RFC 3550. Each side chops its audio into small packets and sends them to the address the other side gave during setup.
\nRTP assumes an imperfect network. The RFC points out that packet networks sometimes lose packets, reorder them and delay them by varying amounts. So each RTP packet carries a sequence number and a timestamp. The receiver uses them to put the audio back in order and play it at the right pace. That is as far as the protocol goes. RTP "does not guarantee delivery or prevent out-of-order delivery," so it cannot make up for a connection that drops too much. Call quality depends on that connection, and also on the equipment at each end and how it is configured. A companion protocol, RTCP, reports how well the audio is arriving. We cover the network side in the connection and bandwidth SIP trunking needs.
\nThe audio can also take its own route. In RFC 3261's tutorial call, "the end-to-end media packets take a different path from the SIP signaling messages." This helps explain a fault that puzzles people the first time they meet it. A call rings, connects and shows as active, yet one person hears nothing. The SIP messages got through, so one possible fault is in how the audio was addressed or delivered. The packets may have been sent to an address they could not reach, or a firewall may have blocked them on the way. Settings on the equipment at either end can matter too. Signaling and audio are checked separately for that reason.
\nWhen either person hangs up, that side sends a BYE and the other answers 200 OK. Both stop sending audio.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":40},{"id":"4b52f7d8-e1b1-424d-8ef6-8745ea2b68a7","sectionKey":"inbound_calls","sectionType":"markdown_section","heading":"Inbound calls run the other way","introMarkdown":"When someone dials one of your numbers, their network delivers the call to your carrier, because your carrier holds that number. The carrier then sends the INVITE to your PBX. The PBX rings an extension or plays the auto attendant, and the same alerting, answer and audio steps follow.\n\nFor this to work, the carrier has to know where your PBX is. One way is a fixed address agreed when the trunk is set up. Another is registration. RFC 3261 describes a device sending REGISTER messages when it starts and at regular intervals afterward, which tells a server where the device can currently be reached. The RFC presents registration as one of the ways a server can learn this. Which method your trunk uses is part of the carrier's setup.","introHtml":"When someone dials one of your numbers, their network delivers the call to your carrier, because your carrier holds that number. The carrier then sends the INVITE to your PBX. The PBX rings an extension or plays the auto attendant, and the same alerting, answer and audio steps follow.
\nFor this to work, the carrier has to know where your PBX is. One way is a fixed address agreed when the trunk is set up. Another is registration. RFC 3261 describes a device sending REGISTER messages when it starts and at regular intervals afterward, which tells a server where the device can currently be reached. The RFC presents registration as one of the ways a server can learn this. Which method your trunk uses is part of the carrier's setup.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":50},{"id":"c1781116-0ddb-4b90-aa3e-bc2c89bb68b8","sectionKey":"session_border_controller","sectionType":"markdown_section","heading":"Where a session border controller fits","introMarkdown":"A trunk may have a session border controller, or SBC, somewhere between the PBX and the carrier's network. [RFC 5853](https://www.rfc-editor.org/rfc/rfc5853.html) describes the functions these devices can perform. An SBC can repair SIP messages from equipment that does not follow the standard closely. It can keep a connection alive to a PBX that sits behind a firewall's address translation. It can also rewrite the session description so the audio passes through a relay the operator controls. Which of these a given SBC performs depends on how it is deployed.\n\nAn SBC is not part of every trunk on the customer's side. A carrier may run its own, and a PBX on a private circuit to the carrier may connect without one. Our answer on the equipment SIP trunking needs covers when you would own one.","introHtml":"A trunk may have a session border controller, or SBC, somewhere between the PBX and the carrier's network. RFC 5853 describes the functions these devices can perform. An SBC can repair SIP messages from equipment that does not follow the standard closely. It can keep a connection alive to a PBX that sits behind a firewall's address translation. It can also rewrite the session description so the audio passes through a relay the operator controls. Which of these a given SBC performs depends on how it is deployed.
\nAn SBC is not part of every trunk on the customer's side. A carrier may run its own, and a PBX on a private circuit to the carrier may connect without one. Our answer on the equipment SIP trunking needs covers when you would own one.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":60},{"id":"15727044-0f5e-4e0b-a9af-b87255f39c02","sectionKey":"call_capacity","sectionType":"markdown_section","heading":"How a trunk's call capacity is set","introMarkdown":"Nothing in this exchange reserves a circuit. RFC 3550 says RTP \"does not address resource reservation,\" and SIP manages sessions without setting aside network capacity for them. So the carrier's limit on simultaneous calls is a setting. It is not a count of physical lines.\n\nThat gives a trunk room to flex. Atlantech's 2023 post mentions a local government customer that adds capacity on its SIP trunks during weather events such as snowstorms and tropical storms. The carrier's limit is one of several ceilings. The PBX, any SBC or gateway and the data connection each have a maximum of their own, and extra calls need room under all of them. Our answer on [how many simultaneous calls you need](https://www.answerstack.io/q/how-many-simultaneous-calls-do-i-need-or-get-on-a-sip-trunk) covers how to choose the number.\n\nThe trunk only defines the link between your PBX and your carrier. Beyond the carrier, the call may finish on a mobile network, a copper landline or another IP phone system, and our answer on [the difference between PSTN and VoIP](https://www.answerstack.io/q/what-is-the-difference-between-pstn-and-voip) compares those networks. In the call above, the request had reached the supplier's mobile network before the supplier's phone was alerted.","introHtml":"Nothing in this exchange reserves a circuit. RFC 3550 says RTP "does not address resource reservation," and SIP manages sessions without setting aside network capacity for them. So the carrier's limit on simultaneous calls is a setting. It is not a count of physical lines.
\nThat gives a trunk room to flex. Atlantech's 2023 post mentions a local government customer that adds capacity on its SIP trunks during weather events such as snowstorms and tropical storms. The carrier's limit is one of several ceilings. The PBX, any SBC or gateway and the data connection each have a maximum of their own, and extra calls need room under all of them. Our answer on how many simultaneous calls you need covers how to choose the number.
\nThe trunk only defines the link between your PBX and your carrier. Beyond the carrier, the call may finish on a mobile network, a copper landline or another IP phone system, and our answer on the difference between PSTN and VoIP compares those networks. In the call above, the request had reached the supplier's mobile network before the supplier's phone was alerted.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":70}],"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":"2026-07-18T00:00:00","supportsText":"SIP trunking as a VoIP service based on SIP by which ITSPs deliver telephone service to IP-PBX customers and connect them to the PSTN; virtual trunk sized in concurrent calls; distinction from hosted systems and from VoIP in general","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 that creates, modifies, and terminates sessions including Internet telephone calls; text-based HTTP-like request/response model; INVITE sets up, ACK confirms, BYE terminates, REGISTER registers","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"RFC 3264: An Offer/Answer Model with the Session Description Protocol (SDP)","url":"https://datatracker.ietf.org/doc/html/rfc3264","excerpt":"the set of media streams and codecs the offerer wishes to use, along with the IP addresses and ports the offerer would like to use to receive the media.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-18T00:00:00","supportsText":"Offer/answer model: one side offers the media streams, codecs, and IP addresses and ports it wants to use, and the other returns an answer accepting a matching set before media flows","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"RFC 3665: Session Initiation Protocol (SIP) Basic Call Flow Examples","url":"https://datatracker.ietf.org/doc/html/rfc3665","excerpt":"Scenarios include SIP Registration and SIP session establishment. Call flow diagrams and message details are shown.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-18T00:00:00","supportsText":"The successful basic call sequence: INVITE, 100 Trying, 180 Ringing, 200 OK, ACK, both-way RTP media, then BYE and 200 OK to end; includes registration scenarios","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"RFC 4566: SDP: Session Description Protocol","url":"https://datatracker.ietf.org/doc/html/rfc4566","excerpt":"SDP is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-18T00:00:00","supportsText":"SDP describes a multimedia session for session invitation, including media type, the codec/format of the media, and the transport port and connection address for the media","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 audio of a SIP call separate from signaling, with sequence numbers and timestamps that let the receiver reconstruct the packet order; RTCP monitors quality of service","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"RFC 3263: Session Initiation Protocol (SIP): Locating SIP Servers","url":"https://datatracker.ietf.org/doc/html/rfc3263","excerpt":"The Session Initiation Protocol (SIP) uses DNS procedures to allow a client to resolve a SIP Uniform Resource Identifier (URI) into the IP address, port, and transport protocol of the next hop to contact.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-18T00:00:00","supportsText":"SIP uses DNS procedures (NAPTR, then SRV, then A/AAAA lookups) to resolve a SIP URI into the IP address, port, and transport protocol of the next hop, which is how a trunk locates the provider's servers","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"RFC 5853: Requirements from Session Initiation Protocol (SIP) Session Border Control (SBC) Deployments","url":"https://datatracker.ietf.org/doc/html/rfc5853","excerpt":"These functions include: 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":"SBCs are by definition back-to-back user agents that sit at the network border; their functions group into perimeter defense (access control, topology hiding, denial-of-service prevention), functionality not in endpoints (NAT traversal, protocol interworking or repair), and traffic management (media","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals","url":"https://datatracker.ietf.org/doc/html/rfc4733","excerpt":"This memo describes how to carry dual-tone multifrequency (DTMF) signalling, other tone signals, and telephony events in RTP packets. It obsoletes RFC 2833.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-18T00:00:00","supportsText":"DTMF keypad digits and other telephony events are carried as named RTP telephone events rather than as tones inside the compressed audio, keeping them accurate across the media path","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"SIP Registration & Authentication Guide","url":"https://connection-technologies.co.uk/help/voip-sip-trunking/sip-registration-and-authentication","excerpt":"The provider configures their system to accept SIP traffic from your office's static public IP address... the digest authentication mechanism (similar to HTTP digest auth) creates a hash that proves knowledge of the password... The password is never sent over the network in plain text.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"Two SIP trunk authentication methods: IP-based authentication where the provider accepts SIP traffic only from your static public IP address, and digest authentication where a challenge is answered with a hash of the password that is never sent in plain text","domain":"connection-technologies.co.uk","publisherName":"Connection Technologies"},{"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":"2026-07-18T00:00:00","supportsText":"Per-call media bandwidth including RTP/UDP/IP/Ethernet headers: G.711 about 85 to 87 kbps per direction, G.729 about 31 kbps per direction at 20 ms packetization","domain":"telcobridges.com","publisherName":"TelcoBridges"},{"title":"47 CFR 9.3, Definitions: Interconnected VoIP service","url":"https://www.law.cornell.edu/cfr/text/47/9.3","excerpt":"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":"2026-07-18T00:00:00","supportsText":"Definition of interconnected VoIP: real-time two-way voice, a broadband connection, IP-compatible customer premises equipment, and the ability to receive calls from and terminate calls to 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":"broadband connection failure, loss of electrical power, and delays that may occur in making a dispatchable location available.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-18T00:00:00","supportsText":"Interconnected VoIP E911 duties: transmit all 911 calls with a call-back number and location, and advise subscribers in plain language of circumstances where E911 may be unavailable, including broadband connection failure and loss of electrical power","domain":"law.cornell.edu","publisherName":"Legal Information Institute, Cornell Law School"},{"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":"2026-07-18T00:00:00","supportsText":"Plain-language corroboration that a SIP trunk replaces the PSTN or PRIs and sends voice traffic over a packet-switched network rather than dedicated circuits; corroborated by [1] and [2]","domain":"atlantech.net","publisherName":"Atlantech Online"}],"revisions":[{"revisedAt":"2026-07-18T15:01:28.862+00:00","versionLabel":"Originally published","note":"","reviewedBy":null}],"relatedAnswers":[{"id":"77b3581e-fb3d-43da-aefa-a3ebd533d57d","slug":"how-many-simultaneous-calls-do-i-need-or-get-on-a-sip-trunk","question":"How many simultaneous calls do I need or get on a SIP trunk?","publishedAt":"2026-07-18T15:16:53.422","confidenceScore":82,"confidenceLabel":"High","industry":{"id":"ff619d7c-d7d7-485e-a05a-53fba07f33ed","slug":"telecommunications","label":"Telecommunications","description":"Business voice, fiber, UCaaS, and network services"},"topic":{"slug":"sip-trunking","label":"SIP Trunking","description":"SIP trunking for business voice: how it works, channels, bandwidth, and PBX connectivity.","schemaKind":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"snippet":"To size a SIP trunk, measure the calls in your busiest hour, convert that traffic to call paths with a blocking target, then check the result against the provider's plan, your equipment and your connection. What you get is the lowest of those limits.","url":"/q/how-many-simultaneous-calls-do-i-need-or-get-on-a-sip-trunk"},{"id":"bc3ff0a8-b843-46c0-a953-41aa557e1110","slug":"what-is-the-difference-between-sip-trunking-and-cloud-pbx","question":"What is the difference between SIP trunking and cloud PBX?","publishedAt":"2026-07-18T15:01:34.706","confidenceScore":84,"confidenceLabel":"High","industry":{"id":"ff619d7c-d7d7-485e-a05a-53fba07f33ed","slug":"telecommunications","label":"Telecommunications","description":"Business voice, fiber, UCaaS, and network services"},"topic":{"slug":"sip-trunking","label":"SIP Trunking","description":"SIP trunking for business voice: how it works, channels, bandwidth, and PBX connectivity.","schemaKind":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"snippet":"SIP trunking connects a phone system you run to the public telephone network. A cloud PBX is the phone system itself, run by a provider and billed per user.","url":"/q/what-is-the-difference-between-sip-trunking-and-cloud-pbx"},{"id":"97ab21fb-1842-45fa-90d8-e34e67a64794","slug":"what-is-the-difference-between-sip-trunking-and-voip","question":"What is the difference between SIP trunking and VoIP?","publishedAt":"2026-07-18T15:01:30.724","confidenceScore":86,"confidenceLabel":"High","industry":{"id":"ff619d7c-d7d7-485e-a05a-53fba07f33ed","slug":"telecommunications","label":"Telecommunications","description":"Business voice, fiber, UCaaS, and network services"},"topic":{"slug":"sip-trunking","label":"SIP Trunking","description":"SIP trunking for business voice: how it works, channels, bandwidth, and PBX connectivity.","schemaKind":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"snippet":"VoIP is a technology category: voice carried over data networks with internet protocols. SIP trunking is one carrier service built on it, connecting a phone system you run to the public telephone network. A SIP trunk is VoIP. The real choice appears when VoIP means a hosted phone service, and it turns on who runs the phone system.","url":"/q/what-is-the-difference-between-sip-trunking-and-voip"},{"id":"fad66bd0-23bc-421f-944d-f514e3d12488","slug":"what-is-sip-trunking","question":"What is SIP trunking?","publishedAt":"2026-07-18T14:40:06.195","confidenceScore":86,"confidenceLabel":"High","industry":{"id":"ff619d7c-d7d7-485e-a05a-53fba07f33ed","slug":"telecommunications","label":"Telecommunications","description":"Business voice, fiber, UCaaS, and network services"},"topic":{"slug":"sip-trunking","label":"SIP Trunking","description":"SIP trunking for business voice: how it works, channels, bandwidth, and PBX connectivity.","schemaKind":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"snippet":"SIP trunking is a carrier service that connects a business phone system to the public telephone network over an IP connection. The PBX keeps handling extensions and routing, SIP messages set calls up, and RTP carries the audio. Phone numbers and call capacity are counted separately from the trunk.","url":"/q/what-is-sip-trunking"}],"contributorStats":{"verifiedAnswers":149,"openDisputes":0},"schemaJson":{"@context":"https://schema.org","@type":"Question","name":"How does SIP trunking work?","text":"How does SIP trunking work?","url":"https://www.answerstack.io/q/how-does-sip-trunking-work","answerCount":1,"datePublished":"2026-07-18T15:01:28.862","about":[{"@type":"Thing","name":"SIP Trunking"},{"@type":"Thing","name":"Telecommunications"}],"acceptedAnswer":{"@type":"Answer","text":"A SIP trunk connects your phone system to a carrier over an IP network in place of telephone circuits. Each call has two parts. The PBX and the carrier exchange short SIP messages that set the call up, and the audio travels as a separate stream of packets until one side hangs up. The carrier passes the call on to whichever network serves the number you dialed.","url":"https://www.answerstack.io/q/how-does-sip-trunking-work","upvoteCount":0,"datePublished":"2026-07-18T15:01:28.862","dateModified":"2026-07-18T00:00:00","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 3264: An Offer/Answer Model with the Session Description Protocol (SDP)","url":"https://datatracker.ietf.org/doc/html/rfc3264"},{"@type":"CreativeWork","name":"RFC 3665: Session Initiation Protocol (SIP) Basic Call Flow Examples","url":"https://datatracker.ietf.org/doc/html/rfc3665"},{"@type":"CreativeWork","name":"RFC 4566: SDP: Session Description Protocol","url":"https://datatracker.ietf.org/doc/html/rfc4566"},{"@type":"CreativeWork","name":"RFC 3550: RTP: A Transport Protocol for Real-Time Applications","url":"https://datatracker.ietf.org/doc/html/rfc3550"},{"@type":"CreativeWork","name":"RFC 3263: Session Initiation Protocol (SIP): Locating SIP Servers","url":"https://datatracker.ietf.org/doc/html/rfc3263"},{"@type":"CreativeWork","name":"RFC 5853: Requirements from Session Initiation Protocol (SIP) Session Border Control (SBC) Deployments","url":"https://datatracker.ietf.org/doc/html/rfc5853"},{"@type":"CreativeWork","name":"RFC 4733: RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals","url":"https://datatracker.ietf.org/doc/html/rfc4733"},{"@type":"CreativeWork","name":"SIP Registration & Authentication Guide","url":"https://connection-technologies.co.uk/help/voip-sip-trunking/sip-registration-and-authentication"},{"@type":"CreativeWork","name":"G.711 vs G.729: VoIP Codec Comparison","url":"https://telcobridges.com/learning/sip-trunking/g711-vs-g729/"},{"@type":"CreativeWork","name":"47 CFR 9.3, Definitions: Interconnected VoIP service","url":"https://www.law.cornell.edu/cfr/text/47/9.3"},{"@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: Efficient Voice Communications","url":"https://www.atlantech.net/blog/sip-trunking-business-phone-service"}]}}}