For SIP trunking you need a broadband internet connection sized to your busy-hour concurrent calls rather than to your total number of phones, with roughly 85 to 90 kilobits per second in each direction for every simultaneous call using the common G.711 codec, or about 31 kilobits per second per call with the more compressed G.729a.[1][2][6] Multiply your peak concurrent calls by the per-call figure and add 10 to 20 percent of headroom for signaling and other traffic to get the floor for the connection.[7][8] Raw speed is only half of it, because the connection also has to hold one-way latency under about 150 milliseconds, jitter under about 30 milliseconds, and packet loss under roughly 1 percent, and voice degrades on delay and loss well before it runs out of raw bandwidth.[5][9] Upload capacity is usually the limiting factor on asymmetric consumer lines, so a symmetric business connection with quality-of-service prioritization, often fiber, is the typical recommendation.[9][11]
\n","summary":"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.","publishedAt":"2026-07-18T15:16:59.312","verifiedAt":"2026-07-18T00:00:00","editorialStatus":"APPROVED","lastReviewedAt":"2026-07-18T00:00:00","nextReviewDueAt":"2026-10-18T00:00:00","templateVersion":"v2","aliases":["How much bandwidth do I need for SIP trunking?","SIP trunk bandwidth requirements","What internet speed is needed for SIP trunking?","SIP trunking bandwidth per call","How much internet does a SIP trunk use?","Bandwidth calculation for SIP trunks","Does SIP trunking need fiber internet?","What kind of internet connection do I need for VoIP SIP trunking?","G.711 vs G.729 bandwidth for SIP trunking","How to size internet for SIP trunk calls","Latency and jitter requirements for SIP trunking","Upload bandwidth for SIP trunk calls"],"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":"7e59dcb7-32dc-45e0-b42e-b12d82ec45f0","sectionKey":"how_much_internet","sectionType":"markdown_section","heading":"How much internet do you need for SIP trunking?","introMarkdown":"The size of the connection follows one number: how many calls you run at the same time during your busiest hour, not how many phones or numbers you own.[1] Each simultaneous call needs roughly 85 to 90 kilobits per second in each direction with the common G.711 codec, or about 31 kilobits per second with the more compressed G.729a, once the IP, UDP, and RTP headers are counted on top of the raw audio.[2][6] Multiply your peak concurrent calls by that per-call figure, add 10 to 20 percent of headroom for signaling and the data traffic sharing the link, and you have a floor for the connection you need.[7][8]\n\n### Start from concurrent calls, not from headsets\n\nA business with fifty employees almost never has fifty calls in progress at once, so sizing the link to fifty calls wastes money.[1] The useful measure is the number of calls active at your busiest hour, which providers call peak concurrent call paths.[1] Count that number, size the bandwidth to it, and add a margin for growth, because a trunk lets you raise or lower channels in software without new wiring once the underlying connection has room.\n\n### Bandwidth alone does not decide call quality\n\nSpeed is only half of the requirement. Voice is sensitive to delay, timing variation, and lost packets, so a connection with plenty of megabits can still produce choppy or clipped audio if those measures drift.[5][9] A SIP-ready connection has to hold one-way latency under about 150 milliseconds, jitter under about 30 milliseconds, and packet loss under roughly 1 percent, and on most consumer lines the upload direction runs out first.[5][9] The rest of this answer covers both the sizing math and the quality targets that decide whether calls actually sound clean.","introHtml":"The size of the connection follows one number: how many calls you run at the same time during your busiest hour, not how many phones or numbers you own.[1] Each simultaneous call needs roughly 85 to 90 kilobits per second in each direction with the common G.711 codec, or about 31 kilobits per second with the more compressed G.729a, once the IP, UDP, and RTP headers are counted on top of the raw audio.[2][6] Multiply your peak concurrent calls by that per-call figure, add 10 to 20 percent of headroom for signaling and the data traffic sharing the link, and you have a floor for the connection you need.[7][8]
\nA business with fifty employees almost never has fifty calls in progress at once, so sizing the link to fifty calls wastes money.[1] The useful measure is the number of calls active at your busiest hour, which providers call peak concurrent call paths.[1] Count that number, size the bandwidth to it, and add a margin for growth, because a trunk lets you raise or lower channels in software without new wiring once the underlying connection has room.
\nSpeed is only half of the requirement. Voice is sensitive to delay, timing variation, and lost packets, so a connection with plenty of megabits can still produce choppy or clipped audio if those measures drift.[5][9] A SIP-ready connection has to hold one-way latency under about 150 milliseconds, jitter under about 30 milliseconds, and packet loss under roughly 1 percent, and on most consumer lines the upload direction runs out first.[5][9] The rest of this answer covers both the sizing math and the quality targets that decide whether calls actually sound clean.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":0},{"id":"7bfbd58b-b811-4c16-b74b-dacce9681ad6","sectionKey":"codec_bandwidth_table","sectionType":"table_section","heading":"How much bandwidth does each call use, by codec?","introMarkdown":"The codec your system uses sets the per-call bandwidth, and the two you will meet most often are G.711 and G.729a.[2] G.711 sends uncompressed audio at a 64 kilobit payload, which with header overhead comes to roughly 85 to 90 kilobits per second in each direction and sounds close to a landline, scoring about 4.2 on the mean opinion score scale.[2][3] G.729a compresses the same speech to an 8 kilobit payload, about 31 kilobits per second on the wire, at a small cost in fidelity that most callers do not notice, scoring about 3.9.[2] The payload compresses eight to one, but because every packet still carries the same 40 bytes of IP, UDP, and RTP headers fifty times a second, the real saving on the wire is closer to three to one.[2][4] The table compares the options a business usually weighs.","introHtml":"The codec your system uses sets the per-call bandwidth, and the two you will meet most often are G.711 and G.729a.[2] G.711 sends uncompressed audio at a 64 kilobit payload, which with header overhead comes to roughly 85 to 90 kilobits per second in each direction and sounds close to a landline, scoring about 4.2 on the mean opinion score scale.[2][3] G.729a compresses the same speech to an 8 kilobit payload, about 31 kilobits per second on the wire, at a small cost in fidelity that most callers do not notice, scoring about 3.9.[2] The payload compresses eight to one, but because every packet still carries the same 40 bytes of IP, UDP, and RTP headers fifty times a second, the real saving on the wire is closer to three to one.[2][4] The table compares the options a business usually weighs.
\n","outroMarkdown":"Most providers default to G.711 on a dedicated or fiber connection because it matches landline clarity, and reserve G.729a for sites where the upstream link is the constraint.[6][7] Mixing is common, since the codec is negotiated per call: a headquarters on fiber can run G.711 while a branch on a thin connection uses G.729a.[2]","outroHtml":"Most providers default to G.711 on a dedicated or fiber connection because it matches landline clarity, and reserve G.729a for sites where the upstream link is the constraint.[6][7] Mixing is common, since the codec is negotiated per call: a headquarters on fiber can run G.711 while a branch on a thin connection uses G.729a.[2]
\n","contentJson":{"rows":[{"cells":["G.711","64 kbps, uncompressed","About 85 to 90 kbps [2][6]","About 4.2, near landline [2]","Default where bandwidth is ample [7]"]},{"cells":["G.729a","8 kbps, compressed","About 31 to 32 kbps [1][2]","About 3.9, slightly compressed [2]","Where upstream bandwidth is tight [1]"]},{"cells":["G.722 (HD voice)","48 to 64 kbps, wideband [6]","About 85 to 95 kbps [6]","Wideband, clearer than G.711","Where both ends support HD and bandwidth allows"]}],"columns":["Codec","Audio payload","Per call, each direction (with overhead)","Voice quality (MOS)","Where it fits"]},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":1},{"id":"39213b17-6e75-44d3-8b91-625c8343ae48","sectionKey":"connection_requirements_table","sectionType":"table_section","heading":"What does a SIP-ready connection have to deliver?","introMarkdown":"A connection that carries voice well meets five targets at once, and missing any one of them shows up as audible trouble even when the others look fine.[5][9] Bandwidth sets how many calls fit, while latency, jitter, and packet loss decide whether each call sounds clean, and quality-of-service settings protect voice when the link gets busy.[9] The table lists the five with the commonly cited targets, and the sections after it explain each one and what to do about it.","introHtml":"A connection that carries voice well meets five targets at once, and missing any one of them shows up as audible trouble even when the others look fine.[5][9] Bandwidth sets how many calls fit, while latency, jitter, and packet loss decide whether each call sounds clean, and quality-of-service settings protect voice when the link gets busy.[9] The table lists the five with the commonly cited targets, and the sections after it explain each one and what to do about it.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{"rows":[{"cells":["Bandwidth per call","About 85 to 90 kbps each way on G.711 [2][6]","Calls fail to connect or drop at peak [1]"]},{"cells":["One-way latency","Under about 150 ms [5]","Talk-over and awkward pauses [5]"]},{"cells":["Jitter","Under about 30 ms [9]","Choppy or robotic audio [9]"]},{"cells":["Packet loss","Under about 1 percent [9]","Missing words and gaps [9]"]},{"cells":["QoS prioritization","Voice marked and served first [9]","Data traffic starves calls at peak [9]"]}],"columns":["Requirement","Common target","What happens if you miss it"]},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":2},{"id":"fe638ffd-fc64-4bee-aa13-9f5e6047b5dd","sectionKey":"bandwidth_both_directions","sectionType":"markdown_section","heading":"Enough bandwidth in both directions","introMarkdown":"A SIP trunk needs its per-call bandwidth available in each direction at the same time, and on most connections the upload path is the one that runs out first.[9] A phone call is symmetric, since both people talk, so a call using G.711 consumes about 85 to 90 kilobits per second upstream and the same downstream.[2][9] Consumer and many business plans are asymmetric, advertising a large download figure and a much smaller upload, so the usable ceiling for concurrent calls is set by the smaller upload number.[8][9] A 50 by 10 megabit line, for instance, has room for roughly 100 G.711 calls on the upstream side before headroom, well short of what the download figure alone would suggest.[9] When you run a speed test to plan capacity, use the upload result, convert it to kilobits by multiplying the megabit figure by a thousand, then divide by your per-call bandwidth to see how many calls the link can hold.[8] Symmetric connections, where upload matches download, make SIP capacity planning far more predictable, which is part of why business fiber is the usual recommendation.[9][11]","introHtml":"A SIP trunk needs its per-call bandwidth available in each direction at the same time, and on most connections the upload path is the one that runs out first.[9] A phone call is symmetric, since both people talk, so a call using G.711 consumes about 85 to 90 kilobits per second upstream and the same downstream.[2][9] Consumer and many business plans are asymmetric, advertising a large download figure and a much smaller upload, so the usable ceiling for concurrent calls is set by the smaller upload number.[8][9] A 50 by 10 megabit line, for instance, has room for roughly 100 G.711 calls on the upstream side before headroom, well short of what the download figure alone would suggest.[9] When you run a speed test to plan capacity, use the upload result, convert it to kilobits by multiplying the megabit figure by a thousand, then divide by your per-call bandwidth to see how many calls the link can hold.[8] Symmetric connections, where upload matches download, make SIP capacity planning far more predictable, which is part of why business fiber is the usual recommendation.[9][11]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":3},{"id":"224e9d26-b9fd-40df-a683-b64e8f97ebe5","sectionKey":"latency","sectionType":"markdown_section","heading":"Latency under roughly 150 milliseconds","introMarkdown":"One-way delay on a voice call should stay under about 150 milliseconds, the threshold the ITU-T sets in Recommendation G.114 for keeping a conversation feeling natural.[5][9] Below that figure, most callers perceive the exchange as immediate; between 150 and 400 milliseconds the delay becomes noticeable and people begin talking over each other, and past 400 milliseconds interactive conversation breaks down.[5][9] Latency comes from the distance packets travel and every device that handles them along the way, so a provider with points of presence near you and a direct path to the phone network matters as much as the speed of your own link.[9] You cannot buy your way past a latency problem with more megabits, because a congested or geographically distant path adds delay no matter how much raw bandwidth sits idle.[9] Testing round-trip time to your provider before you commit, then halving it for a rough one-way figure, tells you whether the path has the margin voice needs.","introHtml":"One-way delay on a voice call should stay under about 150 milliseconds, the threshold the ITU-T sets in Recommendation G.114 for keeping a conversation feeling natural.[5][9] Below that figure, most callers perceive the exchange as immediate; between 150 and 400 milliseconds the delay becomes noticeable and people begin talking over each other, and past 400 milliseconds interactive conversation breaks down.[5][9] Latency comes from the distance packets travel and every device that handles them along the way, so a provider with points of presence near you and a direct path to the phone network matters as much as the speed of your own link.[9] You cannot buy your way past a latency problem with more megabits, because a congested or geographically distant path adds delay no matter how much raw bandwidth sits idle.[9] Testing round-trip time to your provider before you commit, then halving it for a rough one-way figure, tells you whether the path has the margin voice needs.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":4},{"id":"fad0933e-750e-4f68-ac61-793c7330cedf","sectionKey":"jitter","sectionType":"markdown_section","heading":"Jitter under roughly 30 milliseconds","introMarkdown":"Jitter is the variation in how evenly voice packets arrive, and it should stay under about 30 milliseconds for calls to sound smooth.[9] Voice packets leave in an even stream fifty times a second, but a busy or poorly managed network can bunch them up and space them out, so they reach the far end at uneven intervals.[4][9] Phones and session border controllers absorb some of this with a jitter buffer that holds arriving packets briefly and releases them on a steady beat, yet a buffer only has so much room, and once jitter exceeds it the audio turns choppy or drops syllables.[9] High jitter usually points to congestion or to voice competing with bulk data on the same link rather than to a shortage of bandwidth, which is why prioritizing voice traffic addresses it more reliably than buying a bigger connection.[9] Keeping the link from saturating and marking voice as priority traffic are the two moves that hold jitter down.","introHtml":"Jitter is the variation in how evenly voice packets arrive, and it should stay under about 30 milliseconds for calls to sound smooth.[9] Voice packets leave in an even stream fifty times a second, but a busy or poorly managed network can bunch them up and space them out, so they reach the far end at uneven intervals.[4][9] Phones and session border controllers absorb some of this with a jitter buffer that holds arriving packets briefly and releases them on a steady beat, yet a buffer only has so much room, and once jitter exceeds it the audio turns choppy or drops syllables.[9] High jitter usually points to congestion or to voice competing with bulk data on the same link rather than to a shortage of bandwidth, which is why prioritizing voice traffic addresses it more reliably than buying a bigger connection.[9] Keeping the link from saturating and marking voice as priority traffic are the two moves that hold jitter down.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":5},{"id":"d79fa9b5-059b-49ee-a3ef-12dee43c498e","sectionKey":"packet_loss","sectionType":"markdown_section","heading":"Packet loss under about 1 percent","introMarkdown":"Lost packets should stay under about 1 percent, because voice has no time to wait for a dropped packet to be resent.[9] A file transfer recovers a lost packet by asking for it again, but a live call has already moved on, so the audio for that missing slice is simply gone, heard as a clipped word or a brief gap.[9] Even loss around 1 percent is noticeable on a G.711 call, and more compressed codecs tolerate less before quality falls off, since each of their packets carries more of the conversation.[9] Loss tends to appear when a link saturates and a router discards packets it cannot queue, so it often arrives together with rising jitter during the busy hour.[9] Sizing the connection with headroom above your peak call bandwidth, and giving voice priority so it is not the traffic dropped under load, are what keep loss inside the tolerable range.[8][9]","introHtml":"Lost packets should stay under about 1 percent, because voice has no time to wait for a dropped packet to be resent.[9] A file transfer recovers a lost packet by asking for it again, but a live call has already moved on, so the audio for that missing slice is simply gone, heard as a clipped word or a brief gap.[9] Even loss around 1 percent is noticeable on a G.711 call, and more compressed codecs tolerate less before quality falls off, since each of their packets carries more of the conversation.[9] Loss tends to appear when a link saturates and a router discards packets it cannot queue, so it often arrives together with rising jitter during the busy hour.[9] Sizing the connection with headroom above your peak call bandwidth, and giving voice priority so it is not the traffic dropped under load, are what keep loss inside the tolerable range.[8][9]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":6},{"id":"3235cad0-109b-4112-af9a-dc18e359b9d7","sectionKey":"qos","sectionType":"markdown_section","heading":"Prioritizing voice with quality of service","introMarkdown":"Quality-of-service settings tell your network to move voice packets ahead of everything else, and on a shared connection they are the single most effective protection for call quality.[9] Voice is marked with the Expedited Forwarding class, DSCP value 46, so routers and switches recognize it and serve it before email, backups, and file downloads that can wait a few milliseconds without anyone noticing.[9] Without prioritization, a large upload or a cloud backup can fill the link and push voice packets into delay and loss precisely when calls are busiest.[9] QoS works well when you control the network end to end, which inside your own building you do, though it cannot govern the public internet between you and your provider, and that gap is one reason a dedicated or provider-supplied connection is often recommended for voice.[9][11] On a business fiber connection paired with the SIP provider, voice can be prioritized across the whole path rather than only inside your walls.[10][11]","introHtml":"Quality-of-service settings tell your network to move voice packets ahead of everything else, and on a shared connection they are the single most effective protection for call quality.[9] Voice is marked with the Expedited Forwarding class, DSCP value 46, so routers and switches recognize it and serve it before email, backups, and file downloads that can wait a few milliseconds without anyone noticing.[9] Without prioritization, a large upload or a cloud backup can fill the link and push voice packets into delay and loss precisely when calls are busiest.[9] QoS works well when you control the network end to end, which inside your own building you do, though it cannot govern the public internet between you and your provider, and that gap is one reason a dedicated or provider-supplied connection is often recommended for voice.[9][11] On a business fiber connection paired with the SIP provider, voice can be prioritized across the whole path rather than only inside your walls.[10][11]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":7},{"id":"dc55c0b3-ea7f-42fa-8641-bb969d825c4d","sectionKey":"connection_type","sectionType":"markdown_section","heading":"What kind of connection works best","introMarkdown":"For most businesses a symmetric, business-grade connection with quality-of-service support is the practical answer, and fiber is the common choice because it offers matching upload and download with low latency and jitter.[10][11] A dedicated internet access circuit, where the bandwidth is yours rather than shared across a neighborhood, gives the most predictable voice quality because your calls never compete with other subscribers for capacity.[9] Cable and DSL can carry SIP trunking, and many small offices run it successfully, but their asymmetric upload and shared design make capacity planning less certain, so they suit lower call volumes better than a busy call center.[8][9] Some organizations keep voice on a separate circuit or a separate VLAN from data so the two never contend, and higher-volume sites add a second connection from another provider so calls fail over if the primary link goes dark.[9] The connection you already have may be enough for a handful of concurrent calls, which is why measuring your peak call volume and testing your current upload before ordering anything new is the sensible first step.[8][11]","introHtml":"For most businesses a symmetric, business-grade connection with quality-of-service support is the practical answer, and fiber is the common choice because it offers matching upload and download with low latency and jitter.[10][11] A dedicated internet access circuit, where the bandwidth is yours rather than shared across a neighborhood, gives the most predictable voice quality because your calls never compete with other subscribers for capacity.[9] Cable and DSL can carry SIP trunking, and many small offices run it successfully, but their asymmetric upload and shared design make capacity planning less certain, so they suit lower call volumes better than a busy call center.[8][9] Some organizations keep voice on a separate circuit or a separate VLAN from data so the two never contend, and higher-volume sites add a second connection from another provider so calls fail over if the primary link goes dark.[9] The connection you already have may be enough for a handful of concurrent calls, which is why measuring your peak call volume and testing your current upload before ordering anything new is the sensible first step.[8][11]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":8},{"id":"d1640561-3de5-45e0-9b4b-eb6f30924f87","sectionKey":"contributor_perspective","sectionType":"markdown_section","heading":"How this answer was researched","introMarkdown":"The bandwidth figures here come from a mix of the standards that define the codecs and independent industry guidance on how those codecs behave on a real connection.[2][3][4] The per-call numbers, the calculation method, and the headroom advice are attributed to independent telecom sources rather than presented as fixed law, since exact figures shift with codec settings, packetization, and how much lower-layer overhead a given source counts.[1][6][7][8] The latency target comes from the ITU-T recommendation that sets it, and the jitter and packet-loss targets reflect the range independent network sources consistently cite.[5][9] Voice codecs, provider networks, and pricing change over time, so the specific numbers reflect what the cited sources reported on the verification date. Practitioners who design, provision, or troubleshoot SIP trunk connectivity are welcome to suggest corrections or additions, which are checked against primary sources before any update.","introHtml":"The bandwidth figures here come from a mix of the standards that define the codecs and independent industry guidance on how those codecs behave on a real connection.[2][3][4] The per-call numbers, the calculation method, and the headroom advice are attributed to independent telecom sources rather than presented as fixed law, since exact figures shift with codec settings, packetization, and how much lower-layer overhead a given source counts.[1][6][7][8] The latency target comes from the ITU-T recommendation that sets it, and the jitter and packet-loss targets reflect the range independent network sources consistently cite.[5][9] Voice codecs, provider networks, and pricing change over time, so the specific numbers reflect what the cited sources reported on the verification date. Practitioners who design, provision, or troubleshoot SIP trunk connectivity 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":"b4438cc8-9415-45ac-bbcc-fca212ece374","sectionKey":"trade_offs","sectionType":"markdown_section","heading":"Trade-offs and limits worth knowing","introMarkdown":"Sizing a connection for SIP trunking involves a few tensions worth planning around rather than discovering after cutover.\n\n### Bigger is not the same as better\n\nBuying far more bandwidth than your call volume needs does not improve call quality once you have enough, because delay, jitter, and loss come from congestion, routing, and prioritization rather than from a shortage of raw megabits.[9] A modest link with quality of service and low latency carries voice better than a large link with neither.[9]\n\n### Compression saves bandwidth but costs fidelity\n\nChoosing G.729a over G.711 cuts per-call bandwidth by roughly two-thirds on the wire, which helps where the upstream link is tight, though it lowers the mean opinion score and can compound with any packet loss already on the path.[1][2] Where bandwidth is comfortable, G.711 keeps audio closest to landline quality.[6]\n\n### Voice competes with your other traffic\n\nOn a single shared connection, a cloud backup, a large file upload, or a video meeting can crowd out calls at the worst moment, so either prioritizing voice or separating it onto its own circuit is what keeps the busy hour clean.[9][11]\n\n### The public internet is outside your control\n\nQuality of service inside your building does not extend across the open internet to your provider, so latency and loss introduced along that path are only partly manageable, which is why many businesses choose a provider that also supplies the underlying connection.[9][11]","introHtml":"Sizing a connection for SIP trunking involves a few tensions worth planning around rather than discovering after cutover.
\nBuying far more bandwidth than your call volume needs does not improve call quality once you have enough, because delay, jitter, and loss come from congestion, routing, and prioritization rather than from a shortage of raw megabits.[9] A modest link with quality of service and low latency carries voice better than a large link with neither.[9]
\nChoosing G.729a over G.711 cuts per-call bandwidth by roughly two-thirds on the wire, which helps where the upstream link is tight, though it lowers the mean opinion score and can compound with any packet loss already on the path.[1][2] Where bandwidth is comfortable, G.711 keeps audio closest to landline quality.[6]
\nOn a single shared connection, a cloud backup, a large file upload, or a video meeting can crowd out calls at the worst moment, so either prioritizing voice or separating it onto its own circuit is what keeps the busy hour clean.[9][11]
\nQuality of service inside your building does not extend across the open internet to your provider, so latency and loss introduced along that path are only partly manageable, which is why many businesses choose a provider that also supplies the underlying connection.[9][11]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":10},{"id":"c556feac-ce79-4375-8f9d-5b502863ad5a","sectionKey":"what_it_is_not","sectionType":"markdown_section","heading":"What this is not","introMarkdown":"Sizing internet for SIP trunking gets confused with a few neighboring ideas, and separating them prevents over-buying or under-buying.\n\n### It is not just a download-speed question\n\nThe advertised download figure on an internet plan says little about SIP capacity, because a call needs equal room upstream and the upload number is usually the smaller of the two.[8][9] Two plans with the same download speed can hold very different numbers of calls depending on their upload.[9]\n\n### More bandwidth is not a fix for bad quality\n\nAdding megabits does not repair latency, jitter, or packet loss that come from congestion or a distant routing path, so a quality problem calls for prioritization, a better path, or a dedicated circuit rather than a larger pipe.[9]\n\n### Signaling bandwidth is not the media bandwidth\n\nThe SIP messages that set up and tear down a call use very little bandwidth, since almost all of a call's traffic is the RTP media stream carrying the audio, which is what the per-call figures measure.[4][10] Planning around the media rather than the signaling is what matters for sizing.[4]\n\n### It is not a one-time calculation\n\nCall volume grows, new sites come online, and codecs get changed, so the connection that fit last year may not fit today, which makes peak-hour capacity something to revisit rather than set once.[1]","introHtml":"Sizing internet for SIP trunking gets confused with a few neighboring ideas, and separating them prevents over-buying or under-buying.
\nThe advertised download figure on an internet plan says little about SIP capacity, because a call needs equal room upstream and the upload number is usually the smaller of the two.[8][9] Two plans with the same download speed can hold very different numbers of calls depending on their upload.[9]
\nAdding megabits does not repair latency, jitter, or packet loss that come from congestion or a distant routing path, so a quality problem calls for prioritization, a better path, or a dedicated circuit rather than a larger pipe.[9]
\nThe SIP messages that set up and tear down a call use very little bandwidth, since almost all of a call's traffic is the RTP media stream carrying the audio, which is what the per-call figures measure.[4][10] Planning around the media rather than the signaling is what matters for sizing.[4]
\nCall volume grows, new sites come online, and codecs get changed, so the connection that fit last year may not fit today, which makes peak-hour capacity something to revisit rather than set once.[1]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":11}],"citations":[{"title":"Tips for planning SIP trunk bandwidth","url":"https://www.business.att.com/learn/tech-advice/tips-for-planning-sip-trunk-bandwidth.html","excerpt":"SIP Trunk Peak Bandwidth = Peak CCP x 80Kb for G.711, or Peak CCP x 32Kb for the more efficient G.729a.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"Per-call bandwidth of about 80 kbps on G.711 and 32 kbps on G.729a; peak concurrent call paths as the sizing basis; multiply peak concurrent calls by per-call bandwidth; a SIP trunk replaces multiple PRIs over one IP connection","domain":"business.att.com","publisherName":"AT&T Business"},{"title":"G.711 vs G.729: VoIP Codec Comparison","url":"https://www.telcobridges.com/learning/sip-trunking/g711-vs-g729/","excerpt":"G.711 comes to around 85 to 87 kbps per direction and scores roughly 4.2 on MOS; G.729 is roughly 31 kbps per direction at about 3.9. The compression ratio in the payload is 8 to 1, but on the wire it collapses to roughly 2.8 to 1.","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 from a 64 kbps payload, G.729 about 31 kbps per direction; 50 packets per second at 20 ms; MOS about 4.2 for G.711 and 3.9 for G.729; 8:1 payload compression collapses to about 2.8:1 on the wire","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 64 kbps reference narrowband codec used as the baseline payload for per-call bandwidth; status in force","domain":"itu.int","publisherName":"International Telecommunication Union (ITU-T)"},{"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. The first twelve octets are present in every RTP packet.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-18T00:00:00","supportsText":"RTP carries the real-time audio media of a call, separate from signaling, with a 12-octet fixed header on every packet that contributes to per-call overhead","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"ITU-T G.114: One-way transmission time","url":"https://www.itu.int/rec/T-REC-G.114/","excerpt":"One-way transmission time. Status: In force. Includes guidance on one-way delay for Voice over IP.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-18T00:00:00","supportsText":"ITU-T G.114 sets one-way transmission time limits for voice; the preferred range keeps one-way delay under about 150 ms for natural interactivity, with 150 to 400 ms acceptable but degraded; status in force","domain":"itu.int","publisherName":"International Telecommunication Union (ITU-T)"},{"title":"Calculating VoIP bandwidth needs for SIP trunk services","url":"https://www.techtarget.com/searchunifiedcommunications/tip/Calculating-VoIP-bandwidth-needs-for-SIP-trunk-services","excerpt":"The bandwidth requirements above do not include IP overhead, which is an additional 23 Kbps. Necessary bandwidth is calculated according to the number of simultaneous streams (calls) and the codec.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"G.711 uses 64 kbps plus about 23 kbps of IP overhead for roughly 87 kbps per call; G.722 uses 48 to 64 kbps; G.729 uses 8 kbps; total bandwidth is per-call bandwidth times the number of simultaneous streams","domain":"techtarget.com","publisherName":"TechTarget"},{"title":"Calculating Bandwidth for SIP Trunks Made Easy","url":"https://www.nojitter.com/telecommunication-technology/calculating-bandwidth-for-sip-trunks-made-easy","excerpt":"For nearly every situation, it's safe to use 90 Kbps for G.711 and 32 Kbps for G.729a. Add an additional 20% of overhead for network inefficiencies.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"Practical per-call planning figures of about 90 kbps for G.711 and 32 kbps for G.729a; effect of sample size on data rate; multiply concurrent calls by per-call bandwidth and add roughly 20 percent overhead","domain":"nojitter.com","publisherName":"No Jitter"},{"title":"How Much Bandwidth Do I Need for SIP Trunking?","url":"https://www.sip.us/blog/latest-news/how-much-bandwidth-do-i-need-for-sip-trunking/","excerpt":"Multiply the number of phone calls you expect by the necessary bandwidth amount, then add a minimum of 10% additional capacity. Use the figure that is lower, typically the upload number.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"G.711 needs about 87 kbps per call including overhead; multiply expected concurrent calls by the per-call amount; add at least 10 percent additional capacity; use the lower speed-test figure, usually upload, and convert Mbps to kbps","domain":"sip.us","publisherName":"SIP.US"},{"title":"VoIP Network Requirements: Bandwidth, QoS, and Infrastructure","url":"https://www.ipcomms.net/blog/voip-network-requirements/","excerpt":"Target under 150 ms one-way latency, jitter below 30 ms, and packet loss under about 1 percent. Upload speed is typically the bottleneck; voice RTP should use DSCP EF (46). A 50/10 Mbps connection has enough upload for about 100 G.711 calls.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-18T00:00:00","supportsText":"Quality targets of under 150 ms one-way latency, jitter below 30 ms, and packet loss under about 1 percent; upload is typically the bottleneck; a 50/10 Mbps line holds about 100 G.711 calls; voice marked DSCP EF 46; a dedicated circuit advised for high volumes; about 100 kbps per call rule of thumb","domain":"ipcomms.net","publisherName":"IPComms"},{"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.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-18T00:00:00","supportsText":"SIP is the application-layer signaling protocol that sets up, modifies, and ends calls; the signaling is separate from the RTP media that dominates per-call bandwidth","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"Need a SIP Trunk? Here are 5 Things to Consider First","url":"https://www.atlantech.net/blog/need-a-sip-trunk-here-are-5-things-to-consider-first","excerpt":"With IP-based phone service, securing sufficient bandwidth is critical to quality of service (QoS). A provider that offers fiber Internet service provides better network transparency and voice data prioritization.","quoteText":null,"sourceRole":"SUPPORTING","verifiedAt":"2026-07-18T00:00:00","supportsText":"Corroborates that SIP trunking depends on sufficient internet bandwidth for quality of service, that bandwidth scales with users times usage, and that business fiber with voice prioritization suits SIP; corroborated by [1], [9], and [10]","domain":"atlantech.net","publisherName":"Atlantech Online"}],"revisions":[],"relatedAnswers":[{"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"},{"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 kind of internet connection and bandwidth do I need for SIP trunking?","text":"What kind of internet connection and bandwidth do I need for SIP trunking?","url":"https://www.answerstack.io/q/what-kind-of-internet-connection-and-bandwidth-do-i-need-for-sip-trunking","answerCount":1,"datePublished":"2026-07-18T15:16:59.312","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":"For SIP trunking you need a broadband internet connection sized to your busy-hour concurrent calls rather than to your total number of phones, with roughly 85 to 90 kilobits per second in each direction for every simultaneous call using the common G.711 codec, or about 31 kilobits per second per call with the more compressed G.729a.[1][2][6] Multiply your peak concurrent calls by the per-call figure and add 10 to 20 percent of headroom for signaling and other traffic to get the floor for the connection.[7][8] Raw speed is only half of it, because the connection also has to hold one-way latency under about 150 milliseconds, jitter under about 30 milliseconds, and packet loss under roughly 1 percent, and voice degrades on delay and loss well before it runs out of raw bandwidth.[5][9] Upload capacity is usually the limiting factor on asymmetric consumer lines, so a symmetric business connection with quality-of-service prioritization, often fiber, is the typical recommendation.[9][11]","url":"https://www.answerstack.io/q/what-kind-of-internet-connection-and-bandwidth-do-i-need-for-sip-trunking","upvoteCount":0,"datePublished":"2026-07-18T15:16:59.312","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":"Tips for planning SIP trunk bandwidth","url":"https://www.business.att.com/learn/tech-advice/tips-for-planning-sip-trunk-bandwidth.html"},{"@type":"CreativeWork","name":"G.711 vs G.729: VoIP Codec Comparison","url":"https://www.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":"RFC 3550: RTP: A Transport Protocol for Real-Time Applications","url":"https://datatracker.ietf.org/doc/html/rfc3550"},{"@type":"CreativeWork","name":"ITU-T G.114: One-way transmission time","url":"https://www.itu.int/rec/T-REC-G.114/"},{"@type":"CreativeWork","name":"Calculating VoIP bandwidth needs for SIP trunk services","url":"https://www.techtarget.com/searchunifiedcommunications/tip/Calculating-VoIP-bandwidth-needs-for-SIP-trunk-services"},{"@type":"CreativeWork","name":"Calculating Bandwidth for SIP Trunks Made Easy","url":"https://www.nojitter.com/telecommunication-technology/calculating-bandwidth-for-sip-trunks-made-easy"},{"@type":"CreativeWork","name":"How Much Bandwidth Do I Need for SIP Trunking?","url":"https://www.sip.us/blog/latest-news/how-much-bandwidth-do-i-need-for-sip-trunking/"},{"@type":"CreativeWork","name":"VoIP Network Requirements: Bandwidth, QoS, and Infrastructure","url":"https://www.ipcomms.net/blog/voip-network-requirements/"},{"@type":"CreativeWork","name":"RFC 3261: SIP: Session Initiation Protocol","url":"https://datatracker.ietf.org/doc/html/rfc3261"},{"@type":"CreativeWork","name":"Need a SIP Trunk? 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