A single VoIP call typically uses about 85 to 100 kbps of bandwidth in each direction when it runs the common G.711 codec, and closer to 31 kbps when it uses a compressed codec such as G.729.[1][2] The audio itself is smaller than that, 64 kbps for G.711 and 8 kbps for G.729, but every packet also carries 40 bytes of IP, UDP, and RTP headers, and at the usual rate of 50 packets per second those headers add roughly 16 kbps of overhead on top of the audio before any link-layer framing is counted.[1][7] A conversation runs in both directions at once, so you double the one-way figure to size a full call, and because voice needs steady delivery, most providers plan around 100 kbps per concurrent call with headroom to spare.[9][11] Cloud services vary the rate to match the network: Microsoft Teams lists 58 kbps as its recommended audio bitrate, while the Opus codec many apps use can range from 6 to 510 kbps depending on the quality setting.[10][6]
\n","summary":"Per-call VoIP bandwidth depends mostly on the codec. G.711 runs about 87 kbps in each direction once IP, UDP, and RTP overhead is added to its 64 kbps of audio, while G.729 compresses the same call to roughly 31 kbps. Adaptive codecs like Opus range from 6 to 510 kbps, and Microsoft Teams recommends 58 kbps for audio. Since calls are two-way, you double the one-direction figure, and most connections are sized at about 100 kbps per concurrent call plus headroom.","publishedAt":"2026-07-18T03:24:27.71","verifiedAt":"2026-07-17T00:00:00","editorialStatus":"APPROVED","lastReviewedAt":"2026-07-17T00:00:00","nextReviewDueAt":"2026-10-17T00:00:00","templateVersion":"v2","aliases":["How much bandwidth does VoIP use per call?","How much internet speed do I need for VoIP?","VoIP bandwidth per call in kbps","How much data does a VoIP call use?","What is the bandwidth of a G.711 call?","How many kbps does an internet phone call use?","VoIP bandwidth requirements per call","How much bandwidth for a VoIP phone call?","Bandwidth per concurrent VoIP call","How much upload speed does VoIP need?","G.729 vs G.711 bandwidth","How to calculate VoIP bandwidth"],"confidenceScore":92,"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":"49f73117-6f38-4aa0-9baa-8ae6278c08e4","sectionKey":"how_bandwidth_adds_up","sectionType":"markdown_section","heading":"How does bandwidth add up on a VoIP call?","introMarkdown":"A VoIP call uses bandwidth in two layers: the digitized audio itself, and the network packaging that wraps around it. A codec turns your voice into a digital stream at a set bitrate, 64 kbps for the standard G.711 codec or 8 kbps for the compressed G.729.[3][5] That stream is sliced into small packets, usually holding 20 milliseconds of audio each, and every packet gets a 40-byte header so the network knows where to send it.[1][7] Because a device sends 50 of those packets every second at the 20 millisecond interval, the headers alone add about 16 kbps in each direction, which is why the number you actually provision is always higher than the codec's raw bitrate.[1][7]\n\n### The payload is only part of the number\n\nThe audio payload sets the floor, not the total. A G.711 call carries 64 kbps of voice, but once you add the 40 bytes of IP, UDP, and RTP headers on every packet plus the framing the physical link adds, the call consumes roughly 87 kbps in each direction over Ethernet.[1][11] The same math hits a compressed codec harder in relative terms, because G.729 sends only 8 kbps of audio yet the fixed header cost pushes each call to about 31 kbps, so the eight-to-one compression ratio shrinks to closer to three-to-one on the wire.[1][2]\n\n### A call runs in both directions\n\nEvery figure so far describes one direction of audio. A phone call is full duplex, meaning both people can talk at once, so a live conversation carries a stream each way and you double the one-direction number to size the whole call.[9] A G.711 call that reads as 87 kbps per direction therefore occupies close to 174 kbps of total capacity while it is connected, split evenly between the upload and download sides.[1][9]","introHtml":"A VoIP call uses bandwidth in two layers: the digitized audio itself, and the network packaging that wraps around it. A codec turns your voice into a digital stream at a set bitrate, 64 kbps for the standard G.711 codec or 8 kbps for the compressed G.729.[3][5] That stream is sliced into small packets, usually holding 20 milliseconds of audio each, and every packet gets a 40-byte header so the network knows where to send it.[1][7] Because a device sends 50 of those packets every second at the 20 millisecond interval, the headers alone add about 16 kbps in each direction, which is why the number you actually provision is always higher than the codec's raw bitrate.[1][7]
\nThe audio payload sets the floor, not the total. A G.711 call carries 64 kbps of voice, but once you add the 40 bytes of IP, UDP, and RTP headers on every packet plus the framing the physical link adds, the call consumes roughly 87 kbps in each direction over Ethernet.[1][11] The same math hits a compressed codec harder in relative terms, because G.729 sends only 8 kbps of audio yet the fixed header cost pushes each call to about 31 kbps, so the eight-to-one compression ratio shrinks to closer to three-to-one on the wire.[1][2]
\nEvery figure so far describes one direction of audio. A phone call is full duplex, meaning both people can talk at once, so a live conversation carries a stream each way and you double the one-direction number to size the whole call.[9] A G.711 call that reads as 87 kbps per direction therefore occupies close to 174 kbps of total capacity while it is connected, split evenly between the upload and download sides.[1][9]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":0},{"id":"dfe4f29a-ce08-4d23-b00e-df8285aa7dd3","sectionKey":"codec_bandwidth_table","sectionType":"table_section","heading":"How much bandwidth do common codecs use?","introMarkdown":"The codec is the biggest single factor in per-call bandwidth, because it sets the audio bitrate that everything else builds on.[1] The figures below show each codec's audio payload and the bandwidth one direction of a call consumes once IP, UDP, and RTP headers and typical Ethernet framing are included. The sections after the table explain when each codec tends to be used.","introHtml":"The codec is the biggest single factor in per-call bandwidth, because it sets the audio bitrate that everything else builds on.[1] The figures below show each codec's audio payload and the bandwidth one direction of a call consumes once IP, UDP, and RTP headers and typical Ethernet framing are included. The sections after the table explain when each codec tends to be used.
\n","outroMarkdown":"These are typical figures at a 20 millisecond packet interval over Ethernet. The exact number shifts with the packet size and the type of link, as the following sections describe.[1][12]","outroHtml":"These are typical figures at a 20 millisecond packet interval over Ethernet. The exact number shifts with the packet size and the type of link, as the following sections describe.[1][12]
\n","contentJson":{"rows":[{"cells":["G.711","64 kbps","About 87 kbps","Narrowband, toll quality; the default on most business systems [1][3]"]},{"cells":["G.722","64 kbps","About 87 kbps","Wideband HD voice; clearer audio at about the same bandwidth [4][8]"]},{"cells":["G.729","8 kbps","About 31 kbps","Narrowband; used where bandwidth is limited [2][5]"]},{"cells":["Opus","6 to 510 kbps","Variable, often 24 to 40 kbps for speech","Adaptive wideband; common in apps and browsers [6][8]"]}],"columns":["Codec","Audio payload","Per-call bandwidth, one direction","Audio quality and typical use"]},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":1},{"id":"5b3478f4-31b4-4558-a942-d585af9695ef","sectionKey":"codec_g711","sectionType":"markdown_section","heading":"G.711: the toll-quality baseline","introMarkdown":"G.711 is the codec most business phone systems default to, and it consumes about 87 kbps in each direction per call.[1][11] It digitizes voice with pulse code modulation at 64 kbps and applies almost no compression, so the audio matches the quality of a traditional landline, which keeps it the common choice on networks that have bandwidth to spare.[3] The overhead is the reason the call needs 87 kbps rather than 64: 50 packets per second, each carrying 160 bytes of audio and 40 bytes of headers plus Ethernet framing, work out to roughly 87 kbps once multiplied through.[1] For most office connections that cost is easy to absorb, and the payoff is audio that sounds clean without the artifacts heavier compression can introduce. G.711 also handles fax tones and analog signaling more reliably than compressed codecs, so it is often kept in place for fax lines and alarm circuits even where regular voice moves to something lighter.[2]","introHtml":"G.711 is the codec most business phone systems default to, and it consumes about 87 kbps in each direction per call.[1][11] It digitizes voice with pulse code modulation at 64 kbps and applies almost no compression, so the audio matches the quality of a traditional landline, which keeps it the common choice on networks that have bandwidth to spare.[3] The overhead is the reason the call needs 87 kbps rather than 64: 50 packets per second, each carrying 160 bytes of audio and 40 bytes of headers plus Ethernet framing, work out to roughly 87 kbps once multiplied through.[1] For most office connections that cost is easy to absorb, and the payoff is audio that sounds clean without the artifacts heavier compression can introduce. G.711 also handles fax tones and analog signaling more reliably than compressed codecs, so it is often kept in place for fax lines and alarm circuits even where regular voice moves to something lighter.[2]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":2},{"id":"e938e95a-ae16-4e15-a4f4-ed9445c76cab","sectionKey":"codec_g722","sectionType":"markdown_section","heading":"G.722 and wideband HD voice","introMarkdown":"G.722 uses about the same bandwidth as G.711, near 87 kbps per direction, but spends it on a wider slice of the audio spectrum to produce high-definition voice.[4][11] Where narrowband codecs carry frequencies up to roughly 3.4 kHz, G.722 encodes up to about 7 kHz within the same 64 kbps envelope, so speech sounds fuller and consonants are easier to make out, which matters most on conference calls with several people talking.[4][8] The bandwidth cost is effectively a wash against G.711, so the choice between them is about audio quality and endpoint support rather than saving capacity. Many IP desk phones and softphones support G.722, and a call only uses it when both ends agree to it during setup, so a wideband call between two compatible devices falls back to narrowband the moment it reaches a device or carrier that does not offer it.[8]","introHtml":"G.722 uses about the same bandwidth as G.711, near 87 kbps per direction, but spends it on a wider slice of the audio spectrum to produce high-definition voice.[4][11] Where narrowband codecs carry frequencies up to roughly 3.4 kHz, G.722 encodes up to about 7 kHz within the same 64 kbps envelope, so speech sounds fuller and consonants are easier to make out, which matters most on conference calls with several people talking.[4][8] The bandwidth cost is effectively a wash against G.711, so the choice between them is about audio quality and endpoint support rather than saving capacity. Many IP desk phones and softphones support G.722, and a call only uses it when both ends agree to it during setup, so a wideband call between two compatible devices falls back to narrowband the moment it reaches a device or carrier that does not offer it.[8]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":3},{"id":"169f1c80-8aaf-41d7-b504-6696a444c80a","sectionKey":"codec_g729","sectionType":"markdown_section","heading":"G.729: compressed for constrained links","introMarkdown":"G.729 shrinks a call to about 31 kbps in each direction, roughly a third of what G.711 needs, by compressing the audio down to an 8 kbps payload.[2][5] It models speech with a technique called conjugate-structure algebraic-code-excited linear prediction rather than sampling it directly, which trades a small amount of fidelity for a large saving on the audio portion of the call.[5] The saving is real but smaller than the raw codec numbers suggest, because the 40-byte header on every packet does not compress along with the audio: an eight-to-one payload ratio becomes closer to three-to-one once headers are counted, so G.729 delivers meaningful relief only on links where bandwidth is genuinely tight.[1][2] It tends to fit branch offices on slower connections, high-density call centers packing many simultaneous calls onto one circuit, and older WAN links, and it is worth confirming any licensing terms your platform attaches to it before you standardize on it.","introHtml":"G.729 shrinks a call to about 31 kbps in each direction, roughly a third of what G.711 needs, by compressing the audio down to an 8 kbps payload.[2][5] It models speech with a technique called conjugate-structure algebraic-code-excited linear prediction rather than sampling it directly, which trades a small amount of fidelity for a large saving on the audio portion of the call.[5] The saving is real but smaller than the raw codec numbers suggest, because the 40-byte header on every packet does not compress along with the audio: an eight-to-one payload ratio becomes closer to three-to-one once headers are counted, so G.729 delivers meaningful relief only on links where bandwidth is genuinely tight.[1][2] It tends to fit branch offices on slower connections, high-density call centers packing many simultaneous calls onto one circuit, and older WAN links, and it is worth confirming any licensing terms your platform attaches to it before you standardize on it.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":4},{"id":"df8a3039-8eb4-4822-aa72-144a1b0018b4","sectionKey":"codec_opus","sectionType":"markdown_section","heading":"Opus and adaptive codecs in cloud calling","introMarkdown":"Opus does not have one fixed bandwidth figure, because it adjusts its bitrate to the network in real time, ranging from 6 kbps for low-quality narrowband speech up to 510 kbps for high-fidelity stereo.[6] For a typical voice call it settles in the tens of kilobits per second, often around 24 to 40 kbps for wideband speech, and it can drop lower when a connection degrades so the call stays up rather than failing.[6][8] That adaptability is why most modern softphones, browser-based calling tools, and cloud platforms use Opus or a similar variable codec instead of a fixed one. Microsoft Teams is a useful reference point: for one-to-one audio it lists 10 kbps as the minimum, 58 kbps as the recommended bitrate, and 76 kbps for best performance, each measured per endpoint in each direction.[10] The practical effect is that a cloud call often uses less bandwidth than a G.711 call while sounding better, since the codec spends its bits on audio quality rather than on fixed overhead.","introHtml":"Opus does not have one fixed bandwidth figure, because it adjusts its bitrate to the network in real time, ranging from 6 kbps for low-quality narrowband speech up to 510 kbps for high-fidelity stereo.[6] For a typical voice call it settles in the tens of kilobits per second, often around 24 to 40 kbps for wideband speech, and it can drop lower when a connection degrades so the call stays up rather than failing.[6][8] That adaptability is why most modern softphones, browser-based calling tools, and cloud platforms use Opus or a similar variable codec instead of a fixed one. Microsoft Teams is a useful reference point: for one-to-one audio it lists 10 kbps as the minimum, 58 kbps as the recommended bitrate, and 76 kbps for best performance, each measured per endpoint in each direction.[10] The practical effect is that a cloud call often uses less bandwidth than a G.711 call while sounding better, since the codec spends its bits on audio quality rather than on fixed overhead.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":5},{"id":"a51a6c4a-4ebc-4100-8a29-d1c56ab520d9","sectionKey":"beyond_codec","sectionType":"markdown_section","heading":"What else changes the number beyond the codec?","introMarkdown":"The codec sets the baseline, but a handful of other factors move the real per-call figure up or down, sometimes enough to matter when you are sizing a link.\n\n### Packet size and packet rate\n\nSmaller packets mean lower delay but more overhead, because each packet carries the same 40-byte header no matter how little audio is inside it.[7][12] The common 20 millisecond interval produces 50 packets per second; stretching it to 30 or 40 milliseconds sends fewer, larger packets and trims the header overhead, at the cost of a little added latency and the fact that any lost packet then removes more speech.[12] Most systems stay near 20 milliseconds because it balances the two reasonably well.\n\n### The fixed header tax\n\nEvery packet carries 40 bytes of IP, UDP, and RTP headers, and the physical link adds its own framing on top, around 18 bytes for Ethernet.[1][7] That overhead is constant regardless of codec, so it weighs far more heavily on a compressed call than a raw one: the same 16 kbps of header cost is trivial next to G.711's 64 kbps of audio but larger than G.729's entire 8 kbps payload.[1] It is the single reason no codec ever uses only its advertised bitrate on a real network.\n\n### Two-way audio\n\nA conversation is full duplex, so the capacity a call occupies is double the one-direction figure, split between upload and download.[9] Sizing only for the download side understates the need, and on connections where upload is slower than download, the upload side is usually what limits how many calls fit.[13]\n\n### Encryption\n\nEncrypting the media and signaling, now standard on business systems, adds a small amount of per-packet overhead on top of the codec and headers.[12] The increase is modest for a single call, but it is worth including in the margin when you plan capacity for many calls at once.","introHtml":"The codec sets the baseline, but a handful of other factors move the real per-call figure up or down, sometimes enough to matter when you are sizing a link.
\nSmaller packets mean lower delay but more overhead, because each packet carries the same 40-byte header no matter how little audio is inside it.[7][12] The common 20 millisecond interval produces 50 packets per second; stretching it to 30 or 40 milliseconds sends fewer, larger packets and trims the header overhead, at the cost of a little added latency and the fact that any lost packet then removes more speech.[12] Most systems stay near 20 milliseconds because it balances the two reasonably well.
\nEvery packet carries 40 bytes of IP, UDP, and RTP headers, and the physical link adds its own framing on top, around 18 bytes for Ethernet.[1][7] That overhead is constant regardless of codec, so it weighs far more heavily on a compressed call than a raw one: the same 16 kbps of header cost is trivial next to G.711's 64 kbps of audio but larger than G.729's entire 8 kbps payload.[1] It is the single reason no codec ever uses only its advertised bitrate on a real network.
\nA conversation is full duplex, so the capacity a call occupies is double the one-direction figure, split between upload and download.[9] Sizing only for the download side understates the need, and on connections where upload is slower than download, the upload side is usually what limits how many calls fit.[13]
\nEncrypting the media and signaling, now standard on business systems, adds a small amount of per-packet overhead on top of the codec and headers.[12] The increase is modest for a single call, but it is worth including in the margin when you plan capacity for many calls at once.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":6},{"id":"10f7573c-b02e-4e0d-b154-b1923639e771","sectionKey":"sizing_and_tradeoffs","sectionType":"markdown_section","heading":"How do you size a connection, and what should you watch?","introMarkdown":"A practical way to size a connection is to multiply your peak number of simultaneous calls by about 100 kbps in each direction, then add headroom.[11][13] The 100 kbps figure is deliberately higher than any single codec needs, because it leaves room for headers, the occasional retransmission, and normal fluctuation, so you are not sizing to the theoretical floor.[9]\n\n### Working the numbers\n\nFor a team that might have 20 calls active at once, that formula points to roughly 2 Mbps of dedicated capacity in each direction for voice, before any other traffic.[13] A useful sanity check on the load is data volume rather than rate: an hour on a G.711 call moves on the order of 600 MB, while the same hour on G.729 moves closer to 220 MB, which is why compressed codecs still matter on metered or capped links.[9] Because voice is symmetric, plan the upload side with the same care as the download, since residential and many business broadband plans give far less upload, and that is where call quality tends to break first under load.[11][13]\n\n### Bandwidth alone does not guarantee a clean call\n\nHaving enough capacity is necessary but not sufficient, because voice is sensitive to how packets arrive, not just how many fit.[13] Even with plenty of raw speed, a call degrades when packets arrive at uneven intervals, a problem called jitter, or when some packets are lost outright, and high latency adds delay of its own.[13] Prioritizing voice with quality-of-service settings and reserving capacity for it usually does more for call quality than simply buying a bigger connection.[11] The compression trade-off runs the same way, because a codec like G.729 saves bandwidth but gives up some audio fidelity, so it earns its place on constrained links and adds little on a connection that can comfortably carry G.711 or a wideband codec.[2]","introHtml":"A practical way to size a connection is to multiply your peak number of simultaneous calls by about 100 kbps in each direction, then add headroom.[11][13] The 100 kbps figure is deliberately higher than any single codec needs, because it leaves room for headers, the occasional retransmission, and normal fluctuation, so you are not sizing to the theoretical floor.[9]
\nFor a team that might have 20 calls active at once, that formula points to roughly 2 Mbps of dedicated capacity in each direction for voice, before any other traffic.[13] A useful sanity check on the load is data volume rather than rate: an hour on a G.711 call moves on the order of 600 MB, while the same hour on G.729 moves closer to 220 MB, which is why compressed codecs still matter on metered or capped links.[9] Because voice is symmetric, plan the upload side with the same care as the download, since residential and many business broadband plans give far less upload, and that is where call quality tends to break first under load.[11][13]
\nHaving enough capacity is necessary but not sufficient, because voice is sensitive to how packets arrive, not just how many fit.[13] Even with plenty of raw speed, a call degrades when packets arrive at uneven intervals, a problem called jitter, or when some packets are lost outright, and high latency adds delay of its own.[13] Prioritizing voice with quality-of-service settings and reserving capacity for it usually does more for call quality than simply buying a bigger connection.[11] The compression trade-off runs the same way, because a codec like G.729 saves bandwidth but gives up some audio fidelity, so it earns its place on constrained links and adds little on a connection that can comfortably carry G.711 or a wideband codec.[2]
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":7},{"id":"4fa607c7-c779-4a50-9b0b-3aac5fc3c4d3","sectionKey":"contributor_perspective","sectionType":"markdown_section","heading":"How this answer was researched","introMarkdown":"The bandwidth figures here come from the codec standards themselves and from independent calculations that add real-world network overhead. The audio bitrates are drawn from the ITU-T recommendations that define G.711, G.722, and G.729, and the Opus range from the IETF document that specifies it.[3][4][5][6] The per-call figures that include IP, UDP, and RTP headers rely on the packet-level method Cisco published and on independent codec comparisons, cross-checked against a cloud provider's own published numbers.[1][2][10] The sizing guidance and data-per-call volumes come from several independent VoIP providers rather than any single vendor.[9][11][13] Exact per-call bandwidth depends on the packet interval and the type of link, so the numbers here are typical values rather than fixed constants, and they were verified on the date shown. Practitioners who design or operate VoIP networks are welcome to suggest corrections, which are checked against primary sources before any update.","introHtml":"The bandwidth figures here come from the codec standards themselves and from independent calculations that add real-world network overhead. The audio bitrates are drawn from the ITU-T recommendations that define G.711, G.722, and G.729, and the Opus range from the IETF document that specifies it.[3][4][5][6] The per-call figures that include IP, UDP, and RTP headers rely on the packet-level method Cisco published and on independent codec comparisons, cross-checked against a cloud provider's own published numbers.[1][2][10] The sizing guidance and data-per-call volumes come from several independent VoIP providers rather than any single vendor.[9][11][13] Exact per-call bandwidth depends on the packet interval and the type of link, so the numbers here are typical values rather than fixed constants, and they were verified on the date shown. Practitioners who design or operate VoIP networks are welcome to suggest corrections, which are checked against primary sources before any update.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":"This answer was written and reviewed by the AnswerStack Editorial Team, which has no commercial stake in the products, companies, or methods discussed. Every claim is cited inline and verified on the dates shown.","noteHtml":"This answer was written and reviewed by the AnswerStack Editorial Team, which has no commercial stake in the products, companies, or methods discussed. Every claim is cited inline and verified on the dates shown.
\n","sortOrder":8}],"citations":[{"title":"How to Calculate Bandwidth for Cisco IP Calls","url":"https://www.cbtnuggets.com/blog/technology/networking/how-to-calculate-bandwidth-for-cisco-ip-calls","excerpt":"1000 milliseconds / 20 milliseconds = 50 packets per second ... 160 bytes voice payload plus 40 bytes headers plus 18 bytes Ethernet ... 87.2 Kbps.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-17T00:00:00","supportsText":"Packet-level per-call bandwidth method: 40-byte IP/UDP/RTP header, 50 packets per second at 20 ms, G.711 160-byte payload plus 18-byte Ethernet framing equals 87.2 kbps, G.729 equals 31.2 kbps","domain":"cbtnuggets.com","publisherName":"CBT Nuggets"},{"title":"G.711 vs G.729: VoIP Codec Comparison","url":"https://telcobridges.com/learning/sip-trunking/g711-vs-g729/","excerpt":"G.711 ... each carrying 160 bytes of audio plus about 54 bytes of RTP/UDP/IP/Ethernet headers, for a total of around 85 to 87 kbps per direction. G.729 ... around 31 kbps per direction.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-17T00:00:00","supportsText":"G.711 64 kbps payload reaching about 85 to 87 kbps per direction with headers; G.729 8 kbps payload reaching about 31 kbps; the eight-to-one payload ratio narrows to about three-to-one once overhead is included","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":"2026-07-17T00:00:00","supportsText":"G.711 defines PCM voice coding at 64 kbit/s, the toll-quality narrowband reference","domain":"itu.int","publisherName":"International Telecommunication Union (ITU-T)"},{"title":"ITU-T G.722: 7 kHz audio-coding within 64 kbit/s","url":"https://www.itu.int/rec/T-REC-G.722/","excerpt":"7 kHz audio-coding within 64 kbit/s. Status: In force.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-17T00:00:00","supportsText":"G.722 provides wideband (7 kHz) audio coding within a 64 kbit/s envelope, the basis for HD voice","domain":"itu.int","publisherName":"International Telecommunication Union (ITU-T)"},{"title":"ITU-T G.729: Coding of speech at 8 kbit/s using CS-ACELP","url":"https://www.itu.int/rec/T-REC-G.729/","excerpt":"Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear prediction (CS-ACELP). Status: In force.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-17T00:00:00","supportsText":"G.729 codes speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear prediction","domain":"itu.int","publisherName":"International Telecommunication Union (ITU-T)"},{"title":"RFC 6716: Definition of the Opus Audio Codec","url":"https://datatracker.ietf.org/doc/html/rfc6716","excerpt":"It scales from low bitrate narrowband speech at 6 kbit/s to very high quality stereo music at 510 kbit/s ... designed to handle a wide range of interactive audio applications, including Voice over IP.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-17T00:00:00","supportsText":"Opus scales from 6 kbit/s narrowband speech to 510 kbit/s stereo and is designed for Voice over IP","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":"The first twelve octets are present in every RTP packet ... audio data in small chunks of, say, 20 ms duration.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-17T00:00:00","supportsText":"RTP carries real-time audio, sends it in roughly 20 ms chunks, and adds a fixed 12-byte header on every packet","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"What Are VoIP Codecs & How Do They Affect Call Sound Quality?","url":"https://www.nextiva.com/blog/voip-codecs.html","excerpt":"G.711: 64 kbps ... G.722: 48/56/64 kbps (wideband/HD) ... G.729: 8 kbps ... Opus: 8-512 kbps (wideband).","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-17T00:00:00","supportsText":"Codec bitrates and the narrowband versus wideband distinction: G.711 64 kbps, G.722 48 to 64 kbps wideband, G.729 8 kbps, Opus 8 to 512 kbps","domain":"nextiva.com","publisherName":"Nextiva"},{"title":"How Much Data Does VoIP Use? Tips To Save Bandwidth","url":"https://www.nextiva.com/blog/voip-data-usage.html","excerpt":"G.711: 87.2 Kbps ... G.729: 31.2 Kbps ... one-way; double them for two-way ... 613 MB for a 1-hour call ... 219 MB for a 1-hour call ... a minimum speed of 100 kbps per phone line.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-17T00:00:00","supportsText":"Per-call bandwidth of 87.2 kbps for G.711 and 31.2 kbps for G.729 one-way, doubled for two-way; data volume of about 613 MB per hour for G.711 and 219 MB for G.729; 80 to 100 kbps per call and 100 kbps per line","domain":"nextiva.com","publisherName":"Nextiva"},{"title":"Prepare your organization's network for Teams","url":"https://learn.microsoft.com/en-us/microsoftteams/prepare-network","excerpt":"Audio One-to-one 10/10 58/58 76/76 ... Minimum, Recommended, and Best performance bandwidth requirements are based on per-endpoint usage.","quoteText":null,"sourceRole":"PRIMARY","verifiedAt":"2026-07-17T00:00:00","supportsText":"Microsoft Teams one-to-one audio bandwidth per endpoint in each direction: 10 kbps minimum, 58 kbps recommended, 76 kbps best performance","domain":"learn.microsoft.com","publisherName":"Microsoft Learn"},{"title":"VoIP Bandwidth Requirements","url":"https://support.intermedia.com/app/articles/detail/a_id/11220/~/voip-bandwidth-requirements","excerpt":"The minimum internet connection speed required for VoIP is between 90 kbps to 156 kbps ... Your phones need a consistent, reserved amount of bandwidth.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-17T00:00:00","supportsText":"Recommended per-call bandwidth of roughly 90 to 156 kbps, sizing by multiplying phones by the per-call figure, and equal upload and download allocation with reserved bandwidth for voice","domain":"support.intermedia.com","publisherName":"Intermedia"},{"title":"VoIP Bandwidth Calculation","url":"https://voipstudio.com/blog/voip-bandwidth-calculation/","excerpt":"smaller packets will reduce time delays while larger packets are more efficient in terms of bandwidth ... encryption imposes additional bandwidth requirements on each VoIP call.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-17T00:00:00","supportsText":"Packet interval trade-off (smaller packets lower delay but higher overhead, most use 20 to 30 ms), the per-packet header tax, and that compression and encryption change per-call bandwidth","domain":"voipstudio.com","publisherName":"VoIPstudio"},{"title":"VoIP Network Requirements: Bandwidth, QoS, and Infrastructure","url":"https://www.ipcomms.net/blog/voip-network-requirements/","excerpt":"multiply your peak concurrent calls by 100 kbps ... 20 simultaneous calls need approximately 2 Mbps of dedicated bandwidth in each direction ... bandwidth must be available in both directions.","quoteText":null,"sourceRole":"INDEPENDENT","verifiedAt":"2026-07-17T00:00:00","supportsText":"Codec bandwidth including overhead (G.711 87 kbps, G.729 32 kbps, G.722 87 kbps), sizing by multiplying peak concurrent calls by 100 kbps (20 calls equals about 2 Mbps each direction), the need for capacity in both directions, and upload as the usual bottleneck","domain":"ipcomms.net","publisherName":"IPComms"}],"revisions":[],"relatedAnswers":[{"id":"2bc2d203-d1bf-4685-91a1-cd586dddf234","slug":"do-i-need-a-phone-line-for-voip","question":"Do I need a phone line for VoIP?","publishedAt":"2026-07-18T03:33:04.199","confidenceScore":92,"confidenceLabel":"High","industry":{"id":"ff619d7c-d7d7-485e-a05a-53fba07f33ed","slug":"telecommunications","label":"Telecommunications","description":"Business voice, fiber, UCaaS, and network services"},"topic":{"slug":"voip","label":"VoIP","description":"Voice over IP for business: how it works, what it requires, and how it compares to traditional phone service.","schemaKind":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"snippet":"VoIP replaces the phone line with a broadband internet connection, not dial-tone service. You need bandwidth of about 85 to 100 kbps per call and a device to talk on: an IP phone, a softphone app, or a regular phone on an adapter. You can keep your existing landline number by porting it. The real trade-offs are power and internet dependence, which battery backup helps cover, and 911 that routes to an address you register.","url":"/q/do-i-need-a-phone-line-for-voip"},{"id":"316da151-4ac7-4ec8-9c85-ce74f5593693","slug":"do-i-need-a-special-phone-for-voip","question":"Do I need a special phone for VoIP?","publishedAt":"2026-07-18T03:30:09.851","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":"voip","label":"VoIP","description":"Voice over IP for business: how it works, what it requires, and how it compares to traditional phone service.","schemaKind":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"snippet":"A dedicated VoIP handset is optional. The same call can run through a softphone app on a phone or laptop, a purpose-built IP desk phone, a web browser using WebRTC, or your current analog phone plugged into an adapter. Each path needs an internet connection and equipment that can speak VoIP. Certified or approved phones exist for full feature support, but any SIP-compliant device generally works, and a whole team can pilot VoIP on hardware people already own.","url":"/q/do-i-need-a-special-phone-for-voip"},{"id":"0ef1cfdf-0d57-4903-8eb6-b422a49ece05","slug":"what-type-of-internet-connection-do-i-need-for-voip","question":"What type of internet connection do I need for VoIP?","publishedAt":"2026-07-18T03:27:41.235","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":"voip","label":"VoIP","description":"Voice over IP for business: how it works, what it requires, and how it compares to traditional phone service.","schemaKind":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"snippet":"A VoIP call needs very little bandwidth, roughly 100 kbps in each direction, so the deciding factors are upload capacity and network quality rather than raw download speed. Aim for latency under 150 ms, jitter under 30 ms, and packet loss under 1 percent, and size the link to your peak number of simultaneous calls. Fiber, cable, DSL, fixed wireless, and low-orbit satellite can all carry voice; symmetrical fiber handles many concurrent lines best, while geostationary satellite adds too much delay.","url":"/q/what-type-of-internet-connection-do-i-need-for-voip"},{"id":"3e40a234-e3c8-43ca-885e-273a0204f4b3","slug":"how-is-voip-different-from-a-landline","question":"How is VoIP different from a landline?","publishedAt":"2026-07-18T03:27:24.417","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":"voip","label":"VoIP","description":"Voice over IP for business: how it works, what it requires, and how it compares to traditional phone service.","schemaKind":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"snippet":"A landline carries your voice as an electrical signal over a dedicated copper circuit on the traditional phone network, while VoIP sends it as digital packets over your broadband internet. That split drives the rest: a landline keeps working in a power outage and reports a fixed 911 address, while VoIP costs less, moves between devices, adds features like mobile apps and voicemail-to-email, and depends on your own power and internet. US rules class VoIP that reaches the phone network as 'interconnected VoIP,' with its own 911 duties.","url":"/q/how-is-voip-different-from-a-landline"}],"contributorStats":{"verifiedAnswers":224,"openDisputes":0},"schemaJson":{"@context":"https://schema.org","@type":"Question","name":"How much bandwidth does a VoIP call use?","text":"How much bandwidth does a VoIP call use?","url":"https://www.answerstack.io/q/how-much-bandwidth-does-a-voip-call-use","answerCount":1,"datePublished":"2026-07-18T03:24:27.71","author":{"@type":"Person","name":"AnswerStack Editorial Team","worksFor":{"@type":"Organization","name":"AnswerStack"},"url":"https://www.answerstack.io/contributors/answer-stack"},"about":[{"@type":"Thing","name":"VoIP"},{"@type":"Thing","name":"Telecommunications"}],"acceptedAnswer":{"@type":"Answer","text":"A single VoIP call typically uses about 85 to 100 kbps of bandwidth in each direction when it runs the common G.711 codec, and closer to 31 kbps when it uses a compressed codec such as G.729.[1][2] The audio itself is smaller than that, 64 kbps for G.711 and 8 kbps for G.729, but every packet also carries 40 bytes of IP, UDP, and RTP headers, and at the usual rate of 50 packets per second those headers add roughly 16 kbps of overhead on top of the audio before any link-layer framing is counted.[1][7] A conversation runs in both directions at once, so you double the one-way figure to size a full call, and because voice needs steady delivery, most providers plan around 100 kbps per concurrent call with headroom to spare.[9][11] Cloud services vary the rate to match the network: Microsoft Teams lists 58 kbps as its recommended audio bitrate, while the Opus codec many apps use can range from 6 to 510 kbps depending on the quality setting.[10][6]","url":"https://www.answerstack.io/q/how-much-bandwidth-does-a-voip-call-use","upvoteCount":0,"datePublished":"2026-07-18T03:24:27.71","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":"How to Calculate Bandwidth for Cisco IP Calls","url":"https://www.cbtnuggets.com/blog/technology/networking/how-to-calculate-bandwidth-for-cisco-ip-calls"},{"@type":"CreativeWork","name":"G.711 vs G.729: VoIP Codec Comparison","url":"https://telcobridges.com/learning/sip-trunking/g711-vs-g729/"},{"@type":"CreativeWork","name":"ITU-T G.711: Pulse code modulation (PCM) of voice frequencies","url":"https://www.itu.int/rec/T-REC-G.711/"},{"@type":"CreativeWork","name":"ITU-T G.722: 7 kHz audio-coding within 64 kbit/s","url":"https://www.itu.int/rec/T-REC-G.722/"},{"@type":"CreativeWork","name":"ITU-T G.729: Coding of speech at 8 kbit/s using CS-ACELP","url":"https://www.itu.int/rec/T-REC-G.729/"},{"@type":"CreativeWork","name":"RFC 6716: Definition of the Opus Audio Codec","url":"https://datatracker.ietf.org/doc/html/rfc6716"},{"@type":"CreativeWork","name":"RFC 3550: RTP: A Transport Protocol for Real-Time Applications","url":"https://datatracker.ietf.org/doc/html/rfc3550"},{"@type":"CreativeWork","name":"What Are VoIP Codecs & How Do They Affect Call Sound Quality?","url":"https://www.nextiva.com/blog/voip-codecs.html"},{"@type":"CreativeWork","name":"How Much Data Does VoIP Use? Tips To Save Bandwidth","url":"https://www.nextiva.com/blog/voip-data-usage.html"},{"@type":"CreativeWork","name":"Prepare your organization's network for Teams","url":"https://learn.microsoft.com/en-us/microsoftteams/prepare-network"},{"@type":"CreativeWork","name":"VoIP Bandwidth Requirements","url":"https://support.intermedia.com/app/articles/detail/a_id/11220/~/voip-bandwidth-requirements"},{"@type":"CreativeWork","name":"VoIP Bandwidth Calculation","url":"https://voipstudio.com/blog/voip-bandwidth-calculation/"},{"@type":"CreativeWork","name":"VoIP Network Requirements: Bandwidth, QoS, and Infrastructure","url":"https://www.ipcomms.net/blog/voip-network-requirements/"}]}}}