Cisco's published calculation puts a VoIP call using the G.711 codec at 87.2 kbps in each direction, and a call using the compressed G.729 codec at 31.2 kbps. Both figures count Ethernet framing and assume the audio is sent in packets that each hold 20 milliseconds of speech. A different codec, packet size or type of link gives a different number, and some links add overhead of their own. So the useful thing to know is how the figure is built.
\n","summary":"Cisco's calculation puts a G.711 VoIP call at 87.2 kbps in each direction and a G.729 call at 31.2 kbps, counting IP, UDP, RTP and Ethernet headers with 20 milliseconds of speech per packet. The figure changes with codec, packet size and link type. Multiply it by simultaneous calls and compare it with upload and download separately.","publishedAt":"2026-05-09T03:24:27.71","verifiedAt":"2026-09-14T00:00:00","editorialStatus":"APPROVED","lastReviewedAt":"2026-10-10T04:43:37.043","nextReviewDueAt":"2026-12-14T00:00:00","templateVersion":"v2","aliases":["How much data does a VoIP call use?","How much internet speed do I need for VoIP?","How many MB does an hour of VoIP use?","How much bandwidth does G.711 use per call?","VoIP bandwidth per call","How much bandwidth does a SIP call use?"],"confidenceScore":84,"confidenceLabel":"High","canonicalUrl":null},"contributor":{"id":"ec39deab-44fe-48d8-9029-fefe993ab85a","slug":"answer-stack","displayName":"AnswerStack","websiteUrl":null},"contributorOrganizationProfile":null,"contributorPerson":null,"sections":[{"id":"95acb68e-ca25-4ab4-9c72-15741ef024ae","sectionKey":"where_87_2_kbps_comes_from","sectionType":"markdown_section","heading":"Where 87.2 kbps comes from","introMarkdown":"G.711 produces 64 kbps of audio. At 20 milliseconds per packet, each packet holds 160 bytes of speech, and the phone sends 50 packets a second.\n\nEvery packet also carries addressing. Cisco's [voice bandwidth calculation](https://www.cisco.com/c/en/us/support/docs/voice/voice-quality/7934-bwidth-consume.html) counts 40 bytes of IP, UDP and RTP headers, plus 18 bytes of Ethernet framing. That makes each packet 218 bytes on the wire, or 1,744 bits. Fifty of those a second is 87,200 bits per second, which is 87.2 kbps.\n\nSo about a quarter of a G.711 call is packaging. The headers stay the same size when the audio gets smaller, and that limits what compression can save. G.729 squeezes the audio to 8 kbps, an eighth of G.711. Each packet then carries 20 bytes of speech and the same 58 bytes of headers and framing. The call uses 31.2 kbps, which is a little over a third of the G.711 figure.","introHtml":"G.711 produces 64 kbps of audio. At 20 milliseconds per packet, each packet holds 160 bytes of speech, and the phone sends 50 packets a second.
\nEvery packet also carries addressing. Cisco's voice bandwidth calculation counts 40 bytes of IP, UDP and RTP headers, plus 18 bytes of Ethernet framing. That makes each packet 218 bytes on the wire, or 1,744 bits. Fifty of those a second is 87,200 bits per second, which is 87.2 kbps.
\nSo about a quarter of a G.711 call is packaging. The headers stay the same size when the audio gets smaller, and that limits what compression can save. G.729 squeezes the audio to 8 kbps, an eighth of G.711. Each packet then carries 20 bytes of speech and the same 58 bytes of headers and framing. The call uses 31.2 kbps, which is a little over a third of the G.711 figure.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":10},{"id":"2003165e-c39a-41d5-ad1e-cae382c0f624","sectionKey":"per_call_figures_by_codec_and_packet_size","sectionType":"table_section","heading":"Per-call figures by codec and packet size","introMarkdown":null,"introHtml":null,"outroMarkdown":null,"outroHtml":null,"contentJson":{"rows":[{"cells":["G.711","64 kbps","20 ms","87.2 kbps"]},{"cells":["G.711","64 kbps","30 ms","79.5 kbps"]},{"cells":["G.729","8 kbps","20 ms","31.2 kbps"]},{"cells":["G.729","8 kbps","30 ms","23.5 kbps"]}],"columns":["Codec","Audio rate","Speech per packet","Each direction on Ethernet"]},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":20},{"id":"2e5d4ff8-667e-49f7-ac74-e3b150ad1b61","sectionKey":"each_direction_is_counted_separately","sectionType":"markdown_section","heading":"Each direction is counted separately","introMarkdown":"A call is two streams of audio, one from you and one to you. Cisco's [contact center design guide](https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cust_contact/contact_center/crs/express_12_5/design/guide/html/index/Bandwidth-Latency-and-QoS-Considerations-for-Unified-CCX-for-Release-12-5-1.html) adds the two together. It puts a G.711 call at 174.4 kbps with 20-millisecond packets and 159.0 kbps with 30-millisecond packets, and a G.729 call at 62.4 and 47.0 kbps. The 30-millisecond rows in the table are half of those two-way totals.\n\nThat total is easy to misapply. An internet connection has separate capacity for upload and download, and each stream uses only one of them. The guide makes the point with a 100 Mbps full-duplex link, which has 100 Mbps in each direction. On that link a G.711 call takes 87.2 kbps of the upload and 87.2 kbps of the download.\n\nSo compare the one-way figure with each direction of your connection. If your service has less upload than download, the upload figure sets the limit on how many calls fit.","introHtml":"A call is two streams of audio, one from you and one to you. Cisco's contact center design guide adds the two together. It puts a G.711 call at 174.4 kbps with 20-millisecond packets and 159.0 kbps with 30-millisecond packets, and a G.729 call at 62.4 and 47.0 kbps. The 30-millisecond rows in the table are half of those two-way totals.
\nThat total is easy to misapply. An internet connection has separate capacity for upload and download, and each stream uses only one of them. The guide makes the point with a 100 Mbps full-duplex link, which has 100 Mbps in each direction. On that link a G.711 call takes 87.2 kbps of the upload and 87.2 kbps of the download.
\nSo compare the one-way figure with each direction of your connection. If your service has less upload than download, the upload figure sets the limit on how many calls fit.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":30},{"id":"f72947f9-a717-4a08-8172-bb401dd00ee6","sectionKey":"what_changes_the_number","sectionType":"markdown_section","heading":"What changes the number","introMarkdown":"Packet size is the first variable. Putting more speech into each packet means fewer packets a second, and fewer packets means fewer headers. That's why the 30-millisecond rows in the table are lower. Cisco notes the cost: a larger payload reduces bandwidth and increases overall delay. [RFC 3551](https://www.rfc-editor.org/rfc/rfc3551.html), which sets 20 milliseconds as the default for audio over RTP, adds that longer packets make packet loss more noticeable, because each lost packet takes more speech with it.\n\nThe link is the second. The 18 bytes of framing belong to Ethernet. Cisco's formula uses the header of whichever link carries the call, so the same codec gives a different figure on a different kind of circuit. Cisco's own sample shows how far that can go. With G.729 on a link that has a 6-byte header and compresses the IP, UDP and RTP headers down to 2 bytes, the call comes to 11.2 kbps. Cisco also notes that this header compression isn't available over Ethernet.\n\nEncryption and VPN tunnels move the number the other way, because they wrap each packet in more bytes. The amount depends on the method, so add it from your own equipment's documentation.\n\nSilence is the last variable. Some systems stop sending packets when nobody is talking. Cisco says this can save up to 35 percent on average across more than 24 calls, and it also says the saving must not be counted when you size a network. A link has to carry the moments when everyone talks at once.","introHtml":"Packet size is the first variable. Putting more speech into each packet means fewer packets a second, and fewer packets means fewer headers. That's why the 30-millisecond rows in the table are lower. Cisco notes the cost: a larger payload reduces bandwidth and increases overall delay. RFC 3551, which sets 20 milliseconds as the default for audio over RTP, adds that longer packets make packet loss more noticeable, because each lost packet takes more speech with it.
\nThe link is the second. The 18 bytes of framing belong to Ethernet. Cisco's formula uses the header of whichever link carries the call, so the same codec gives a different figure on a different kind of circuit. Cisco's own sample shows how far that can go. With G.729 on a link that has a 6-byte header and compresses the IP, UDP and RTP headers down to 2 bytes, the call comes to 11.2 kbps. Cisco also notes that this header compression isn't available over Ethernet.
\nEncryption and VPN tunnels move the number the other way, because they wrap each packet in more bytes. The amount depends on the method, so add it from your own equipment's documentation.
\nSilence is the last variable. Some systems stop sending packets when nobody is talking. Cisco says this can save up to 35 percent on average across more than 24 calls, and it also says the saving must not be counted when you size a network. A link has to carry the moments when everyone talks at once.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":40},{"id":"e2e39705-11da-400a-9e44-3b5a7d4b4df1","sectionKey":"multiplying_by_simultaneous_calls","sectionType":"markdown_section","heading":"Multiplying by simultaneous calls","introMarkdown":"The per-call figure applies to each call in progress at the same moment, so multiply by calls and not by employees. Suppose ten people in your office are on G.711 calls at once. They need 872 kbps of upload and 872 kbps of download, on a connection that's also carrying everything else the office does. With G.729 the same ten calls need 312 kbps each way.\n\nWorking out how many calls happen at once is the harder estimate. We cover it in [how many simultaneous calls you need on a SIP trunk](https://www.answerstack.io/q/how-many-simultaneous-calls-do-i-need-or-get-on-a-sip-trunk). If the calls are part of a wider setup with video and screen sharing, see [what UCaaS bandwidth requirements to plan for](https://www.answerstack.io/q/what-ucaas-bandwidth-requirements-should-businesses-plan-for).\n\nThe same arithmetic gives a rough figure for data use. At 87.2 kbps, a G.711 call moves about 39 megabytes an hour in each direction, or roughly 78 megabytes for both. At 31.2 kbps, a G.729 call moves about 14 megabytes an hour each way. Those totals carry the same Ethernet assumptions as the rates, so a mobile or metered connection with different framing will count somewhat differently.","introHtml":"The per-call figure applies to each call in progress at the same moment, so multiply by calls and not by employees. Suppose ten people in your office are on G.711 calls at once. They need 872 kbps of upload and 872 kbps of download, on a connection that's also carrying everything else the office does. With G.729 the same ten calls need 312 kbps each way.
\nWorking out how many calls happen at once is the harder estimate. We cover it in how many simultaneous calls you need on a SIP trunk. If the calls are part of a wider setup with video and screen sharing, see what UCaaS bandwidth requirements to plan for.
\nThe same arithmetic gives a rough figure for data use. At 87.2 kbps, a G.711 call moves about 39 megabytes an hour in each direction, or roughly 78 megabytes for both. At 31.2 kbps, a G.729 call moves about 14 megabytes an hour each way. Those totals carry the same Ethernet assumptions as the rates, so a mobile or metered connection with different framing will count somewhat differently.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":50},{"id":"b5f10cd0-b3d4-4f22-9b62-4c554edbdb92","sectionKey":"bandwidth_has_to_exist_along_the_whole_path","sectionType":"markdown_section","heading":"Bandwidth has to exist along the whole path","introMarkdown":"A call needs capacity on every stretch between the phone and the provider's voice equipment, and the office connection is only the first stretch. Each stretch has its own framing, so use the allowance that fits each link. The routes differ too.\n\nAn Atlantech Online article from 2016, [BYOB vs. dedicated connection](https://care.atlantech.net/news/posts/byob-vs-dedicated-connection-which-is-best-for-voip), sets out two of them. A business can buy internet access from one company and phone service from another, and then each call crosses the public internet to reach the phone provider's switch. Or the office connection can sit on the same network as the voice switch. In that case, Atlantech wrote, the provider manages the routing equipment on both sides and can prioritize voice traffic.\n\nThe arrangement doesn't change how the per-call figure is worked out. It changes who controls the stretches where voice competes with other traffic. On your own office network that's you, and a large upload or a video meeting can crowd a call even when the monthly plan looks generous.\n\nBandwidth is also only one requirement. RTP, the protocol that carries call audio, doesn't guarantee delivery or timely arrival, according to [RFC 3550](https://www.rfc-editor.org/rfc/rfc3550.html). A link with room to spare can still deliver packets late or out of order. The receiving end can absorb some of that, and too much delay or loss hurts the audio. We explain how the audio stream works in how VoIP works and what to look for in a connection in [what type of internet connection you need for VoIP](https://www.answerstack.io/q/what-type-of-internet-connection-do-i-need-for-voip).","introHtml":"A call needs capacity on every stretch between the phone and the provider's voice equipment, and the office connection is only the first stretch. Each stretch has its own framing, so use the allowance that fits each link. The routes differ too.
\nAn Atlantech Online article from 2016, BYOB vs. dedicated connection, sets out two of them. A business can buy internet access from one company and phone service from another, and then each call crosses the public internet to reach the phone provider's switch. Or the office connection can sit on the same network as the voice switch. In that case, Atlantech wrote, the provider manages the routing equipment on both sides and can prioritize voice traffic.
\nThe arrangement doesn't change how the per-call figure is worked out. It changes who controls the stretches where voice competes with other traffic. On your own office network that's you, and a large upload or a video meeting can crowd a call even when the monthly plan looks generous.
\nBandwidth is also only one requirement. RTP, the protocol that carries call audio, doesn't guarantee delivery or timely arrival, according to RFC 3550. A link with room to spare can still deliver packets late or out of order. The receiving end can absorb some of that, and too much delay or loss hurts the audio. We explain how the audio stream works in how VoIP works and what to look for in a connection in what type of internet connection you need for VoIP.
\n","outroMarkdown":null,"outroHtml":null,"contentJson":{},"configJson":{},"noteMarkdown":null,"noteHtml":null,"sortOrder":60}],"citations":[{"title":"Modify Bandwidth Consumption Calculation for Voice Calls","url":"https://www.cisco.com/c/en/us/support/docs/voice/voice-quality/7934-bwidth-consume.html","excerpt":null,"quoteText":"Total packet size = (L2 header: MP or FRF.12 or Ethernet) + (IP/UDP/RTP header) + (voice payload size) ... Bandwidth = total packet size * PPS","sourceRole":"PRIMARY","verifiedAt":"2026-09-14T00:00:00","supportsText":"The framing model used throughout this answer: 40 bytes of IP/UDP/RTP headers, 18 bytes of Ethernet L2 headers including the 4-byte FCS, 50 packets per second at a 20 ms interval, G.711 voice payload of 160 bytes giving 87.2 kbps over Ethernet and G.729 payload of 20 bytes giving 31.2 kbps, and the ","domain":"cisco.com","publisherName":"Cisco"},{"title":"RFC 3550: RTP: A Transport Protocol for Real-Time Applications","url":"https://datatracker.ietf.org/doc/html/rfc3550","excerpt":null,"quoteText":"The first twelve octets are present in every RTP packet ... participant sends audio data in small chunks of, say, 20 ms duration.","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-14T00:00:00","supportsText":"RTP adds a fixed 12-octet header to every packet and carries audio in small chunks of about 20 ms, which is the basis for the 50 packets per second rate.","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"ITU-T G.711: Pulse code modulation (PCM) of voice frequencies","url":"https://www.itu.int/rec/T-REC-G.711/","excerpt":null,"quoteText":"Pulse code modulation (PCM) of voice frequencies. Status: In force.","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-14T00:00:00","supportsText":"G.711 defines pulse code modulation of voice frequencies at 64 kbit/s, the uncompressed narrowband reference used for the 160-byte payload figure.","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":null,"quoteText":"7 kHz audio-coding within 64 kbit/s. Status: In force.","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-14T00:00:00","supportsText":"G.722 codes audio up to 7 kHz within a 64 kbit/s envelope, which is why HD voice costs the same per-call bandwidth as G.711 in this table.","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":null,"quoteText":"Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear prediction (CS-ACELP). Status: In force.","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-14T00:00:00","supportsText":"G.729 codes speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear prediction, the payload bitrate behind the 20-byte packet payload and 31.2 kbps wire rate.","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":null,"quoteText":"The Opus codec scales from 6 kbit/s narrowband mono speech to 510 kbit/s fullband stereo music ... The maximum representable length is 255*4+255=1275 bytes. For 20 ms frames, this represents a bitrate of 510 kbit/s, which is approximately the highest useful rate for lossily compressed fullband stereo music.","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-14T00:00:00","supportsText":"Opus scales from 6 kbit/s narrowband mono speech to 510 kbit/s fullband stereo music, and the 510 kbit/s ceiling corresponds to the maximum representable Opus frame of 1,275 bytes at 20 ms, which is why the ceiling describes music rather than a voice call.","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"RFC 7587: RTP Payload Format for the Opus Speech and Audio Codec","url":"https://datatracker.ietf.org/doc/html/rfc7587","excerpt":null,"quoteText":"For a frame size of 20 ms, these are the bitrate “sweet spots” for Opus in various configurations: 8-12 kbit/s for NB speech, 16-20 kbit/s for WB speech, 28-40 kbit/s for FB speech ... If no value is specified, the default is 20.","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-14T00:00:00","supportsText":"Recommended Opus bitrates for speech at a 20 ms frame size (8 to 12 kbit/s narrowband, 16 to 20 kbit/s wideband, 28 to 40 kbit/s fullband), the default ptime of 20 ms, and the 6000 to 510000 maxaveragebitrate range negotiated in SDP.","domain":"datatracker.ietf.org","publisherName":"IETF"},{"title":"Prepare your organization's network for Teams","url":"https://learn.microsoft.com/en-us/microsoftteams/prepare-network","excerpt":null,"quoteText":"Audio One-to-one 10/10 58/58 76/76 ... Minimum, Recommended, and Best performance bandwidth requirements are based on per-endpoint usage.","sourceRole":"PRIMARY","verifiedAt":"2026-09-14T00:00:00","supportsText":"Microsoft's per-endpoint planning figures for Teams: one-to-one audio 10/10 kbit/s minimum, 58/58 recommended, 76/76 best performance, one-to-one video 1,500/1,500 recommended, all stated as per-endpoint usage up and down; plus Microsoft's guidance that Wi-Fi needs its own planning and QoS and that ","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":null,"quoteText":"The minimum internet connection speed required for VoIP is between 90 kbps to 156 kbps ... Your phones need a consistent, reserved amount of bandwidth for excellent call and fax quality.","sourceRole":"PRIMARY","verifiedAt":"2026-09-14T00:00:00","supportsText":"Intermedia's stated per-call planning range of 90 to 156 kbps, its 90 kbps per phone and 100 kbps per fax adapter example with matching upload and download minimums, and its instruction to reserve bandwidth for the phones rather than rely on available capacity.","domain":"support.intermedia.com","publisherName":"Intermedia"},{"title":"ITU-T G.114: One-way transmission time","url":"https://www.itu.int/rec/T-REC-G.114/","excerpt":null,"quoteText":"Regardless of the type of application, it is recommended to not exceed a one-way delay of 400 ms for general network planning ... if delays can be kept below this figure [150 ms], most applications, both speech and non-speech, will experience essentially transparent interactivity.","sourceRole":"INDEPENDENT","verifiedAt":"2026-09-14T00:00:00","supportsText":"G.114 (05/2003) recommends not exceeding 400 ms of one-way delay for general network planning and states that if mouth-to-ear delay is kept below 150 ms, most applications experience close to transparent interactivity. Verified against the free ITU PDF of the 05/2003 edition, clause 4.","domain":"itu.int","publisherName":"International Telecommunication Union (ITU-T)"},{"title":"VoIP Network Requirements: Bandwidth, QoS, and Infrastructure","url":"https://www.ipcomms.net/blog/voip-network-requirements/","excerpt":null,"quoteText":"For G.711, multiply your peak concurrent calls by 100 kbps (with some overhead margin) ... 20 simultaneous calls need approximately 2 Mbps of dedicated bandwidth in each direction.","sourceRole":"SUPPORTING","verifiedAt":"2026-09-14T00:00:00","supportsText":"A provider planning rule scoped to G.711: multiply peak concurrent calls by 100 kbps including an overhead margin, with 20 simultaneous calls at approximately 2 Mbps in each direction, capacity needed in both directions, and upload as the usual bottleneck on asymmetric connections.","domain":"ipcomms.net","publisherName":"IPComms"},{"title":"VoIP Bandwidth Calculation","url":"https://voipstudio.com/blog/voip-bandwidth-calculation/","excerpt":null,"quoteText":"Short packets travel quicker over the network. If a packet goes missing, you don't lose as much data as with a bigger packet. But shorter packets also require more bandwidth due to packet overhead ... encryption imposes additional bandwidth requirements on each VoIP call.","sourceRole":"SUPPORTING","verifiedAt":"2026-09-14T00:00:00","supportsText":"The packetization trade-off (shorter packets lower delay and lose less audio per loss but carry a higher share of overhead, with most services settling on 20 or 30 ms) and the statement that encryption adds bandwidth to each VoIP call.","domain":"voipstudio.com","publisherName":"VoIPstudio"},{"title":"How Much Data Does VoIP Use? Tips To Save Bandwidth","url":"https://www.nextiva.com/blog/voip-data-usage.html","excerpt":null,"quoteText":"G.711: 87.2 Kbps ... 613 MB [1-hour call] ... G.729: 31.2 Kbps ... 219 MB [1-hour call]","sourceRole":"SUPPORTING","verifiedAt":"2026-09-14T00:00:00","supportsText":"Cited as the example of the circulating hourly totals this answer corrects. The page's per-call rates of 87.2 kbps for G.711 and 31.2 kbps for G.729 match Cisco; its hourly volumes of 613 MB and 219 MB do not, because they divide the hourly bit count by 1,024 without first dividing by 8 to convert b","domain":"nextiva.com","publisherName":"Nextiva"}],"revisions":[{"revisedAt":"2026-07-18T03:24:27.71+00:00","versionLabel":"Originally published","note":"","reviewedBy":null}],"relatedAnswers":[{"id":"a96e1365-05b5-48e7-94dd-91a34d711268","slug":"does-voip-require-an-internet-connection","question":"Does VoIP require an internet connection?","publishedAt":"2026-07-03T03:26:50.582","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":"Not always. VoIP needs an IP network path between the phone and the system that handles its calls. That path can be the public internet, a private connection to the provider or an office network with the phone system on site, depending on how the service is set up.","url":"/q/does-voip-require-an-internet-connection"},{"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":"2026-06-25T03:27:41.235","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":"VoIP needs no particular connection type. It needs an IP connection to your voice provider with room in each direction for the simultaneous calls you plan to support, plus your other traffic. One G.711 call takes about 87 to 100 kbps each way. Also decide whether calls cross the public internet or the provider's network.","url":"/q/what-type-of-internet-connection-do-i-need-for-voip"},{"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":"2026-05-02T03:33:04.199","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":"No. VoIP needs a data connection and a compatible device or app, not a traditional analog phone line. Calling ordinary phone numbers also takes a service connected to the public telephone network. Check your internet, number and connected devices before cancelling the old line.","url":"/q/do-i-need-a-phone-line-for-voip"},{"id":"09cc8e4a-38e6-4d0f-af1c-342b8e394204","slug":"what-is-voip","question":"What is VoIP?","publishedAt":"2026-04-23T03:09:30.888","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, voice over Internet Protocol, is the group of technologies that carries voice calls as data over a network using internet protocols. It describes how a call travels, not a product. It is sold in several forms, including hosted phone systems, SIP trunks and calling inside collaboration apps.","url":"/q/what-is-voip"}],"contributorStats":{"verifiedAnswers":149,"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":"2026-05-09T03:24:27.71","about":[{"@type":"Thing","name":"VoIP"},{"@type":"Thing","name":"Telecommunications"}],"acceptedAnswer":{"@type":"Answer","text":"Cisco's published calculation puts a VoIP call using the G.711 codec at 87.2 kbps in each direction, and a call using the compressed G.729 codec at 31.2 kbps. Both figures count Ethernet framing and assume the audio is sent in packets that each hold 20 milliseconds of speech. A different codec, packet size or type of link gives a different number, and some links add overhead of their own. So the useful thing to know is how the figure is built.","url":"https://www.answerstack.io/q/how-much-bandwidth-does-a-voip-call-use","upvoteCount":0,"datePublished":"2026-05-09T03:24:27.71","dateModified":"2026-09-14T00:00:00","citation":[{"@type":"CreativeWork","name":"Modify Bandwidth Consumption Calculation for Voice Calls","url":"https://www.cisco.com/c/en/us/support/docs/voice/voice-quality/7934-bwidth-consume.html"},{"@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.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 7587: RTP Payload Format for the Opus Speech and Audio Codec","url":"https://datatracker.ietf.org/doc/html/rfc7587"},{"@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":"ITU-T G.114: One-way transmission time","url":"https://www.itu.int/rec/T-REC-G.114/"},{"@type":"CreativeWork","name":"VoIP Network Requirements: Bandwidth, QoS, and Infrastructure","url":"https://www.ipcomms.net/blog/voip-network-requirements/"},{"@type":"CreativeWork","name":"VoIP Bandwidth Calculation","url":"https://voipstudio.com/blog/voip-bandwidth-calculation/"},{"@type":"CreativeWork","name":"How Much Data Does VoIP Use? Tips To Save Bandwidth","url":"https://www.nextiva.com/blog/voip-data-usage.html"}]}}}