Do I Need a Dedicated Internet Line for VoIP?

No — most offices do not need a dedicated internet line for VoIP. A G.711 call uses about 87 kbps in each direction once packet headers are counted, so eight simultaneous calls fit inside 700 kbps. Bandwidth is almost never the constraint. Contention on the upload and a misconfigured router usually are.

That holds for the overwhelming majority of small offices. The exceptions are real and specific, and listed further down.

How much bandwidth does one call use?

About 87 kbps in each direction for a G.711 call, the codec most business VoIP uses by default.

The arithmetic is the whole argument:

• G.711 encodes voice at 64 kbps of payload.
• It sends a packet every 20 ms, so 50 packets per second.
• Each packet carries 40 bytes of RTP, UDP, and IP headers — 12 + 8 + 20.
• 40 bytes × 50 packets × 8 bits = 16 kbps of overhead.
• 64 + 16 = 80 kbps at the IP layer, roughly 87 kbps once Ethernet framing is included.

Call it 100 kbps per call per direction and you have a planning number with headroom built in.

Those figures are per direction, and voice is symmetric. Compare them to the connection in front of you: a 200/20 Mbps cable circuit carries sixteen concurrent G.711 calls using 7% of its upload. A phone system is a rounding error on a modern business connection.

Note also that concurrent calls are far fewer than phones. A ten-person office rarely has more than three or four calls up at once; assuming every handset is in use overstates demand by a factor of three.

Do I need a separate internet line for VoIP phones?

Not for bandwidth. The reasons people buy one are almost always mislabeled versions of a different problem.

Here is what usually happens. Calls sound bad. Someone runs a speed test, sees a healthy number, and buys a second circuit because nothing else got fixed. It sometimes helps — not because it added capacity, but because it moved the phones off a saturated link and onto a router that was not mangling SIP. Both are fixable on one circuit for no monthly cost.

Voice does not need much bandwidth. It needs consistent delivery. A link at 30% utilization carries perfect calls; the same link at 90% with a backup running carries terrible ones at the same nominal speed. Speed tests measure throughput, not consistency, which is why they fail to predict call quality.

What actually matters more than bandwidth?

Four things, in the order they cause trouble.

Upload, not download. Most connections are asymmetric. Your outbound audio rides the upload, and so does every backup, file sync, and video call. On a 500/20 circuit, the 20 is where voice competes.

Contention. What else uses the link and when. A nightly backup overlapping a morning call spike is a scheduling problem with a free fix.

Bufferbloat. Consumer routers queue packets deeply under load rather than dropping them, producing latency spikes of hundreds of milliseconds during ordinary use — the condition voice tolerates worst.

SIP handling. SIP ALG, aggressive NAT timeouts, and restrictive UDP rules cause one-way audio and dropped registrations, none of which a bigger circuit affects. VoIP network requirements covers what to set, and why your VoIP calls sound bad covers how to tell these apart by symptom.

Does QoS actually help?

Yes, when the bottleneck is your own router. QoS prioritizes voice packets over everything else in the queue, so a large upload cannot push audio behind it.

The important limit: QoS only works where you control the queue. Your router can prioritize traffic leaving your network; it cannot prioritize anything on your ISP's network or on the public internet. That is why QoS reliably fixes "calls degrade when someone uploads a video" and reliably does nothing for "calls to one area code sound thin."

Configure it where the office has real contention — a small upload, heavy cloud use, a shared link with guest Wi-Fi. Skip it where nothing competes. Marking DSCP EF on voice and giving it strict priority at the router takes about ten minutes on capable gear.

When is a dedicated line genuinely right?

Four cases, and they are narrower than the sales pitch suggests.

1. The existing circuit is technically unsuitable. Satellite, some fixed wireless, and oversubscribed rural DSL carry jitter and loss no amount of QoS repairs. The second line is not extra capacity; it is a better path.
2. You cannot control the shared link. A suite with landlord-provided internet, no router access, no ability to configure anything. Your own circuit buys control, which is the actual product.
3. Concurrent call volume is high relative to the circuit. Forty or more simultaneous calls on a modest business connection is a different arithmetic problem.
4. You want failover, not capacity. This is the good reason, and it gets its own section.

Should I get a second connection for failover?

If a phone outage genuinely stops your business, yes — but buy it as failover, not as a voice line, and buy it from a different provider on different physical infrastructure.

Two circuits from the same cable company down the same conduit fail together. Cable plus fixed wireless, or fiber plus LTE, gives you genuine diversity.

Understand what failover covers. Phones on a dead circuit cannot register, so a second WAN keeps handsets working. It does not help during a power cut, and it does not make VoIP behave like a copper line in an emergency — see keeping the phones answered during an outage.

The cheaper answer, which most small offices should reach for first: routing that fails over rather than hardware that does. Signal is hosted, so when your office goes dark the phone system is still running. Point the main number at a mobile number, a ring group including cell phones, or a voicemail box with voicemail-to-email, and calls keep being answered from wherever people are. Staff can also work from the browser-based web phone or the Signal Phone desktop app on any connection, including a hotspot.

That costs nothing. A second circuit covers the narrower case where the office must keep working on site.

Frequently asked questions

How much bandwidth does a VoIP call use? About 87 kbps in each direction for a G.711 call — 64 kbps of payload plus roughly 16 kbps of RTP, UDP, and IP headers across 50 packets per second, with Ethernet framing on top. Plan at 100 kbps per concurrent call per direction and you have comfortable headroom.

Do I need a separate internet line for VoIP phones? Rarely. Eight simultaneous calls fit in about 700 kbps each way, a fraction of any modern business connection. The problems people buy a second line to solve are usually upload contention, bufferbloat, or SIP ALG — all fixable on the existing circuit for no monthly cost.

How much internet speed do I need for 10 phones? Less than you would guess, because ten phones do not mean ten calls. A ten-person office typically peaks at three or four concurrent calls, under 400 kbps each direction. Even with all ten active, you need roughly 1 Mbps up and down for voice.

Does QoS actually help VoIP call quality? Yes, when the congestion is on your own router's outbound queue — it stops a large upload from pushing audio behind it. It cannot prioritize traffic once packets leave your network, so it fixes "calls degrade when someone uploads" and does nothing for problems originating upstream.

Should I get a second internet connection for phone failover? Only if an outage genuinely stops your business, and only from a different provider on different physical infrastructure — two circuits in one conduit fail together. For most small offices, failover routing to mobile numbers costs nothing and covers the same risk, since a hosted phone system keeps running when your office does not.

If someone has quoted you a second circuit for voice, check the arithmetic against your actual concurrent call volume first. We are happy to work through what your office would really need.