VoIP • Network setup

Your VoIP call quality problem is almost certainly a network problem

Q: How much internet speed do I need for 10 VoIP lines?

For 10 concurrent calls using G.711 audio, you need approximately 1 Mbps dedicated to voice traffic. In practice, add headroom for your other business internet usage. A 10-20 Mbps symmetric fibre line is comfortable for a 10-line setup with typical office internet use alongside it.

Q: What is the minimum internet speed for VoIP?

The technical minimum is 100 kbps per concurrent call (G.711) or 30 kbps (G.729). For a single user, any connection above 1 Mbps is sufficient for bandwidth. The real constraints are latency (below 150 ms), jitter (below 30 ms) and packet loss (below 1%), which are about network quality rather than raw speed.

Q: Can VoIP work on ADSL or LTE?

Yes, but with caveats. ADSL upload speeds are often low (1-4 Mbps), which limits the number of concurrent calls. LTE introduces variable latency depending on signal strength and tower load. Both can work for small teams. For more than 5 concurrent calls, a dedicated fibre line is preferable. If you use LTE as a primary or backup, ensure QoS is configured on the LTE router and test jitter separately from download speed.

Q: Does VoIP work over Wi-Fi?

Yes. Modern 5 GHz Wi-Fi with QoS enabled performs well for softphones on laptops and mobile devices. Jitter is typically 2-20 ms over Wi-Fi versus near-zero over a wired connection, but this is within the acceptable threshold. Problems arise in congested office environments with many devices on the same access point or on 2.4 GHz channels. Desk phones should be wired where possible.

Q: What ports do I need to open for VoIP?

SIP signalling typically uses port 5060 (UDP and TCP). RTP media streams commonly use UDP ports 10000-20000, though your specific VoIP provider may use a different range. Check your provider's documentation for their exact port requirements. Most modern routers handle outbound SIP automatically; problems usually occur when deep packet inspection interferes with SIP headers.

Q: What causes choppy or robotic VoIP calls?

Choppy calls are almost always caused by jitter above 30 ms or packet loss above 1%. This typically results from network congestion (too many devices competing for bandwidth without QoS), poor Wi-Fi signal or interference, an overloaded router, or ISP-side packet loss. Run a ping test to your VoIP provider's server and check the jitter and loss figures before investigating other causes.

Q: Is 5 Mbps enough for a small business using VoIP?

For a small business with 1-5 staff and no more than 2-3 concurrent calls, 5 Mbps is sufficient for voice. The constraint is usually upload speed: confirm your 5 Mbps package includes at least 2-3 Mbps upload. If your team also uses cloud software or video calls on the same line, size up to 10-20 Mbps to avoid contention.

Before you call your VoIP provider, check your network. This guide covers exactly what bandwidth, latency and QoS settings your business needs for reliable calls.

Last verified 16 June 2026
By the WhichVoIP editorial team
8 min read

100 kbpsper concurrent call (G.711 uncompressed)

<150 msmaximum one-way latency (ITU-T G.114)

<30 msjitter tolerance before voice quality drops

<1%packet loss threshold for clear audio

The short answer

How much internet speed does a business need for VoIP?

Roughly 100 kbps per concurrent call using standard G.711 audio (or 30 kbps with G.729 compression). A 10-person office with half the team on calls simultaneously needs around 5 Mbps dedicated to voice. Speed alone is not the issue: latency below 150 ms, jitter below 30 ms, and packet loss below 1% matter just as much as raw Mbps.

bandwidth per G.711 call

max one-way latency (ITU-T G.114)

jitter ceiling before drop-outs

packet loss threshold

Business Professional On A Clear Voip Call In A South African Office With A Desk Phone — Good VoIP call quality depends on your network configuration, not just your internet speed package.

What actually affects VoIP call quality

Most providers get blamed for call quality that is entirely within the customer’s control. Four network factors determine whether your calls are clear or choppy.

VoIP converts voice into data packets and sends them across your internet connection in real time. Unlike downloading a file, voice packets cannot wait: they must arrive in the right order, at the right time, with no gaps. When they don’t, you hear choppy audio, echo, one-sided calls or dropped lines.

The four variables that determine call quality are bandwidth (enough capacity for concurrent calls), latency (the time a packet takes to travel), jitter (variation in packet arrival timing) and packet loss (packets that never arrive at all). A business on a 100 Mbps fibre line can still have terrible VoIP calls if any of these four are misconfigured or congested.

How a VoIP call travels from your ISP to your phone

ISP
Fibre / LTE
your line speed

QoS ENABLED
Router
Voice traffic
prioritised

Network
Switch
Voice VLAN

IP Desk Phone
or Softphone
Clear call

Speed matters here
QoS is your lever
VLAN separates voice
Wired preferred

✓

The key insight: A properly configured 20 Mbps line beats a poorly configured 100 Mbps line for VoIP. Speed is a prerequisite; QoS is the real differentiator.

How much bandwidth does your business actually need?

The rule of thumb is 100 kbps per concurrent call using G.711. Most businesses run fewer concurrent calls than they have staff.

G.711 is the default uncompressed codec used by most SA business VoIP platforms. Each active call consumes roughly 100 kbps (including overhead). G.729 compresses the signal to around 30 kbps per call, but not all platforms support it and some introduce subtle audio quality trade-offs at the compression level.

The table below uses G.711 as the baseline and accounts for your other internet traffic running alongside VoIP. Double the figures if your team does regular video conferencing on the same line.

Business size	Concurrent calls (est.)	VoIP bandwidth needed	Recommended package
1–5 users	2–3 calls	~0.5 Mbps	5 Mbps or higher
6–15 users	4–8 calls	~1–2 Mbps	10–20 Mbps
16–50 users	8–25 calls	~2–5 Mbps	20–50 Mbps
51–100 users	25–50 calls	~5–10 Mbps	50 Mbps or higher
100+ users	50+ calls	10+ Mbps (voice only)	100 Mbps dedicated

The recommended package column builds in headroom for your other internet traffic (email, browsing, cloud apps). If your team uses heavy cloud-based software such as ERP or large file sharing, size up by one tier.

Asymmetric lines: Most ADSL and entry-level VDSL packages have far less upload speed than download. VoIP sends audio in both directions, so your upload speed must also meet the concurrent-call requirement. A 20/5 Mbps line (20 down, 5 up) supports around 50 concurrent downloads but only about 40–50 VoIP calls on the upload path.

Minimum network requirements for VoIP

Bandwidth gets the most attention, but latency and jitter kill more calls than insufficient speed does.

ITU-T G.114 (the international standard for voice quality over IP) sets the technical thresholds below. These are the numbers your network administrator should verify before you deploy VoIP and should check again if call quality degrades.

Parameter	Acceptable	Degraded	Unacceptable
One-way latency	<150 ms	150–400 ms	>400 ms
Jitter	<30 ms	30–50 ms	>50 ms
Packet loss	<1%	1–2%	>2%
Bandwidth per call (G.711)	100 kbps	80 kbps (compression artefacts)	<64 kbps
Bandwidth per call (G.729)	30 kbps	25 kbps	<20 kbps

You can test your current network’s latency, jitter and packet loss using a VoIP-specific speed test or by running a ping -c 100 to your VoIP provider’s server. Both packet loss and average jitter will show in the results. If you see packet loss above 0.5% or jitter above 20 ms consistently, investigate your router, cabling or ISP before changing any VoIP settings.

Upgrading your VoIP plan won’t fix a jitter problem. A packet that arrives 80 ms late sounds like a click regardless of whether you’re paying R200 or R2000 a month.

WhichVoIP editorial view

How to set up QoS for VoIP

Quality of Service (QoS) tells your router to prioritise voice packets over all other traffic. It is the single highest-impact change you can make to improve call quality.

Without QoS, your router treats a VoIP packet exactly the same as a large file download or a video stream. When a team member starts a 4K YouTube video or a backup starts syncing to the cloud, voice packets queue behind data packets and latency spikes. QoS puts voice packets at the front of that queue.

Three ways to implement QoS

Approach	How it works	Requires
DSCP marking	VoIP device marks packets with Differentiated Services Code Point EF (Expedited Forwarding); router reads the tag and prioritises accordingly. Industry standard.	DSCP-aware router (most business-grade units)
Port-based QoS	Router prioritises traffic on SIP ports (5060 UDP/TCP) and RTP ports (10000–20000 UDP). Simpler to configure.	Router with port-based QoS rules
Bandwidth reservation	Reserve a fixed share of upload/download capacity for VoIP devices (e.g. reserve 20 Mbps for voice VLAN). Blunt but reliable.	Managed switch and VLAN support

For most SA SMEs, DSCP marking at the router is the correct starting point. Your VoIP provider’s hardware or softphone client will already tag packets with the EF mark; you only need to ensure your router honours those tags rather than stripping them.

✓

Quick check: Log into your router’s admin panel and search for “QoS”, “Traffic Control” or “Bandwidth Management”. If those menus are absent, your router does not support QoS and upgrading to a business-grade unit will have a bigger impact than upgrading your line.

Separate your voice traffic with a VLAN

A Voice VLAN isolates VoIP traffic at the network switch level, keeping it separate from your general office data traffic. This eliminates contention between voice and data packets at the switch layer, not just at the router. If you have a managed switch (and most business-class setups do), creating a dedicated voice VLAN is worth the 30-minute configuration time.

Business Router And Network Switch With Ethernet Cables For Voip Setup — A managed switch with a dedicated voice VLAN prevents data traffic from competing with VoIP packets at the switch layer.

Wi-Fi vs wired: which is better for VoIP?

Wired ethernet is more reliable for VoIP. Wi-Fi is fine for most calls in a well-configured office, but it introduces jitter that a cable never does.

Wi-Fi is a shared medium: every device in range competes for airtime, and the radio environment changes constantly. In a busy open-plan office with 20+ devices on the same access point, jitter spikes become common during peak usage, which translates directly to audio glitches. Modern Wi-Fi 6 access points with band steering and MU-MIMO reduce this substantially, but they don’t eliminate it.

Factor	Wired ethernet	Wi-Fi (5 GHz)
Jitter	Near-zero (<1 ms)	2–20 ms (varies with load)
Packet loss	<0.1%	0.1–2% (interference-dependent)
Latency consistency	Highly consistent	Variable with congestion
Setup cost	Higher (cabling)	Lower (existing infrastructure)
Mobile / flexible seating	Not suitable	Ideal

Recommendation: wire your desk phones and fixed workstations. Softphones on laptops can use Wi-Fi provided your access points support at least 5 GHz and QoS DSCP marking on the wireless interface. If your current access points are more than five years old, replacing them will have a greater impact on call quality than most router tweaks.

Shared residential Wi-Fi: If your team works from home, make sure they are on their own broadband line rather than shared estate or apartment Wi-Fi. A 20 Mbps shared estate connection serving 40 units will have erratic jitter that no amount of QoS configuration will fix.

Your 6-step VoIP network setup checklist

Work through these in order. Each step takes 10–30 minutes and builds on the previous one.

Confirm your line speed and symmetry
Run a speed test during business hours (not off-peak). Check both download and upload. Your upload speed must support the number of concurrent calls you plan to run. If it falls short, contact your ISP about a symmetric fibre product.

Test latency, jitter and packet loss
Use ping or a VoIP-specific test tool to check your current network metrics against the ITU-T thresholds in the table above. Run the test from a wired device, not over Wi-Fi, to eliminate wireless variables. Repeat during your busiest call hour.

Enable QoS on your router
Log into your router admin panel and enable QoS with DSCP EF prioritisation for voice traffic. If your router has a preset “VoIP” or “Voice” QoS mode, use that. If there is no QoS option, your router needs to be replaced with a business-grade unit.

Create a voice VLAN on your switch
If you have a managed switch, tag your IP phones and VoIP adapters to a dedicated voice VLAN (typically VLAN 10 or VLAN 100 by convention). This prevents data traffic from competing with voice packets at the switch layer.

Wire your desk phones; move softphones to 5 GHz
Patch cable your desk phones to the switch directly. For laptop-based softphone users, ensure they are connected to a 5 GHz access point rather than 2.4 GHz. Check that your access points support DSCP marking on the wireless interface (most enterprise-grade units do).

Check firewall and port configuration
Your firewall must allow outbound SIP (port 5060 UDP/TCP) and RTP media streams (commonly UDP 10000–20000, though your VoIP provider may specify a different range). Stateful inspection is generally fine; deep packet inspection on voice traffic can introduce latency. Confirm the exact port requirements with your provider.

Our verdict

Most SA businesses do not have a VoIP provider problem. They have a network configuration problem. The ITU-T thresholds above are achievable on any business-grade fibre or LTE connection with QoS enabled. A business running 10 concurrent calls on a properly configured 10 Mbps line will have better call quality than one running the same calls on an unconfigured 100 Mbps line.

If you have worked through the 6-step checklist and call quality is still poor, your ISP’s routing to your VoIP provider’s data centre may be the culprit. At that point, comparing VoIP providers makes sense.

Our recommendation: Run the bandwidth and latency tests before contacting your VoIP provider about call quality. Nine times out of ten, the fix is in your router, not your plan. If you are still shopping for a provider, get matched quotes and ask specifically whether they offer local SA media servers.

Ready to compare VoIP providers?

Once your network is configured, use our free matching tool to compare SA VoIP providers on price, features and support.

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Frequently asked questions

How much internet speed do I need for 10 VoIP lines?

For 10 concurrent calls using G.711 audio, you need approximately 1 Mbps dedicated to voice traffic. In practice, add headroom for your other business internet usage. A 10–20 Mbps symmetric fibre line is comfortable for a 10-line setup with typical office internet use alongside it.

What is the minimum internet speed for VoIP?

The technical minimum is 100 kbps per concurrent call (G.711) or 30 kbps (G.729). For a single user, any connection above 1 Mbps is sufficient for bandwidth. The real constraints are latency (below 150 ms), jitter (below 30 ms) and packet loss (below 1%), which are about network quality rather than raw speed.

Can VoIP work on ADSL or LTE?

Yes, but with caveats. ADSL upload speeds are often low (1–4 Mbps), which limits the number of concurrent calls. LTE introduces variable latency depending on signal strength and tower load. Both can work for small teams. For more than 5 concurrent calls, a dedicated fibre line is preferable. If you use LTE as a primary or backup, ensure QoS is configured on the LTE router and test jitter separately from download speed.

Does VoIP work over Wi-Fi?

Yes. Modern 5 GHz Wi-Fi with QoS enabled performs well for softphones on laptops and mobile devices. Jitter is typically 2–20 ms over Wi-Fi versus near-zero over a wired connection, but this is within the acceptable threshold. Problems arise in congested office environments with many devices on the same access point or on 2.4 GHz channels. Desk phones should be wired where possible.

What ports do I need to open for VoIP?

SIP signalling typically uses port 5060 (UDP and TCP). RTP media streams commonly use UDP ports 10000–20000, though your specific VoIP provider may use a different range. Check your provider’s documentation for their exact port requirements. Most modern routers handle outbound SIP automatically; problems usually occur when deep packet inspection interferes with SIP headers.

What causes choppy or robotic VoIP calls?

Choppy calls are almost always caused by jitter above 30 ms or packet loss above 1%. This typically results from: network congestion (too many devices competing for bandwidth without QoS), poor Wi-Fi signal or interference, an overloaded router, or ISP-side packet loss. Run a ping test to your VoIP provider’s server and check the jitter and loss figures before investigating other causes.

Is 5 Mbps enough for a small business using VoIP?

For a small business with 1–5 staff and no more than 2–3 concurrent calls, 5 Mbps is sufficient for voice. The constraint is usually upload speed: confirm your 5 Mbps package includes at least 2–3 Mbps upload. If your team also uses cloud software or video calls on the same line, size up to 10–20 Mbps to avoid contention.

Keep reading

How to troubleshoot choppy VoIP calls
Compare VoIP providers in South Africa
Best VoIP desk phones for SA businesses
VoIP for business: what you need to know

Sources: ITU-T G.114 (2003) “One-way transmission time” (international standard for voice latency); ITU-T G.711 (1988, revised 1993) “Pulse code modulation of voice frequencies” (64 kbps codec standard); ITU-T G.729 (1996) “Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear prediction”. Bandwidth figures represent real-world packet overhead at the transport layer. Verified 16 June 2026.