Network Requirements for VoIP

Why Network Quality Matters for VoIP

Unlike traditional phone lines, VoIP calls travel as data packets over your internet connection. This means the quality of your calls is directly tied to the quality of your network. A slow or poorly configured network can cause one-way audio, dropped calls, echo, and choppy voice — all of which are frustrating for your customers and staff.

The good news is that VoIP does not require a lot of bandwidth. A well-configured network with a modest internet connection can support excellent call quality. The key factors are bandwidth, latency, packet loss, and correct router settings.

Bandwidth Requirements

Each active phone call uses a small but consistent amount of bandwidth. The amount varies depending on the audio codec your system uses:

To calculate the bandwidth your office needs, multiply the number of simultaneous calls you expect by the per-call bandwidth. For example, a 10-person office where 5 people might be on calls at once using G.711 needs approximately 500 kbps (0.5 Mbps) dedicated to voice traffic.

As a general guideline for DVGVoIP customers:

• Minimum recommended: 10 Mbps download / 5 Mbps upload for up to 5 concurrent calls
• Recommended for most small offices: 25 Mbps download / 10 Mbps upload
• Offices with 10+ concurrent calls: 50 Mbps or more with a dedicated voice VLAN

Note: These figures are for voice traffic only and assume the rest of your internet usage (web browsing, email, file transfers) runs alongside. If your office does heavy file uploads or video conferencing simultaneously, plan for additional headroom.

Latency and Packet Loss

Bandwidth is only part of the picture. Two other metrics have a major impact on call quality:

Latency is the delay between when audio is sent and when it arrives. For VoIP, one-way latency under 150ms is ideal. Above 200ms, callers start noticing a delay that makes conversation awkward. Latency is measured in milliseconds (ms) and is typically shown as "ping" in speed tests.

Packet loss is the percentage of audio packets that are dropped in transit. Even 1–2% packet loss causes noticeable audio quality degradation — choppy voice, missing syllables, and robotic-sounding audio. Packet loss above 5% makes calls nearly unusable.

You can test your current latency and packet loss using the free VoIP readiness test at /voip-readiness-test.

Starlink and Satellite Internet

DVGVoIP serves many Northern Ontario businesses in areas where Starlink or other satellite internet is the only available option. Here is what to expect:

• Starlink latency is typically 20–40ms — acceptable for VoIP and significantly better than traditional geostationary satellite
• Starlink bandwidth (50–200 Mbps download) is more than sufficient for voice traffic
• Packet loss during brief obstructions can occasionally cause short audio interruptions
• Traditional geostationary satellite (e.g., HughesNet, Xplornet legacy) has latency of 600ms or more and is not suitable for VoIP

If you are using Starlink and experiencing call quality issues, the most common cause is obstructions (trees, buildings) causing intermittent packet loss rather than the connection itself. Check your Starlink app for obstruction data.

Quality of Service (QoS)

Quality of Service (QoS) is a router feature that prioritizes certain types of network traffic over others. When QoS is configured for VoIP, your router ensures that voice packets are processed first, even when your internet connection is busy with other traffic — file downloads, video streaming, backups, and so on.

Without QoS, a large file download can temporarily consume your upload bandwidth and cause audio dropouts on active calls. With QoS, voice traffic is protected regardless of what else is happening on the network.

To configure QoS for DVGVoIP:

1. Log into your router's administration panel (typically at 192.168.1.1 or 192.168.0.1).
2. Find the QoS settings — this may be under Advanced, Traffic Management, or a similar label depending on your router brand.
3. Create a high-priority rule for traffic on UDP ports 9000–10999 (RTP audio) and UDP/TCP port 5060 (SIP signalling).
4. If your router supports application-based QoS, look for a "VoIP" or "SIP" category and enable it.
5. Save and apply your changes. Active calls will benefit immediately.

If you are unsure how to configure QoS on your specific router, contact DVGVoIP support — we can advise based on your equipment.

VLAN Separation for VoIP

For offices with 5 or more phones, or any environment where call quality is critical, separating your voice traffic onto a dedicated VLAN (Virtual Local Area Network) is strongly recommended.

A voice VLAN keeps phone traffic completely isolated from your regular computer and guest network traffic. This eliminates competition for bandwidth at the switch level, not just at the router level. It also improves security by keeping your phone system on a separate network segment.

VLAN configuration requires a managed switch and a router that supports VLAN tagging. DVGVoIP can assist with network design if you are setting up a new office or upgrading your infrastructure.

SIP ALG — Disable This Setting

SIP ALG (Application Layer Gateway) is a feature present on many consumer and small-business routers. Despite the name suggesting it helps VoIP, SIP ALG almost universally causes problems including one-way audio, calls that connect but have no sound, and registration failures.

SIP ALG attempts to rewrite SIP packets as they pass through the router, but it typically does this incorrectly, corrupting the packet headers that 3CX and other VoIP systems depend on.

Disable SIP ALG on your router. The setting is usually found in the router admin panel under Advanced, Firewall, or NAT settings. Look for terms like "SIP ALG", "SIP Transformations", "SIP Helper", or "H.323 ALG" and disable all of them.

One-way audio (you can hear the other person but they cannot hear you, or vice versa) is the most common symptom of SIP ALG being enabled. If you experience this, disabling SIP ALG is the first thing to check.

Firewall Ports Required for 3CX

If your office has a firewall, the following ports need to be open for DVGVoIP and 3CX to function correctly. Most modern routers handle outbound traffic automatically, so these rules are primarily relevant if you have an enterprise firewall (e.g., Fortinet, Cisco, pfSense) with strict outbound rules.

Tip: If you are using a strict firewall and calls work internally but fail when employees are remote, the 3CX Tunnel (port 5090) is almost always the cause. Ensuring port 5090 is open outbound from your office resolves the majority of remote access issues.

Testing Your Network

Before going live with DVGVoIP, or if you are troubleshooting call quality issues, use the free VoIP readiness test available at /voip-readiness-test. The test checks your bandwidth, latency, jitter, and packet loss and gives a pass/fail result for VoIP suitability.

If your network fails the readiness test, DVGVoIP can help identify the cause and recommend corrective steps — whether that is a router configuration change, QoS adjustment, or an upgrade to your internet plan.

Need Help?

Open a support ticket by emailing support@dvgvoip.com before calling — this helps us resolve your issue faster. Or call (807) 700-7111.