Network Optimization for VoIP Telephony

Overview

VoIP telephony places different demands on your network than normal web browsing or email. While a 200-millisecond delay on a website is barely noticeable, it causes disruptive dropouts or echo during a phone call. This article shows you how to optimize your network for VoIP — whether in the office or at home.

The three most important factors for VoIP

1. Latency (delay)

Latency is the time it takes for a data packet to travel from your phone to the other party. For smooth phone calls, latency should be below 150 ms (ideally below 80 ms). Higher values lead to noticeable delays — you and your conversation partner talk over each other because responses arrive too late.

2. Jitter (latency variation)

Jitter describes the variation in latency. Even if average latency is good, strong fluctuations can affect voice quality. Jitter should be below 30 ms. High jitter manifests as "choppy" or distorted voices.

3. Packet loss

Packet loss means data packets get lost in transit. For VoIP, packet loss should be below 1%. Even 2–3% packet loss can cause audible dropouts. Individual lost packets are compensated by the codec, but beyond a certain threshold, the conversation becomes unintelligible.

Network optimization step by step

Set up Quality of Service (QoS)

QoS is the most important measure for good VoIP quality. It tells your router to give voice data priority — even when someone is downloading a large file.

1. Open your router's configuration interface.
2. Find the QoS or traffic shaping settings.
3. Create a rule with highest priority for:
   • SIP traffic (Port 5060 UDP/TCP)
   • RTP traffic (Port range 10000–20000 UDP) — these are the actual voice data packets
4. Alternatively: Prioritize all traffic from your phones' IP addresses.

Wired over Wi-Fi

The simplest and most effective measure: connect your IP phones via Ethernet cable. Wi-Fi is generally suitable for VoIP but more susceptible to:

• Interference from other devices (microwaves, Bluetooth, neighboring Wi-Fi networks)
• Signal fluctuations depending on distance and obstacles
• Higher and more irregular jitter

If Wi-Fi is unavoidable (e.g., for softphones on laptops), use the 5 GHz band instead of 2.4 GHz — it's less congested and provides more stable connections.

VLAN for VoIP (advanced)

In larger offices, we recommend setting up a separate VLAN (Virtual LAN) for voice traffic. This physically separates VoIP data from other network traffic.

Benefits:

• Guaranteed bandwidth for voice data
• Protection from network congestion caused by downloads or backups
• Better security (phones in their own network segment)

VLAN setup requires a managed switch. Most professional IP phones (including Yealink) support VLAN tagging.

Disable SIP-ALG

SIP-ALG (Application Layer Gateway) is a feature in many routers that is supposed to help SIP connections — but in practice frequently causes problems. Disable SIP-ALG in your router. See our separate article Disable SIP-ALG in Your Router or Firewall for details.

Calculate bandwidth

How much bandwidth do you need? A rule of thumb:

• Per concurrent call: approx. 100 kbps (upload and download)
• Example: An office with 10 employees, of whom a maximum of 5 are on the phone simultaneously → 500 kbps reserved bandwidth for VoIP

With modern internet connections (50 Mbps and above), raw bandwidth is rarely the problem. What matters more is that bandwidth is consistently available and not completely consumed by other applications (hence QoS).

Troubleshooting

One-way audio or no audio

Most common cause: Firewall blocking RTP packets or SIP-ALG manipulating the connection.

Solution: Disable SIP-ALG, open firewall for SIP/RTP ports, test phone IP in DMZ.

Choppy audio

Most common cause: High jitter or packet loss, often caused by Wi-Fi or network congestion.

Solution: Use wired connection, set up QoS, limit concurrent downloads.

Echo

Most common cause: Acoustic feedback at the device (speaker-microphone loop) or excessive latency.

Solution: Use a headset instead of speakerphone, check latency, reduce phone volume.

Dropped calls

Most common cause: NAT timeout — the router closes the connection after inactivity.

Solution: Enable SIP keepalive in the PBX (enabled by default with Virtual-Call), increase NAT timeout in the router (to at least 300 seconds).

Quick checklist

• ☐ IP phones connected via Ethernet cable
• ☐ QoS enabled in router for SIP/RTP
• ☐ SIP-ALG disabled in router
• ☐ Router and phone firmware up to date
• ☐ If on Wi-Fi: using 5 GHz band
• ☐ Internet connection stable (no packet loss, jitter