VoIP Call Quality Problems: Jitter, Latency and Packet Loss

What Affects VoIP Call Quality?

VoIP call quality depends on the health of your network connection. Unlike traditional phone lines that have dedicated bandwidth, VoIP calls share your internet connection with all other data traffic. Three key network metrics determine whether your calls sound clear or suffer from quality issues: jitter, latency and packet loss.

Understanding these metrics and how to measure them is essential for diagnosing and fixing call quality problems.

Understanding the Key Metrics

Latency (Delay)

Latency is the time it takes for a voice packet to travel from your phone to the other party. It is measured in milliseconds (ms). High latency causes a noticeable delay in conversation — you say something and the other person hears it a moment later, leading to people talking over each other.

Under 150ms — acceptable for VoIP calls, conversation feels natural
150ms–300ms — noticeable delay, conversation becomes awkward
Over 300ms — unacceptable, calls feel like satellite phone conversations

Jitter (Variation in Delay)

Jitter is the variation in latency between packets. Even if your average latency is low, if individual packets arrive at inconsistent intervals, the audio will sound choppy or distorted. Your phone has a jitter buffer that attempts to smooth out these variations, but it can only compensate for so much.

Under 30ms — acceptable, jitter buffer can compensate
30ms–50ms — may cause occasional audio glitches
Over 50ms — significant audio quality problems

Packet Loss

Packet loss occurs when voice data packets fail to reach their destination. Lost packets result in gaps in the audio — words or syllables go missing, making speech difficult to understand. Even small amounts of packet loss significantly impact call quality.

Under 1% — acceptable, barely noticeable
1%–3% — noticeable audio degradation
Over 3% — calls become unusable

How to Test Your Network

Basic Tests

Start with these simple tests to assess your network health:

Ping test — open a command prompt and run ping your-sip-server.com to check latency and packet loss to your VoIP provider
Traceroute — run tracert your-sip-server.com (Windows) or traceroute your-sip-server.com (Mac/Linux) to identify where delays occur along the network path
Speed test — check your upload and download speeds at speedtest.net — ensure you have sufficient bandwidth for your number of concurrent calls

VoIP-Specific Quality Tests

For more detailed analysis, use VoIP quality testing tools that specifically measure jitter, latency and packet loss:

Ping with statistics — run a sustained ping (e.g. ping -n 100 your-sip-server.com) and check the minimum, maximum and average latency plus packet loss percentage
Your provider's quality test — many VoIP providers offer network quality test tools on their website
Wireshark — for advanced users, capture and analyse RTP streams to measure jitter and packet loss in real-time

Fixing VoIP Call Quality Problems

1. Configure QoS (Quality of Service)

QoS is the single most effective way to improve VoIP call quality. It tells your router to prioritise voice traffic over other data. Configure your router to:

Prioritise traffic on UDP ports 5060 and 10000–20000
Apply DSCP EF (46) marking to voice packets
Limit bandwidth available to non-critical applications during peak usage

See our firewall and network configuration guide for detailed QoS setup instructions.

2. Separate Voice Traffic with VLANs

Place your VoIP phones on a dedicated VLAN separate from your data network. This prevents large file downloads, video streaming or software updates from competing with voice traffic for bandwidth.

3. Upgrade Your Broadband

If your internet connection is consistently saturated, no amount of QoS configuration will fix quality issues. Each concurrent VoIP call requires approximately 100 kbps of bandwidth in each direction. Calculate your requirements:

10 concurrent calls — need at least 1 Mbps upload and download dedicated to voice
50 concurrent calls — need at least 5 Mbps upload and download dedicated to voice

Consider upgrading to a leased line or dedicated business broadband if your current connection cannot support your call volumes.

4. Disable SIP ALG

While SIP ALG is more commonly associated with one-way audio and registration failures, it can also contribute to call quality issues by adding processing delay to SIP packets. Disable SIP ALG on your router.

5. Identify Bandwidth Hogs

Check for applications or devices consuming excessive bandwidth during business hours:

Cloud backup services running during the day
Windows/software updates downloading simultaneously across multiple PCs
Video streaming or large file transfers
Guest Wi-Fi without bandwidth limits

Schedule large downloads for outside business hours and apply bandwidth limits to non-critical services.

Monitoring Ongoing Quality

Call quality issues can be intermittent, so ongoing monitoring is important:

Ask your VoIP provider about call quality reports and MOS (Mean Opinion Score) data
Set up network monitoring to alert you when jitter, latency or packet loss exceeds acceptable thresholds
Keep a log of quality complaints with dates and times to correlate with network events

If you continue to experience call quality issues after following these steps, contact our team for a detailed network assessment.