ADAM PULSE How to troubleshoot VoIP call quality: packet loss, jitter, latency and network problems
The phone call connects.
But something is wrong.
The customer sounds robotic.
Words disappear.
Audio cuts in and out.
There is an uncomfortable delay between speakers.
The call disconnects.
Or perhaps the most frustrating complaint of all:
“Our phones sound terrible sometimes, but not all the time.”
VoIP call quality problems can be difficult to troubleshoot because the phone system itself may be operating correctly.
The real problem may exist somewhere between the user's phone and the voice platform.
That path can include:
Phone → LAN → Switch → Firewall → WAN → Internet Provider → Internet → Voice Platform
A problem anywhere along that path can affect call quality.
This guide explains how to troubleshoot poor VoIP call quality and how latency, jitter, packet loss, bandwidth, WiFi, firewalls and internet circuits can affect business communications.
What Causes Poor VoIP Call Quality?
Common causes include:
- Packet loss
- Jitter
- High latency
- Network congestion
- Insufficient bandwidth
- WiFi problems
- Firewall configuration
- WAN instability
- ISP problems
- VPN routing
- SD WAN problems
- QoS configuration
- Local network problems
- Endpoint issues
The important lesson is:
Poor VoIP quality does not automatically mean there is a problem with the phone platform.
Start by identifying where quality degradation occurs.
What Is VoIP?
VoIP stands for Voice over Internet Protocol.
Instead of sending voice across a traditional telephone circuit, VoIP converts voice into network traffic.
That traffic travels across IP networks.
This provides enormous flexibility.
But it also means voice quality depends on network performance.
What Network Metrics Affect VoIP Quality?
Four measurements are particularly important:
Latency
Jitter
Packet loss
Bandwidth
For troubleshooting, you should also consider:
- Availability
- Route behavior
- WiFi performance
- Firewall health
- WAN health
- Device performance
What Is VoIP Latency?
Latency is the delay between sending and receiving network traffic.
In a voice conversation, excessive latency can create noticeable conversational delay.
You may hear:
Hello?
pause
Yes, I'm here.
pause
Can you hear me?
The conversation begins feeling unnatural because speakers start talking over one another.
What Causes High VoIP Latency?
Possible causes include:
- Long network paths
- Congestion
- VPN routing
- ISP routing
- Firewall processing
- Network saturation
- Poor wireless connectivity
- Remote user connectivity
The important troubleshooting question is:
Where does latency increase?
What Is Jitter?
Network packets do not always arrive at perfectly consistent intervals.
Jitter describes variation in packet timing.
Imagine packets arriving like this:
Packet
Packet
Packet
Packet
That is relatively consistent.
Now imagine:
Packet
pause
Packet Packet
pause
Packet
That timing variation can affect real time voice quality.
What Does High Jitter Sound Like?
Users may describe:
- Robotic audio
- Choppy audio
- Distorted speech
- Missing words
- Inconsistent voice quality
These symptoms can be intermittent, which makes historical monitoring valuable.
What Is Packet Loss?
Packet loss occurs when network packets fail to reach their destination.
For normal data applications, missing packets can sometimes be retransmitted.
Real time voice is different.
A conversation cannot simply stop and wait for every missing piece of audio to be resent.
That is why packet loss can quickly become noticeable.
What Does VoIP Packet Loss Sound Like?
Users may experience:
- Missing words
- Broken audio
- Robotic speech
- Audio dropouts
- One way communication problems
- Call instability
The more severe the loss, the more obvious the degradation can become.
What Is One Way Audio?
One way audio occurs when:
Person A can hear Person B
but:
Person B cannot hear Person A.
This may be associated with:
- Firewall rules
- NAT
- SIP configuration
- RTP handling
- Network routing
- VPN
- Security devices
One way audio is different from general poor call quality and should be investigated accordingly.
Why Do VoIP Calls Sound Robotic?
Robotic audio is frequently associated with unstable network delivery.
Investigate:
- Packet loss
- Jitter
- WiFi
- Congestion
- WAN performance
- VPN paths
Do not immediately replace the phone or blame the voice provider.
Look at the network conditions during the call.
Why Does VoIP Audio Cut In and Out?
Intermittent audio can result from:
- Packet loss
- WiFi instability
- Network congestion
- WAN instability
- Firewall behavior
- Internet circuit degradation
The key word is:
Intermittent.
If the condition disappears before IT investigates it, real time troubleshooting may reveal nothing.
Historical network evidence becomes extremely valuable.
Why Are VoIP Problems So Difficult to Troubleshoot?
Consider:
10:14 AM
Packet loss begins.
10:16 AM
Call quality deteriorates.
10:18 AM
User ends call.
10:21 AM
Help desk ticket opened.
10:32 AM
Network technician begins testing.
Network is healthy.
The user's complaint was real.
The technician simply missed the event.
Why Does VoIP Work Sometimes but Not Others?
Intermittent quality can be associated with changing network conditions.
Examples include:
- Peak bandwidth usage
- WiFi congestion
- ISP congestion
- Backup jobs
- Cloud synchronization
- SD WAN path changes
- VPN changes
- Circuit instability
Look for patterns.
Does poor call quality occur:
At the same time every day?
At one location?
Only on WiFi?
Only during high network usage?
Only across one carrier?
Patterns narrow the investigation.
How Do You Troubleshoot Poor VoIP Quality?
Start with the affected user.
Determine:
- Which location?
- Which device?
- Wired or WiFi?
- Which call?
- Exact time?
- Inbound or outbound?
- One person or everyone?
- One location or multiple locations?
- Was Zoom or another application also affected?
- Has it happened before?
Exact timestamps are particularly important.
Step 1: Record the Call Time
Do not document:
“Calls sounded bad this morning.”
Document:
“Call quality deteriorated between approximately 10:14 and 10:18 AM Central.”
This allows IT to correlate the call with network monitoring and platform diagnostics.
Step 2: Determine Whether the Problem Is One User or Many
If one user is affected, investigate:
- Endpoint
- Headset
- WiFi
- Local network
- Device performance
If the entire office is affected, investigate:
- WAN
- Firewall
- ISP
- Local infrastructure
- Voice platform
Scope dramatically changes the troubleshooting process.
Step 3: Compare Wired and WiFi
If possible, compare performance.
If wired users are healthy while wireless users experience poor audio, investigate:
- Signal
- Interference
- Access point load
- Roaming
- Channel utilization
- Coverage
WiFi problems can easily appear to users as VoIP problems.
Step 4: Check Packet Loss
Measure packet loss at logical points.
For example:
Local Gateway
Firewall
Carrier Gateway
External Destination
The question is:
Where does packet loss begin?
Step 5: Check Latency
Compare current and historical latency.
If the site normally operates at:
22 ms
and latency suddenly reaches:
180 ms
during poor calls, that change deserves investigation.
Step 6: Check Jitter
For real time communications, jitter deserves specific attention.
Look at:
- Current jitter
- Historical jitter
- Whether jitter increased during the affected call
- Whether multiple users experienced the same condition
Step 7: Check Bandwidth and Congestion
Investigate whether the network was saturated.
Possible causes include:
- Backups
- File transfers
- Cloud synchronization
- Video streaming
- Software updates
- Large downloads
- Other conferencing traffic
Bandwidth may be plentiful most of the day but constrained during specific periods.
Step 8: Check the Firewall
Review:
- WAN interface
- CPU
- Memory
- NAT
- Security inspection
- Routing
- QoS
- SD WAN
- VPN
- Logs
Look specifically at the incident time.
Step 9: Check the Internet Circuit
Review:
- Availability
- Latency
- Packet loss
- Jitter
- Circuit flapping
- Historical incidents
- Carrier performance
The circuit can remain online while providing poor real time application quality.
Step 10: Check SD WAN
If multiple circuits exist, determine:
Which path carried the call?
Did failover occur?
Was one WAN connection degraded?
Did the path change during the call?
Real time sessions can be sensitive to network path changes.
Does QoS Improve VoIP Quality?
Quality of Service can help prioritize delay sensitive traffic on networks where congestion exists.
But QoS should not be treated as magic.
It does not fix:
- Broken circuits
- Severe packet loss
- Poor WiFi
- Carrier outages
- Bad routing
- Insufficient overall capacity
QoS is one part of a larger network design.
Can a Speed Test Diagnose VoIP Problems?
A speed test can provide useful information.
But it is only a snapshot.
Suppose:
Poor call: 10:14 AM
Speed test: 10:45 AM
Excellent result.
That does not tell you what happened during the call.
VoIP troubleshooting requires time correlated evidence.
Why Historical Monitoring Matters for VoIP
Historical monitoring lets IT ask:
What was the network doing during the bad call?
For example:
10:13 AM: normal
10:14 AM: latency rises
10:15 AM: packet loss appears
10:16 AM: jitter increases
10:18 AM: conditions recover
Now the user complaint has technical context.
VoIP Troubleshooting Should Correlate the Call with the Network
This is one of the most important concepts in this article.
Do not investigate:
Voice platform
and:
Network
as completely separate systems.
Correlate them by timestamp.
If the communications platform reports poor call quality at the exact time the WAN monitoring system reports increased packet loss, that correlation is valuable.
What Is the ADAM Pulse Approach to VoIP Troubleshooting?
ADAM Pulse provides the network side of the investigation.
The objective is to help determine:
Was the location online?
Was the gateway reachable?
Was the firewall reachable?
Was the carrier connection healthy?
Was latency elevated?
Was packet loss occurring?
Was the condition intermittent?
Has this happened before?
Combined with information from the communications platform, this creates a much more complete troubleshooting picture.
Stop Troubleshooting Voice and Network Separately
Modern voice depends on modern networks.
When voice quality deteriorates, investigate the complete path.
Endpoint
LAN
WiFi
Firewall
WAN
ISP
Voice platform
The faster you isolate where degradation begins, the faster you can get to the correct team or provider.
Turn “Calls Sound Bad” into Evidence
Users should not need to understand jitter.
They should simply report what they experienced and when.
The monitoring and support team should translate:
“Calls sounded terrible.”
into:
“Packet loss and jitter increased on the site's WAN circuit during the affected call.”
That is actionable.
ADAM Pulse provides managed network monitoring designed to give USA Telecom customers historical visibility into network conditions affecting voice, video and cloud communications.
Know when quality changed.
Know what the network was doing.
Know where to investigate next.
Learn more about ADAM Pulse and talk with USA Telecom about monitoring the network behind your business communications.
Frequently asked questions
What Causes Poor VoIP Call Quality?
Common causes include: The important lesson is: Start by identifying where quality degradation occurs.
What Is VoIP?
VoIP stands for Voice over Internet Protocol. Instead of sending voice across a traditional telephone circuit, VoIP converts voice into network traffic. That traffic travels across IP networks.
What Network Metrics Affect VoIP Quality?
Four measurements are particularly important: For troubleshooting, you should also consider:
What Is VoIP Latency?
Latency is the delay between sending and receiving network traffic. In a voice conversation, excessive latency can create noticeable conversational delay. You may hear:
What Is Jitter?
Network packets do not always arrive at perfectly consistent intervals. Jitter describes variation in packet timing. Imagine packets arriving like this:
What Does High Jitter Sound Like?
Users may describe: These symptoms can be intermittent, which makes historical monitoring valuable.
What Is Packet Loss?
Packet loss occurs when network packets fail to reach their destination. For normal data applications, missing packets can sometimes be retransmitted. Real time voice is different.
What Does VoIP Packet Loss Sound Like?
Users may experience: The more severe the loss, the more obvious the degradation can become.
Why Do VoIP Calls Sound Robotic?
Robotic audio is frequently associated with unstable network delivery. Investigate: Do not immediately replace the phone or blame the voice provider.
Why Does VoIP Audio Cut In and Out?
Intermittent audio can result from: The key word is: If the condition disappears before IT investigates it, real time troubleshooting may reveal nothing.
Sources
- Cisco — Understanding Jitter in Packet Voice Networks. Delay, jitter and packet loss as the determinants of voice quality.
- Cisco — Troubleshoot Packet Drops. Congestion, buffer exhaustion and interface errors as drop causes.
- Zoom — Using the Zoom Dashboard. Administrator access to per-meeting latency, jitter and packet loss.
- IETF — RFC 792: Internet Control Message Protocol. Why devices may rate-limit or deprioritise the ICMP that ping and traceroute depend on.
Monitoring requirements and the controls appropriate to them vary by organization. A single test from a single location at a single moment rarely proves where a fault sits — correlate against history, test from more than one point, and preserve evidence before changing configuration.
USA Telecom Consulting LLC is a Service-Disabled Veteran-Owned Small Business running a 24/7 NOC. We monitor networks, circuits and firewalls for regulated and defense-supply-chain organizations.