ChecklistVoIPPDF · 2.9 MB

VoIP Call Quality Troubleshooting Checklist

Diagnose and fix VoIP call quality issues — jitter, latency, packet loss, echo, and one-way audio with systematic troubleshooting procedures.

About This Resource

Poor VoIP call quality — choppy audio, delays, echo, dropped calls — is almost always a network issue, not a VoIP provider issue. This troubleshooting checklist provides a systematic approach to diagnosing and resolving the most common call quality problems. Covering jitter, latency, packet loss, echo, one-way audio, and registration failures, it guides you through network diagnostics, QoS verification, and targeted fixes to restore clear, reliable business communications.

What's Included

  • Symptom-based diagnostic flowcharts for choppy audio, delays, echo, and dropped calls
  • Network testing procedures for jitter, latency, and packet loss measurement
  • QoS verification checklist across switches, routers, and firewall configurations
  • Bandwidth analysis comparing available capacity against VoIP requirements
  • Hardware troubleshooting for desk phones, headsets, and PoE switches
  • Escalation criteria defining when to involve your VoIP provider vs fixing internally

Who Is This For?

IT support staff, network administrators, and helpdesk teams who need to systematically diagnose and resolve VoIP call quality issues across the business.

Frequently asked questions

Choppy or robotic audio usually points to packet loss or jitter on the network path, often caused by an under-provisioned internet connection, missing QoS configuration, or Wi-Fi interference for wireless handsets. Running a network test during an active call helps isolate whether the issue is local network congestion or the wider internet connection.

One-way audio is typically a firewall or NAT configuration issue that blocks the return path of the media stream, particularly common after firewall firmware updates or when SIP ALG features interfere with call setup. Disabling SIP ALG and confirming the correct RTP port range is open usually resolves it.

Run a wired jitter, latency, and packet loss test during business hours and compare results against your provider recommended thresholds, typically under 150ms latency and 1 percent packet loss. If your network tests clean but issues persist, the problem is more likely on the provider or carrier side.

Yes, this checklist includes escalation criteria to help distinguish issues you can fix internally, such as router QoS settings, from problems that genuinely need escalating to your VoIP provider or carrier for investigation.

Technology Stack

Powered by industry-leading technologies including SolarWinds, Cloudflare, BitDefender, AWS, Microsoft Azure, and Cisco Meraki to deliver secure, scalable, and reliable IT solutions.

SolarWinds
Cloudflare
BitDefender
AWS
Hono
Opus
Office 365
Microsoft
Cisco Meraki
Microsoft Azure

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