By Sagar Shankaran, Founder of CallSphere
Diagnose and fix VoIP call quality issues with expert troubleshooting. Learn MOS scoring, jitter analysis, packet loss remediation, and monitoring.
Key takeaways
Poor call quality costs businesses more than most leaders realize. Research from Metrigy indicates that 67% of customers will hang up and call a competitor if they experience poor audio quality on a business call. For sales teams, a single dropped call or garbled conversation can mean a lost deal worth thousands of dollars.
Yet most organizations take a reactive approach to call quality — they only investigate when someone complains. By that point, the damage is done. Proactive call quality monitoring detects degradation before it impacts customers and provides the data needed to resolve issues quickly.
MOS is the industry-standard measurement of voice quality, rated on a scale of 1 to 5:
flowchart LR
INPUT(["User intent"])
PARSE["Parse plus<br/>classify"]
PLAN["Plan and tool<br/>selection"]
AGENT["Agent loop<br/>LLM plus tools"]
GUARD{"Guardrails<br/>and policy"}
EXEC["Execute and<br/>verify result"]
OBS[("Trace and metrics")]
OUT(["Outcome plus<br/>next action"])
INPUT --> PARSE --> PLAN --> AGENT --> GUARD
GUARD -->|Pass| EXEC --> OUT
GUARD -->|Fail| AGENT
AGENT --> OBS
style AGENT fill:#4f46e5,stroke:#4338ca,color:#fff
style GUARD fill:#f59e0b,stroke:#d97706,color:#1f2937
style OBS fill:#ede9fe,stroke:#7c3aed,color:#1e1b4b
style OUT fill:#059669,stroke:#047857,color:#fff
| MOS Score | Quality Level | User Perception |
|---|---|---|
| 4.3-5.0 | Excellent | Toll quality, indistinguishable from landline |
| 4.0-4.3 | Good | Minor imperfections noticeable only to trained listeners |
| 3.6-4.0 | Fair | Perceptible degradation but conversation flows normally |
| 3.1-3.6 | Poor | Annoying quality, requires concentration to understand |
| 2.6-3.1 | Bad | Very annoying, callers ask to repeat frequently |
| 1.0-2.6 | Unusable | Call should be disconnected and retried |
Target MOS for business calls: 3.8 or higher. Most VoIP systems achieve 4.0-4.3 under normal conditions.
MOS can be measured two ways:
Latency is the time it takes for voice packets to travel from sender to receiver. It is measured in milliseconds (ms).
Sources of latency in a VoIP call:
Jitter is the variation in packet arrival times. If packets arrive at 20ms, 22ms, 18ms, 45ms, 19ms intervals, the jitter is the deviation from the expected 20ms interval.
Jitter buffers compensate for jitter by holding incoming packets briefly before playing them. There are two types:
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Packet loss occurs when voice packets fail to reach the receiver. The impact on call quality is severe because voice is a real-time protocol — retransmission (used for data) adds too much delay.
Types of packet loss:
Collect metrics from every active call in real-time:
Raw per-call metrics need to be aggregated for trend analysis:
Dashboards should surface three views:
CallSphere provides a built-in call quality monitoring dashboard that covers all three views, with automatic alerting when quality drops below configurable thresholds. This eliminates the need to build custom monitoring infrastructure.
Alert thresholds (recommended starting points):
Symptoms: Words cut in and out, speech sounds robotic or digitized
Root causes and fixes:
Symptoms: Callers hear their own voice repeated with a slight delay
Root causes and fixes:
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Symptoms: One party can hear the other, but not vice versa
Root causes and fixes:
Symptoms: Calls connect and audio works, but disconnect after a consistent interval
Root causes and fixes:
Symptoms: Noticeable delay on calls to international destinations, speakers talk over each other
Root causes and fixes:
Quality of Service ensures voice packets receive priority over data traffic:
Wi-Fi introduces unique challenges for VoIP:
A MOS score of 4.0 or higher indicates good quality that most users will find satisfactory. For critical business communications (sales calls, customer support), target a MOS of 4.2 or higher. Scores between 3.6 and 4.0 are acceptable but indicate room for improvement. Any call with a MOS below 3.5 should be flagged for investigation. Keep in mind that the theoretical maximum for VoIP using the G.711 codec is 4.4, and for Opus it is approximately 4.6, due to inherent digitization and compression artifacts.
Run a VoIP-specific network assessment rather than a simple speed test. Tools like VoIP Spear, Onesight, or PingPlotter measure the metrics that matter: latency, jitter, packet loss, and QoS behavior under load. Run the test for at least 24 hours to capture peak-usage periods. Key thresholds: latency under 100ms, jitter under 20ms, packet loss under 0.5%, and upload bandwidth of at least 100kbps per concurrent call. If your network passes these tests, it is ready for VoIP.
For organizations with more than 50 concurrent calls, a dedicated internet circuit for voice is strongly recommended. This eliminates competition between voice and data traffic entirely. For smaller deployments, proper QoS configuration on a shared connection works well. The critical factor is upstream bandwidth — many business internet connections have asymmetric speeds (faster download than upload), and upload congestion is the most common cause of VoIP quality issues.
Intermittent issues are the hardest to diagnose because they are often not present when you investigate. The solution is continuous monitoring: deploy a call quality monitoring system that records metrics for every call. When an issue is reported, correlate the timestamp with your monitoring data to see exactly what the network conditions were. Common causes of intermittent issues include: large file transfers or backups competing for bandwidth (check for scheduled jobs), Wi-Fi interference during peak hours, ISP congestion during business hours, and VPN reconnections that briefly interrupt traffic.
No. Some level of packet loss is inherent in IP-based networks, especially over the public internet. The goal is to minimize it below perceivable thresholds (under 0.5%) and use codecs with good loss concealment (Opus excels here). On a well-configured LAN with QoS, packet loss should be effectively zero. Over the internet, loss varies by path and time of day. Using a dedicated SIP trunk with SLA guarantees (typically less than 0.1% loss) provides the most reliable connectivity.
Written by
Sagar Shankaran· Founder, CallSphere
LinkedInSagar Shankaran is the founder of CallSphere, where he builds production AI voice and chat agents deployed across healthcare, hospitality, real estate, and home services. He writes about agentic AI, LLM engineering, and shipping voice agents that handle real calls in production.
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