VoIP – from UC to SIP Trunking – offers businesses more flexibility, cost savings, and scalability than traditional phone systems, but these advantages don't matter if the call quality is poor. Choppy audio and dropped calls are more than just annoying technical glitches; they’re business disruptors. These VoIP call quality issues, whether caused by latency, jitter, or packet loss, can quickly erode customer trust and frustrate teams. A consistent VoIP quality test strategy ensures that you spot problems early and keep communication clear.
Delivering consistently great VoIP call quality requires a combination of proper monitoring, network optimization, and a proactive strategy. This guide covers how to measure VoIP quality, what causes common issues, and seven proven strategies to improve your VoIP network’s performance.
Why Is VoIP Call Quality Important?
VoIP call quality refers to the stability and clarity of a voice conversation carried over IP networks. VoIP relies on the internet and its network infrastructure, making it more flexible than traditional phone systems but also more vulnerable to issues like latency, jitter, and packet loss.
For example, ECG worked with a retail bank chain with a large Cisco-based VoIP PBX platform. The PBX worked fine, but the underlying network meant that calls were often difficult to understand. When you're talking to your mortgage banker, you want to be sure he knows when you said to lock in the rate.
VoIP Call Quality Issues
Network Bandwidth and Congestion
Network bandwidth and congestion directly affect how smoothly voice data flows. Bandwidth refers to the overall capacity of your network, while congestion occurs when demand temporarily exceeds that capacity. When networks are overloaded, VoIP calls may experience delays, audio distortion, or dropped connections.
Latency
Latency measures the delay between when someone speaks and when the other party hears them. While all networks have some latency, delays greater than 150 milliseconds begin to disrupt the natural flow of conversation. Higher latency not only slows down communication but can also make echoes more noticeable.
Jitter
Jitter describes inconsistent packet arrival times. When packets arrive unevenly, devices must buffer the data or risk losing parts of the conversation. Unlike latency, which primarily affects conversational timing, jitter can make audio choppy or even incomprehensible.
Packet Loss
Packet loss occurs when data packets fail to reach their destination. Even small amounts of packet loss can create audio gaps, missing words, or complete communication breakdowns making conversations frustrating and unreliable.
Codec Selection and Compression
The choice of codec and compression settings determines how voice data is encoded, transmitted, and decoded. Low-quality codecs or overly aggressive compression can save bandwidth but often sacrifice call clarity. Business-grade codecs are designed to balance efficiency with superior audio quality.
Endpoint Device Performance
VoIP performance also depends on the capabilities of the devices in use. If an endpoint device, such as a phone, headset, or router, cannot meet the technical demands of high-quality voice communication, users may experience distortion, delays, or dropped calls—even when the network is stable.
How Does VoIP Quality of Service Factor In?
Whether you're an enterprise running a large contact center or a service provider supporting thousands of endpoints, maintaining strong VoIP quality of service (QoS) is business-critical. Enterprises simply cannot use a service provider with questionable VoIP call quality.
To meet this need, many voice service providers offer SLA-backed VoIP services where call quality and uptime aren’t just features but contractual commitments.
How To Perform a VoIP Quality Test
Before you can improve your VoIP quality, you need to measure it first. A VoIP quality test provides valuable insights into your network’s performance, helping you detect and resolve potential issues before they impact users. However, these tests require careful planning and the right tools to be effective.
1. Synthetic Call Testing
Synthetic call testing simulates VoIP calls over your network to evaluate performance under real conditions. These test calls are generated by the monitoring tool itself, so end users don’t experience any disruptions. This method helps identify latency, jitter, and packet loss before they affect actual conversations.
2. Passive Monitoring Tools
Passive monitoring involves observing live network traffic without interfering with it. These tools continuously track key performance metrics like jitter, packet loss, and latency to provide a real-time view of network health. It’s a great option for ongoing monitoring and quick troubleshooting.
3. Call Detail Records (CDRs)
CDRs contain detailed information about each VoIP call, including call duration, origin, destination, and quality metrics. Reviewing this historical data helps uncover trends and recurring performance issues. Some advanced CDR systems even include integrated call quality scores for deeper analysis.
4. End-User Feedback
End-user feedback remains one of the most direct ways to assess VoIP performance. Encourage users to report issues such as dropped calls, echo, or poor audio quality. Consistent patterns in feedback can point to underlying network or configuration problems.
5. MOS Score VoIP (Mean Opinion Score)
Another widely used benchmark for VoIP call quality is the MOS (Mean Opinion Score). This standard rates voice clarity on a scale from 1 to 5, with 5 being excellent and 1 being unusable. In practice, a MOS above 4.0 indicates strong call quality, while anything below 3.5 suggests that users will notice issues. Including MOS in your VoIP quality tests gives you a clear, consistent way to measure and track improvements over time.
6. Common Tools for VoIP Quality Testing
- Many voice engineering teams rely on specialized tools to maintain visibility and ensure consistent VoIP performance. Popular options include:
- Wireshark – for packet-level network analysis.
- PRTG Network Monitor – for comprehensive network and VoIP monitoring.
- VoIP Monitor or Oracle Communications Operations Monitor (OCOM) – for dedicated VoIP performance tracking.
These tools help teams measure, visualize, and improve call quality across their VoIP environments.
Troubleshooting VoIP Call Quality Issues
1. Determine the Business Impact
When call quality degrades, first assess whether the problem is isolated to a single user (e.g., a remote employee with Wi-Fi issues) or affecting teams across locations. Widespread problems can disrupt customer conversations and internal collaboration, making quick identification essential for business continuity.
2. Assess Network Performance
Business VoIP depends on stable internet performance. Measure latency, jitter, and packet loss using tools like Wireshark, PRTG, or enterprise-grade VoIP monitoring platforms. Check for bandwidth saturation during peak hours—especially if video conferencing or large data transfers compete with voice traffic.
3. Verify Endpoints and Devices
Even in a corporate environment, outdated desk phones, overloaded softphones, or poor headsets can compromise audio clarity. Regularly update firmware, confirm devices meet business VoIP requirements, and standardize hardware across teams to avoid inconsistent experiences.
4. Review Codec and Compression Policies
Enterprises often balance bandwidth efficiency with call clarity. Ensure that codecs such as G.711 or G.722 are consistently used across the network, and avoid unnecessary transcoding, which can degrade audio. For high-value customer interactions, configure policies to favor less compression for maximum clarity.
5. Audit QoS and Security Configurations
Poorly configured Quality of Service (QoS) rules or firewalls can cause voice packets to drop behind other data traffic. Confirm that routers and switches prioritize VoIP, and that firewall and NAT settings don’t block or delay SIP/RTP traffic. This ensures voice calls remain reliable even under heavy network load.
6. Replicate Issues With Test Calls
For enterprise VoIP, synthetic test calls are valuable for pinpointing recurring problems without waiting for user complaints. Compare performance across office locations, remote worker VPNs, and ISP connections to isolate the source of disruptions.
7. Escalate With Actionable Data
If the root cause lies outside your internal network, escalate to your VoIP provider or ISP. Present quantifiable data—packet loss percentages, MOS scores, or traceroute paths, so they can quickly diagnose and resolve the issue. Providers offering SLA-backed services will act faster when presented with detailed metrics.
7 Strategies To Improve VoIP Call Quality
Once you’ve identified your VoIP performance issues, it’s time to take action. Here are seven strategies to boost VoIP call quality across your network:
1. Prioritize VoIP Traffic With QoS Configuration & Traffic Engineering
Voice packets are time-sensitive since the timeline from when they are transmitted to when they are received matters. However, they can experience delays when competing with large data transfers, like file downloads or video streaming.
Implementing VoIP quality of service policies ensures voice traffic is prioritized across your network routers and switches. You can also use DSCP (Differentiated Services Code Point) tagging to mark VoIP packets for priority handling and avoid performance issues during high-traffic periods. This can work well for parts of the network you directly control but has limited success across the public internet.
If you depend on the internet, you don't have the option of using prioritization – as in, you can't insist that every ISP enforces your requests. You have to use other techniques, like traffic engineering, to choose which ISP to use for a particular customer.
Sometimes, QoS packet prioritization in your network isn't the challenge: it's packet performance in your client's network. ECG once worked with a service provider who had a client with a large multi-branch network architecture. The internal network between sites had substantial packet loss, but the service provider had no visibility to fix that problem. As a VoIP service provider, your role may be simply to advise clients in proper QoS design.
2. Monitor and Optimize Network Bandwidth
When it comes to bandwidth, it isn’t just about having more – it’s about using it efficiently. Effective bandwidth management can transform your VoIP network from a bottleneck into a high-performance communication platform.
Limited bandwidth is a common cause of degraded VoIP quality, so be sure to evaluate your usage patterns and ensure your network can support data and voice traffic simultaneously. For organizations experiencing congestion during peak hours, consider increasing your network or internet capacity or limiting bandwidth-heavy apps during business hours, such as scheduling backups to off-hours.
Traffic engineering (TE) is the practice of choosing links for certain network paths, such as routing traffic to client A over GTT and traffic to client B over Lumen. But you need to analyze the traffic flows – through traceroute and BGP data – to assess which links are best for each path.
3. Reduce Jitter With Network Buffering and Routing Optimization
Jitter is one of the most subtle and disruptive challenges in VoIP communication. This phenomenon creates audio experiences that feel disconnected and robotic. Reduce jitter by:
- Providing adequate capacity for peak calling hours.
- Using direct routes and avoiding over-complicated network paths. You can often acquire dedicated links to key peering partners, such as Microsoft for Teams Calling or Bandwidth.com for PSTN calling.
Addressing jitter can help you create smoother, more reliable communication experiences. Larger jitter buffers aren’t always the way to go, as they increase latency. We have seen VoIP service providers fix their echo problems by reducing the size of jitter buffers in key VoIP devices.
4. Minimize Latency With Local Routing and Low-Hop Connections
Latency is the delay between sending and receiving a voice packet. Even minor delays – as low as 150ms – can disrupt the natural flow of conversation and undermine professional interactions. To minimize latency, consider:
- Choosing data centers or cloud providers located close to users.
- Using direct internet access (DIA) instead of standard broadband.
- Eliminating unnecessary network hops or relay servers.
- Leveraging content delivery networks for improved routing.
In global deployments, using regional SIP trunks and distributed SBCs (Session Border Controllers) can also help reduce latency for international calls.
5. Monitor Devices and Endpoints for Performance Issues
Sometimes, VoIP issues aren’t with the network but with the endpoint. Old headsets, outdated firmware, or insufficient system resources can create poor audio experiences. In BYOD environments, some phones may be Android devices running software from a VoIP provider, and these devices can be overloaded if software is installed.
Make sure to regularly update firmware on desk phones and softphone applications, and verify that all your endpoints meet minimum hardware specs for VoIP. Device-level monitoring tools can alert you to performance bottlenecks before users start complaining.
6. Use High-Quality Codecs and Avoid Unnecessary Compression
VoIP codecs determine how voice data is compressed and transmitted. Common options include G.722/AMR-WB (high definition), G.711 (legacy uncompressed), and G.729 (compressed). While G.729 saves bandwidth, it can reduce call clarity – especially if mobile carriers or other intermediate networks apply additional compression.
Codec compatibility requires all parties to cooperate; service providers and enterprises rarely have the freedom to adopt a particular codec unilaterally. While G.711 is the most common option on VoIP networks, those integrating with cellular networks should consider AMR-WB and other codecs. Keep in mind that using an SBC to transcode audio never improves the audio quality.
For high-value interactions, use less compression to maximize clarity and adjust codec settings based on your network conditions.
7. Establish Proactive Monitoring and Alerting
Maintaining VoIP quality requires continuous monitoring. Alerting systems can notify your IT or voice engineers whenever packet loss exceeds a defined threshold, MOS scores fall below acceptable levels, or issues like jitter or latency spikes occur.
Popular monitoring approaches include:
- Creating alert systems for performance thresholds.
- Monitoring comprehensive performance metrics.
- Developing real-time network visualization dashboards.
- Establishing automatic incident response protocols.
Proactive monitoring gives your teams time to fix issues before they escalate and affect users. Plus, analyzing these trends over time can help you identify areas for long-term improvement.
ECG: VoIP Quality of Service Support for Providers and Enterprises
At ECG, we work with service providers, universities, and large enterprises to ensure they deliver consistent, reliable VoIP quality at scale. From call monitoring and network design to QoS configuration and SBC integration, we offer the engineering expertise and hands-on support you need to keep your voice services performing at their best.
Our teams specialize in solving real-world VoIP performance issues, whether they involve SIP signaling, network bottlenecks, or misconfigured endpoints. We can also help you run comprehensive VoIP quality tests and build long-term monitoring strategies that evolve with your infrastructure.
Want better VoIP call quality across your network? Reach out to ECG today to find out how we can help you ensure every call is crystal clear.
VoIP Call Quality FAQs
