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VoIP Call Quality: How to Monitor & 7 Strategies To Improve It

Written by ECG Team | Apr 15, 2025 1:15:00 PM

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.

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.

Several factors play a part in determining VoIP call quality, including:

  • Network bandwidth and congestion determine how smoothly data flows through your systems. "Bandwidth" is the informal term for capacity. "Congestion" occurs when there is a moment in time of insufficient capacity.
  • Latency is a measurement of the delay in packet transmission that can disrupt natural conversation flow. All networks have latency, but latency that is greater than 150 milliseconds (from the person speaking to the person hearing) begins to change the dynamics of the conversation. Larger latency also makes echoes more apparent.  
  • Jitter refers to inconsistent packet arrival times that can make the devices buffer, or lose, some of the audio data. While standard latency merely changes the conversational ease, jitter can make the audio incomprehensible.
  • Packet loss can create data gaps that result in frustrating communication breakdowns.
  • Codec selection and compression settings determine how voice data is compressed and transmitted.
  • Performance is impacted when an endpoint device can't handle the technical demands of high-quality voice communication.

These technical elements translate directly into human experiences. A dropped word, a moment of static, or an unexpected delay can undermine the trust and professionalism your organization works hard to maintain.

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'll need to measure it. A VoIP quality test can offer a diagnostic window into your network's performance that reveals potential issues before they become problems for your users. But beware: these tests can take real effort and planning! 

There are several ways to run a VoIP quality test, including:

  • Synthetic Call Testing: This creates VoIP calls over your network to evaluate performance. Your customers or end-users don’t experience the problem; rather, the testing tool creates the calls.
  • Passive Monitoring Tools: These track real-time metrics like jitter, packet loss, and latency, but without changing the network.
  • Call Detail Records (CDRs): These provide historical performance data to identify trends and recurring issues. Some CDRs can include call quality data.
  • End-User Feedback: This gauges if users are consistently reporting audio issues.

Many voice engineering teams use tools like Wireshark, PRTG, and dedicated VoIP monitoring platforms such as VoIP Monitor or Oracle Communications Operations Monitor (OCOM) to maintain visibility into their VoIP environments.

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.