Latency is the time delay between when someone speaks on a call and when the other party hears them, measured in milliseconds. For acceptable call quality, one-way latency should stay under 150ms. Measuring latency matters because delays above this threshold cause communication breakdowns, customer frustration, and increased churn.
Are you measuring latency on your customer calls? If not, you should be, because high latency directly undermines customer experience and agent effectiveness. Research by IDC states that “global business leaders rate latency a higher priority than speed.”
Unfortunately, despite this, latency during calls remains a common problem. Undoubtedly, these issues can negatively impact your call quality, frustrating both your contact center agents and customers.
Learn how to improve your CX by monitoring your contact center’s inbound and outbound lines with Cyara’s contact number tests.
What is latency?
Latency is the time delay, measured in milliseconds, between when one party speaks on a call and when the other party hears their voice.
Often referred to as ‘lag,’ latency occurs naturally due to the distance between the caller and recipient. Thankfully, these small delays are typically barely noticeable.
However, longer, more evident delays can be introduced through specific factors along the call path that are most likely outside of your team’s internal control. These include geographic distance between callers, network congestion, codec processing time, carrier routing inefficiencies, the number of network hops, and the chain of hardware components that ultimately connects the call.
What are the effects of high latency?
High latency can adversely affect the quality of your voice and video calls:
- Greater communication delays: High levels result in noticeable delays between speakers during voice calls, making conversations feel unnatural. This issue is amplified in video calls, where it can cause distortion issues between audio and video streams, leading to distracting discrepancies between the words spoken and lip movements.
- Reduced real-time interaction: Latency hampers the effectiveness of real-time interactions, posing difficulties for users to engage in fruitful conversations. This issue becomes particularly problematic in situations where quick responses and immediate feedback are crucial, such as during business negotiations or virtual collaboration sessions.
- Increased frustrations: Users are likely to become frustrated with the delays and disruptions it causes. These frustrations not only impact individual experiences but also carry the potential for reputational damage to your organization.
- Diminished productivity: In business environments, high latency in communication and collaboration tools or systems can hinder participation and productivity. It also significantly increases the likelihood of misunderstandings. Unfortunately, this has the potential to reduce the overall efficiency and success of your distributed teams.
- Escalated customer churn: In terms of customer and user applications, high latency has been shown to drive customers away. Research indicates that poor call quality can increase customer churn rates by up to 15% and negatively impact Net Promoter Scores (NPS). This is because people prefer seamless and responsive communication experiences. Not only does this cause reduced customer satisfaction scores and lower brand loyalty, but your audio issues could also be driving customers towards your competitors.
- Exacerbated audio issues: High latency often contributes to or exacerbates packet loss and jitter problems. This will further degrade the quality of your voice and video calls. Packet loss can result in missing audio or video segments, while jitter can cause uneven and choppy playback.
How do you measure latency?
Latency, measured in milliseconds (ms), can be assessed in two ways:
- One-way: This measures the time it takes for a data packet to travel in one direction only. It is commonly used to diagnose network problems.
- Two-way / round trip: This measures the round-trip time and is frequently used to calculate mean opinion scores (MOS).
What are acceptable latency thresholds for VoIP?
| Measurement type | Acceptable range | Impact on call quality |
| One-way latency (standard) | 20 ms | Normal range; imperceptible to callers |
| One-way latency (maximum) | Under 150 ms | Barely perceptible; generally acceptable |
| One-way latency (problematic) | 150 ms+ | Noticeable delays; significant quality decline |
| Round-trip latency (maximum) | Under 300 ms | Acceptable per ITU standards |
| Enterprise applications | 10 ms or less | Required by 90% of business leaders per IDC |
- For VoIP calls, a one-way latency of 20 ms is standard and falls within the normal range.
- According to research by Gregg Communication Systems, latency less than 150 ms is barely perceptible and is generally acceptable. However, they warn that latency should not exceed 150 ms, emphasizing its potential to dramatically reduce call quality and cause issues.
- The International Telecoms Union (ITU) recommends that one-way latency should not exceed 150 ms for voice calls (or 300 ms for round trip). Anything beyond these thresholds can lead to a significant decline in call quality, resulting in incomprehensible, distorted, and non-productive conversations.
- IDC’s research also highlights that 90% of business leaders specify they need a very low latency of 10 ms or less to confirm the success of their applications.
How to reduce latency
To minimize latency on your calls, implement these techniques:
- Implement Quality of Service (QoS) prioritization: Configure your network to prioritize voice traffic over less time-sensitive data.
- Select efficient codecs: Use codec algorithms optimized for low-latency audio transmission.
- Reduce network hops: Minimize the number of routing points between callers to decrease transmission time.
- Optimize carrier routing: Work with carriers to verify efficient call routing paths.
- Upgrade infrastructure: Invest in improved internet infrastructure and faster processing speeds.
- Use adaptive streaming: Implement adaptive streaming technologies that adjust to network conditions in real-time.
Cyara’s testing solutions
Today’s users hold high expectations for call quality, and businesses, such as yours, rely on clear communication channels, especially for customer interactions.
This is why many organizations are now taking a more proactive approach to their audio quality management, including latency. Reducing this is frequently a priority during the design and optimization of communication systems.
Cyara’s latency test empowers you to replicate the customer call flow, and quantify the latency that your customers’ experience. As a result, you have the power to proactively measure any delay on your calls. What’s more, repeated testing allows you to detect where there are variations over time. Additionally, the data generated enables you to make more informed routing decisions. All these aspects empower your team to take swift action in minimizing latency on your lines, ultimately enhancing the experience for both customers and staff.
For further details on Cyara’s latency test, see our contact center number testing page or reach out to our team directly.
Frequently Asked Questions
One-way latency under 150 ms is generally acceptable for VoIP calls, with 20 ms being the standard target. The ITU recommends round-trip latency should not exceed 300 ms.
One-way latency measures the time for data to travel in one direction (speaker to listener), while round-trip latency measures the total time for data to travel to the recipient and back.
High latency is caused by geographic distance between callers, network congestion, codec processing time, carrier routing inefficiencies, and excessive network hops between connection points.
Call latency, often referred to as “lag,” is the time delay between when one party speaks and when the other person hears their voice. Some natural delay always occurs due to the distance between callers, but longer delays can be introduced by network hops and hardware components along the call path.
The International Telecoms Union recommends a one-way latency of no more than 150 ms for voice calls, or 300 ms for a round trip. For VoIP calls specifically, a one-way latency of 20 ms is considered standard and within the normal range.
One-way latency measures the time it takes for a data packet to travel in a single direction and is commonly used to diagnose network problems. Two-way, or round-trip, latency measures the total travel time there and back and is frequently used to calculate Mean Opinion Scores (MOS).
High latency can cause noticeable communication delays, reduced real-time interaction, increased customer frustration, and diminished productivity. It can also exacerbate other audio issues like packet loss and jitter, further degrading call quality.
High latency has been shown to drive customers away, as people prefer seamless and responsive communication experiences. This can lead to reduced customer satisfaction scores, lower brand loyalty, and customers moving to competitors.
Yes, high latency often contributes to or worsens packet loss and jitter. Packet loss can result in missing audio segments, while jitter causes uneven and choppy playback during calls.
Cyara’s latency test replicates the customer call flow to quantify the latency your customers actually experience. Repeated testing allows you to detect variations over time and generate data to support more informed routing decisions.
Testing from the customer’s perspective confirms you are monitoring the actual experience callers have when contacting your organization. This approach helps identify issues within the network or equipment that internal monitoring alone might miss.
Read more about: Audio quality, Automated testing, Call centers, Contact center, Latency
