VoIP Jitter and Latency Explained: Causes, Effects, and How to Fix Poor Call Quality

When making a call through VoIP technologies, your voice is converted into packets of information and transferred through the internet. Then, it gets decoded and reassembled on the spot. But, in case something goes wrong, there are two issues you have to know about: jitter and latency.

However, before we discuss what these problems are, it's important first to learn more about the nature of VoIP jitter and latency and how they impact voice communications.

Jitter refers to the difference between the times the packets take to reach their destinations. Each packet travels a different path and requires a certain period of time, which results in packets arriving in an unsteady manner – either too early, too late, or in the wrong order.

VoIP phone systems use packets in a stable pattern to recreate audio. Any disruption in the pattern leads to audio interruptions that may sound robotic.

For example, imagine a speaker saying:

"Thank you for contacting our support team."

If packet delivery becomes inconsistent, the listener may hear:

"Thank... you for... contacting... our support... team."

Latency refers to the time taken by a voice packet to get from the sender's device to the receiver's device. Latency is normally measured in milliseconds and involves encoding time, transmission time, and processing time. In contrast to jitter, latency does not affect the sound quality; the sound is generally clear. However, what latency affects is the normal flow of conversation. With high latency, there will be an apparent delay, causing interruptions in conversations or unnecessary repetitions.

For example:

Agent: "Could you please confirm your account number?"

(A brief delay occurs before the customer hears the question.)

Customer: "Hello? Are you still there?"

By the time the customer responds, the agent assumes the line is silent and resumes speaking, causing both voices to overlap.

Jitter = inconsistent delivery of voice packets.

Latency = how long they take to arrive.

It is important for organizations to know about jitter and latency in VoIP since both terms will directly influence customer calls, staff communications, and the effectiveness of the communication process. By knowing how these two differ, one will be able to identify the exact problem behind the poor quality of the call and make the necessary changes to the network.

To understand why jitter and latency happen, it helps to understand the basic path a VoIP call takes:

1. Your voice is recorded and encoded into digital audio using a codec.

2. The digital audio is then divided into data packets and marked with sequence and timing tags.

3. Packets are transmitted from your local network through your internet connection, the public internet, or a private carrier network all the way to the destination of the call.

4. On the destination side, the packets are stored and ordered in a jitter buffer to even out any timing problems.

5. The packets are then put together to produce audio.

The important thing here is that they should stay within acceptable parameters so that communication becomes smooth and natural. Exceeding these parameters will increase the probability of getting choppiness in voice, delays, and interrupted communications.

Tip: For commercial VoIP systems, ensure that latency is under 150 ms, jitter is under 30 ms, and packet loss is under 1% in order to preserve high-quality voice communication.

• Sound quality that is mechanical, jerky, or metallic

• Dropping of words or syllables

• Noticeable delay between when someone speaks and when it's heard

• Both parties frequently talk over each other

• Calls that randomly drop or disconnect

• Echo or overlapping audio during longer calls

• Agents repeatedly asking customers to repeat themselves

Jitter and latency are not only network metrics; they are metrics that show up where they should be: average handle time, first-call resolution, customer satisfaction scores, and even money.

• Higher average handle time: Repetitive agents and customers lead to high average handle time.

• Lower customer satisfaction: Inefficient calls mean lower customer satisfaction regardless of the outcome of the calls.

• Misunderstanding: With dropped words or silence between the speakers, there is an increased risk of misunderstanding during conversations on order, payment, or other issues.

• Burnout of the agent: Understanding the caller requires more concentration and results in faster burnout.

• Lost sales: During sales calls, poor call quality is damaging to the establishment of trust.

It is essential to determine whether jitter first, latency, or other network problems affect your phone calls before attempting any fixes. Testing regularly is a great way to spot such problems before they occur and provide consistent VoIP functionality.

• Ensure prioritization of voice calls through QoS: Set up routers to ensure the packets carrying the voice calls have priority over other forms of traffic.

• Employ wired internet: This is more reliable for voice traffic compared to Wi-Fi.

• Upgrade networking equipment: Old routers and switches that cannot cope with the volume of traffic should be updated.

• Configure jitter buffer sizes: Most VoIP setups enable you to configure the buffer sizes such that you accommodate timing discrepancies without causing delays.

• Separate traffic on your network: Avoid mixing voice calls traffic with traffic associated with downloading and video streaming.

• Consider a cloud VoIP service with distributed network nodes: This allows you to minimize the distance that packets have to travel.

• Pick a provider with regional data centers: Using servers that are located close to you cuts down on distance and latency.

• Get more bandwidth: Make sure you have adequate bandwidth to send your voice traffic without any queuing.

• Cut down the number of hops: Simplify your network path and try not to use unnecessary routing via VPN or other proxy networks.

• Pick effective codecs: Opus is one of the codecs created for providing good sound with low processing load.

• Implement QoS prioritization: Just like with jitter, it helps minimize the delay due to queuing.

• Upgrade and monitor internet plans: Business-grade plans perform better than consumer-grade ones when it comes to latency.

One of the most effective ways to improve VoIP calls is through the use of Quality of Service. With QoS, network administrators will be able to guarantee priority for voice traffic over other less important services like file downloads, video streaming, or software updating.

However, in situations where there is a great deal of network traffic, this may result in additional jitter, latency, and packet loss.

The advantages for businesses using QoS include:

• Reducing jitter and latency in times of high network traffic.

• Making calls clearer and reducing disruptions.

• Reduce jitter and latency during busy network periods.

• Improve call clarity and reduce interruptions.

• Prevent dropped or delayed voice packets.

• Deliver a more consistent communication experience.

However, QoS will not make your Internet connection faster, but it will allow businesses to take better advantage of the available bandwidth when making VoIP calls.

The network performance metrics jitter, latency, and packet loss have a mutual relationship. Despite being different factors that affect VoIP calls in different ways, they usually occur together when there is network congestion or misconfiguration.

How They Work Together

Imagine a VoIP call as a transfer of pages of a document via the postal service:

• Latency refers to the time taken by the package to arrive.

• Jitter happens where the pages do not arrive simultaneously.

• Packet loss occurs where the pages fail to get delivered.

Where either of these factors alone exists, there will be problems with call quality. If the three happen at the same time, there will be poor-quality calls.

• Implement network monitoring and call quality monitoring as a regular process, not as a reaction to user complaints.

• Create QoS policies in the initial stage of network configuration.

• Conduct periodic evaluations of the bandwidth requirements of your team based on growth in the number of members, increased call traffic, or addition of video calling.

• Use a wired connection for all workstations that have to handle large amounts of call traffic.

• Arrange periodic audits of your network, particularly before a major hiring drive or moving to a new office.

• Select a VoIP solution that has analytics as part of its offering to make sure you identify problems before they lead to user complaints.

Unlike cloud-based VoIP providers, the infrastructure of an on-premise VoIP system relies completely on the IT team of the company to monitor and manage bandwidth, route traffic, ensure redundancy, and implement QoS policies. For growing businesses, especially those managing distributed or hybrid teams, this generally means more consistent call quality with far less manual network management required on their end.

TeleCMI's cloud based VoIP platform is built to minimize jitter and latency at the infrastructure level, so businesses spend less time troubleshooting network issues and more time on actual conversations.

• Built-in Call Analytics: TeleCMI's Call Analytics and AI Call Score features surface call quality trends, helping teams spot recurring jitter or latency issues before they affect customer experience.

• Cloud infrastructure for reliable performance: The TeleCMI Cloud PBX and Cloud Contact Center solutions have been built with redundancy and optimized routing to minimize the number of hops the voice packets have to make.

• Flexible Business Phone System: Allows for both softphone and desk phone configurations, depending on which one will ensure the best stable wired connectivity.

• Voice AI and CPaaS with PIOPIY: Aimed at developers who require low-latency programmable voice infrastructure for their custom-built applications and voice AI agents.

• When businesses need to evaluate their current VoIP infrastructure, our specialists can assist in determining the appropriate configuration and network readiness to maintain consistent call quality.

VoIP Jitter and Latency are two of the most frequent causes of poor quality on business calls, which fortunately happen to be the easiest to solve. By recognizing the distinction between the two and establishing proper benchmark values to reach, one may improve call clarity and ensure that it does not interfere with customers' perception of the company and its employees' efficiency.

The good thing is that jitter and latency issues can usually be attributed to a limited number of reasons - congested network, poor QoS, unreliable connection, or a poor-quality infrastructure that is not suitable for handling voice traffic. Using proper monitoring techniques, configuring the network properly, and choosing a reliable cloud VoIP service provider, high-quality voice calls can be achieved with ease.