Minimizing Jitter for VoIP calls

To most organizations, Voice Over Internet Protocol or VoIP mobile services are a perfect option. This approach allows for more versatility than ever. Like for other technologies, running efficiently depends on the correct infrastructure. While VoIP has many important aspects, one of the most widespread and common problems that it faces is jitter.

The simple fact that the service is provided over an internet link is part of what makes VoIP Telephony so efficient. Instead of depending on traditional copper lines and a Publicly Switched Telephone Network to make calls, VoIP transfers the voice over the internet as input. This makes for significant cost reductions and feature-packed services that don’t require expensive hardware equipment.

VoIP may be extremely successful in delivering feature-filled telephone service at a fraction of the regular business telephony cost. Nonetheless, part of what makes VoIP so successful may also be the source of the low quality of service. Because VoIP depends on an active internet connection, the service is vulnerable to delays and hiccups if the internet connectivity is not up to the task. 

VoIP transfers data packets from one computer to the receiver’s machine, enabling a chat back and forth. Packets are delivered in increments to maintain the dialogue going smoothly. However, since your voice data is divided and separated into individual packets, sometimes the information is not necessarily sent in the same order as to how it was spoken, or certain packets reach the recipient quicker than other packets, due to network interference or congestion. This is what is known as jitter.  


Understanding Jitter

Jitter refers to the variation in the latency/time delay of packets carrying voice or video data over a communications channel. Jitter happens when the RTP packet stream traverses the network (LAN, WAN, or Internet) because it has to share network capacity with other data. 

Jitter can cause many participants to leave the phone call and either attempt to re-establish a connection or move to another form of communication.


There are three types of Jitter: 

  • Random Jitter: Also referred to as “Gaussian jitter”, random Jitter is unpredictable electronic timing noise. Usually resulting from clock timing problems or random electronic timing noise.  
  • Deterministic Jitter: This type of jitter can be predicted or determined. Reproducible, limited and may be periodic. Deterministic jitter can either be correlated to the data stream (data-dependent jitter) or uncorrelated to the data stream (bounded uncorrelated jitter).  
  • Total Jitter: Calculated using a bit error ratio (BER), as well as a combination of random and deterministic jitter. The mathematical formula for calculating total jitter is Total jitter = Deterministic jitter+2*BER*Random jitter.


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Common causes of Jitter:

  • Wireless Networks – Although a wireless network provides connectivity and frees the need for cables flowing across the workplace, you can face a weak network connection. WiFi is not inherently safe or reliable enough for our cell phone calls to count on.   
  • Poor and Misbehaving Hardware – Internet networks typically consist of a number of distinct hardware pieces, at least a modem and a router, which often have switches. Bad hardware, including an outdated modem, a broken Ethernet cable, or a malfunctioning router, will quickly contribute to issues with call quality.


Other notable causes

  • Hardware capacity on source/destination
  • Network capacity 
  • Bad routing through network/internet (traffic routed in a suboptimal fashion introducing delays, etc)
  • Network Congestion


While Jitter can be minimized and reduced, through the use of Jitter buffers, this is not a ‘magic solution’. Depending on the size of the jitter buffer, out of sequence packets can be rearranged before being delivered. However, this does introduce some delay in addition to the original latency on the network. Spearline provides insight into these jitter buffers through the use of proactive monitoring and number testing.   

The importance of Network Monitoring

Network monitoring is a critical part of the toolkit to ensure that the hardware and data services of your internal network operate efficiently and effectively. However, traditional tools only look to the private network, but Spearline’s functionality verifies the call routing to the customer from your own private network or vice versa.  

End-to-end Latency

As shown above, yes, network monitoring does measure a nice selection of audio quality factors including Jitter; Number Testing provides a wider variety and end-to-end perspective. Spearline’s number testing will provide latency measurement of real audio across the whole path of the audio, including through the PSTN/mobile telephone network. Our test will be helpful in providing insight as to how to improve quality, reduce jitter and packet loss, and more. 

Network monitoring will focus on the internal network. Spearline helps verify paths from the customer’s vantage point into the private LAN/WAN network space. 

Providing an accurate end-to-end measure of audio quality using PESQ, Spearline proactively monitors by dialing your contact numbers from in-country.


Read our whitepaper for a better in-depth understanding of network monitoring and number testing.

At Spearline, we have expertise in measuring and monitoring customer experience and have the tools which you can use to ensure your network is supporting high-quality customer conversations that grow your business.


New to Spearline?

If you are new to Spearline and would like to find out more about how you can benefit from our platform, we would love to speak with you. Please send us a brief message, and we will be in contact with you shortly.

About Us

Spearline is a technology company that proactively tests toll and toll-free numbers for connectivity and audio quality globally. It enables organizations to provide uninterrupted services to customers around the world. For further information, or if you have any further questions please get in touch with us.

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