Have you ever had a jumbled up conversation, where you and your friend are saying two completely different things, crossing over and having the most incoherent discussion? A little bit frustrating, isn’t it?
This is a widespread problem when it comes to calls online. Pieces of conversation getting mixed up, coming through in parts like a bad game of scrabble, can easily frustrate a customer, impacting the quality of the interaction.
This issue is a common cause of poor VoIP conversations. This can then adversely affect business growth in terms of the cost of the calls, customer satisfaction, and staff churn.
What is VoIP?
Voice over Internet Protocol (VoIP) allows individuals to make calls over the internet. A great solution for most businesses, VoIP telephone systems allows more flexibility than ever before.
Providing feature-filled telephony service at a fraction of the standard company telephony price, VoIP can be highly effective. Part of what makes VoIP so effective, however, can also be the cause of bad service quality. Since VoIP relies on an internet connection, the service is prone to delays and hiccups if your internet connection is not up to the assignment.
Because VoIP services are still dependent on internet connections, interruptions caused by latency cannot be eliminated entirely. The dreaded JITTER is the largest of which.
In telecommunications, jitter further refers to the variation in the latency/time delay of packets carrying voice or video data over a communications channel.
To put it in layman’s terms, jitter is when the information arrives out of order at the recipient’s line — they won’t be obtained in the same order in which they are sent.
There are three types of Jitter:
- Random Jitter: Usually resulting from clock timing problems or random electronic timing noise.
- Deterministic Jitter: It could be predicted or determined. Reproducible and limited and may be periodic.
- 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.
What causes Jitter?
The cause of Jitter can be pinpointed to a number of reasons, some of which are:
- Wireless Networks – While a wireless network allows mobility and frees the need for wires running through the office, there is a chance that you will experience a degraded network link. WiFi is not necessarily secure or stable enough for our mobile devices to depend on for our phone calls.
- Bad Hardware – Internet networks are usually made up of a few distinct parts of hardware, at least a modem and a router, and sometimes includes switches. Bad hardware, such as an obsolete modem, a damaged Ethernet cable, or a misconfigured router, can easily lead to call quality problems.
- Network Congestion – The most common cause, stemming from an overcrowded network. Having too many devices hooked up to the same system, all being used at the same time, will run out of bandwidth, slowing your connection to a crawl. Insufficient bandwidth to handle a VoIP call will lead to packets being dropped or delivered out of order.
Likely to be the most common and frustrating issue of VoIP quality, Jitter can cause many even to leave the standard telephony service. According to Cisco, the acceptable amount of Jitter time for a customer should be limited to 30ms. In reality, Jitter can be readily diagnosed, acknowledged, and also minimized.
The good news is it’s not all doom and gloom when it comes to this latency annoyance. Jitter can, in fact, be minimized through the use of Jitter buffers.
A buffer is a caching system for receiving data packets. A voice call is broken into packets of data and sent over a network. Packets can be stored by a buffer on the other end. Whereas latency measures delays to VoIP calls, jitter measures latency’s variance within a network. However, letting too large a buffer directly increases the latency of actions on the sender’s side being received by a user, so you need to be careful in the size of the buffer.
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.
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.
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Spearline is a technology company that proactively tests toll, toll-free and premium-rate numbers for audio quality and connectivity globally. We support business sectors, such as contact centers, conferencing services, and other applications, in successfully connecting with their customers. If you are interested in benefiting from our platform, please get in touch with us.