Jitter can be a major problem for VoIP, Spearline helps provide the solution.
Have you ever had a jumbled-up phone conversation, where you and your friend are saying two completely different things, crossing over and having the most incoherent discussion? Generally, bad audio quality can be a little bit frustrating, can’t it?
This is a widespread problem when it comes to calls online. Conversation elements mix oddly. Indeed, they come through in parts like a bad game of scrabble. All of this can easily frustrate a customer, impacting the quality of the interaction.
This issue is a common cause of poor VoIP conversations. Significantly, this can 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 allow more flexibility than ever before.
Providing feature-filled telephony service at a fraction of the standard company telephony price, VoIP can be highly effective. However, part of what makes VoIP so effective can also be the cause of bad service quality. VoIP relies on an internet connection. Consequently, 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, eliminating latency and its impact is not possible. 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 occurs when the information arrives out of order at the recipient’s line. Specifically, information packets arrive in a different order from when they were sent.
There are three types of Jitter:
- Random Jitter: Usually resulting from clock timing problems or random electronic timing noise.
- Deterministic Jitter: Predictable 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?
There are numerous causes of Jitter, some of which are:
- Wireless Networks – A wireless network allows mobility and frees the need for wires running through the office. However, 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. As a minimum, there will be a modem and a router. Most often there will also be 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.
Jitter is likely the most common and frustrating issue impacting VoIP quality. In fact, it can cause many users to abandon a call. According to Cisco, 30ms is the acceptable jitter limit for a user. 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. The use of jitter buffers can minimize jitter impact.
A buffer is a caching system for receiving data packets. Software creates packets of data that contain elements of the voice call. These are sent across the network. They are temporarily stored in buffers along the way to their destination. While latency measures delays to VoIP calls, jitter measures latency variation. However, too large a buffer can directly increase the latency, so you need to take care in choosing buffer size.
Spearline provides insight into these jitter buffers through the use of proactive monitoring and number testing.
The importance of Network Monitoring
Network monitoring is part of the toolkit that ensures the hardware and data services of your internal network operate efficiently and effectively. Undeniably, network monitoring is critical. However, traditional tools only look to the private network. Spearline’s functionality verifies the call routing to the customer from your own private network or vice versa.
As shown above, network monitoring does measure a nice selection of audio quality factors, including Jitter. However, number testing provides a wider variety and an end-to-end perspective. Spearline’s number testing will provide accurate latency measurement. Measurement of real audio, across the actual path of the audio, including any transit 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. We 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 and toll-free numbers for audio quality and connectivity, globally. We support businesses in successfully connecting with their customers. Interested in benefiting from our platform? Please get in touch with us.