Tag Archives: voip call quality

V1 VoIP Explains How to Improve Your Call Quality

V1 VoIP explains how resellers improve businesses call quality QoS

V1 VoIP explains to our resellers that there are three main components pertinent to the success of VoIP technology in business: cost efficiency, functionality, and reliability. Our resellers’ customers love that the first two characteristics come standard with our VoIP services, but many desire a call quality guarantee. To improve VoIP call quality, or simply to ensure its consistency, our resellers look at the following to determine if there’s room for quality adjustment:

Network Infrastructure
The infrastructure of a business’ network is a very important factor to consider. Resellers will ask much bandwidth is available, how many users are active online and who is the internet provider. Since it is up to external facilities to transport voice data packets, businesses need to look outside of their immediate environment if they are experiencing poor VoIP call quality. To keep costs down, investigate your current data plan, so that you do not exceed your internet limitations trying to optimize your system.

V1 VoIP resellers recommend using top-grade lossless compression software, otherwise you will significantly deteriorate your VoIP system’s call quality. It is also important to keep your computer running properly, meaning that regular defragmentation, memory clean-up, and system backup are necessary to ensure VoIP call quality consistency.

Location and Wiring
The use of a network monitoring device is a must; it enables tracking for each voice data packet, allowing for easy troubleshoot if a call goes awry. Similarly — and this is not something many people think to check — look into installing high-quality network cables to reduce static. Not only will this stop severe weather conditions from affecting your VoIP call quality, but it will improve the longevity of the network overall. To reduce in-house interference, always place your VoIP ATA away from the router; the placement of equipment can also affect VoIP call quality.

Is your call quality as clear as should be? As clear as it could be? Contact V1 VoIP today to talk to us about call quality issues, how to overcome them, and be in touch with a reseller who can provide you the best call quality in VoIP service and solutions for your business.

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V1 VoIP Explains Jitter and Packet Loss

v1 voip explains jitter and packet loss as it pertains to call quality

When it comes to call quality, V1 VoIP wants our resellers and their customers to hear nothing but clear, crystal quality. To make sure that you understand what goes into proper QoS (read about how V1 VoIP explains Quality of Service), V1 VoIP wants to explain some of the most common quality issues. The two most frequent issues VoIP users come across are jitter and packet loss. But what are they? Let us explain!

What is jitter?
From a customer’s point of view, a jitter means unstable voice flow in a telephone conversation. Have you ever spoken on the phone and heard the echo of everything you’ve said after you’ve said it? That’s a jitter. In some cases, their order might be reversed, too, causing confusion and misunderstanding on both ends. Looking at jitter from the technical side, jitter occurs when voice data packets do not arrive in a steady flow, required by codecs for sustainable playback. Usually, packets are sent from a caller at the same time intervals, imitating the landline connection, but they do not always arrive in the same order or following the same interval pattern.

So what is packet loss?
When jitter rates are especially high they will lead to packet loss. Packet loss is just that: when packets are not delivered at all and thus parts of the conversation are missing. Packet loss occurs either randomly and only by single packets known as “gaps” or in large numbers at once called “bursts”.

What causes jitter?
There are three main causes of connection jitter: First is the wrong application of queuing, as inappropriate storage of voice packets and the wrong order of their transmission can lead to delays. The second is where faulty configuration of a router or a PVC might easily impede connection quality and cause a jitter. The third is network congestion, which might cause irregular spacing between packets.

How to prevent jitter and packet loss
In order to avoid any perceivable interruptions in a conversation, the jitter should be 20 milliseconds or less. As it increases, the connection quality drops. To reduce jitter and ensure higher quality of connection, networks use jitter buffers – devices that collect packets from the caller and send them to the receiving codec in the right sequence and at even time intervals. In case of packet loss, jitter buffers duplicate missing data or adds comfort white noise. Besides using a jitter buffer, companies can examine their networks in order to see what causes the jitter and then either correct the instances of wrong configuration or allocate more bandwidth or use Priority or Law Latency Queuing.

V1 VoIP is dedicated to providing our resellers with information about VoIP termination and origination services, appropriate solutions for small businesses, and useful security tips. Understanding jitter and packet loss is vital to providing a great customer experience which will allow them to get the most of your VoIP technology. Contact V1 VoIP today for answers to your questions about troubleshooting and maintenance assistance.

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Are VoIP Calls the Same as Skype Ones?

Can I keep my phone number is the first question without fail, that V1 VoIP reseller customers always ask. But right after that, comes questions about call quality and whether or not it will stay the same once the switch to VoIP is made. If you’ve ever made a Skype call, you can understand why the question of call quality is asked. VoIP and Skype may use similar technologies to make phone calls over the Internet, but the respective voice quality of each solution differs significantly from the other.

Thankfully, with V1 VoIP providing services and solutions to resellers, you do not have to worry. V1 VoIP is designed to far surpass landline telephones and real-time communications apps in terms of voice quality.

Skype requires the same amount of bandwidth as a VoIP call, but the voice quality still differs. Skype implements the respected Opus audio codec, but the Skype network was not built to be undersubscribed bandwidth wise. Even though the “recommended” client side bandwidth for a Skype call is the same as a VoIP call, the call quality will suffer because of Skype’s own bandwidth limitations.

When on Skype and the call quality deteriorates, you are not dealing with a flawed design, but a limited network infrastructure that cannot provide dedicated service to millions of anonymous individuals. This is one of the key issues that V1 VoIP attempts to rectify. The average VoIP call takes the same amount of bandwidth as a Skype call does (100 kpbs) and yet still sounds of higher quality. V1 VoIP creates separate real-time networks and partner with Tier-1 Internet companies to ensure the highest bitrate possible.

In general, Skype and business VoIP are using comparable audio codecs, both of which are capable of wideband voice quality. But Skype’s network holds its voice quality back. And this is not just a voice quality issue, because Skype is more likely than business VoIP to outright drop calls in the case of severe packet loss, whereas business VoIP companies often make eradication of dropped calls a supreme priority.

If you’re fearful that switching to VoIP services will diminish the quality of your calls, fear not! The only thing your co-workers will notice is a crisper and clearer sound.

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