Tag Archives: voip quality

Mean Opinion Score and V1 VoIP Quality

The Mean Opinion Score (MOS) is a test that has been used for decades in telephony networks to obtain the human user’s experience of the quality of a phone call. This measurement is the result of underlying network attributes that act upon data flow and is useful in predicting call quality and is a good VoIP test tool in determining issues that can affect your VoIP quality and your conversations.

Testing the quality of VoIP has become easier which has furthered the advancement in the quality of services has helped increase the number of VoIP subscribers, but occasionally issues affecting voice quality do arise and being able to test your VoIP and identify these instances can be helpful in addressing them. The MOS is assigned by a group of listeners using a scale of 1 to 5 with

5: Perfect. Like face-to-face conversation or radio reception.
4: Fair. Imperfections can be perceived, but sound still clear. This is (supposedly) the range for cell phones.
3: Not great.
2: Very annoying. Nearly impossible to communicate.
1: Impossible to communicate

By its very name “Opinion”, the MOS, Mean Opinion Score was a subjective measurement used to test the listener’s perception of the voice quality, and clarity of the communication. The test was performed in a ‘quiet room’ meeting specific size and noise requirements, in which listeners would receive and score calls on the quality as they perceived it. VoIP MOS score measurements, on the other hand, is usually more objective, the score providing a measure of the network over which it is carried.

Having a metric to measure changes or degradation in the quality of the voice/VoIP connection after testing can help identify problems. VoIP calls often are in the 3.5 to 4.2 MOS range. MOS is a relative scale and is built upon many factors which can affect voice quality. Real voice signals are used to test clarity, delay, packet loss, jitter and determine a probable MOS score. Yes, this is an estimate to the human based MOS score, but it is more practical and scalable. MOS takes into account factors including listening quality, transmission quality, and conversational quality.

To learn more about what your MOS score is and how you can improve it with V1 VoIP, contact us today.

By checking this box, I agree to V1 VoIP's Terms and Conditions


V1 VoIP Explains Jitter and Packet Loss

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

By checking this box, I agree to V1 VoIP's Terms and Conditions