Experiencing poor VoIP call quality is not uncommon for businesses. Whether it’s caused by a jitter or a latency issue or something to do with your below-part internet connection, sub-standard voice call quality can be a bummer. However, most of the time, it’s not difficult to find and fix the nuisances. In the following article, we will talk about some of the most common reasons why VoIP call quality gets sacrificed and how the effects can be minimized.
In the world of packet-switching or connectionless networks, jitter is a usual culprit when it comes to bad or distorted call quality. For the large chunk of information (voice data) to travel over the network, it has to be divided into smaller packets. Each data packet can take its own path to travel from the sender to the receiver. If the sequence of the packets get altered during transfer (packets reach the receiver in a different order compared to the one they were dispatched in), it leads to a poor reconstruction of the voice data at the receiving end. This eventually leads to distorted or scrambled audio.
Jitter can be defined as the change in the delay after which the packets are usually received. Common causes for the delay can be inappropriate queuing, configuration problems and network congestion etc.
But how do you remove jitters?
Jitter buffers can be used to minimize the variations in the delay of received packets (i.e. jitter). They store the incoming packets temporarily to ensure that the order of the packets is preserved. The packets that arrive a little too late are discarded.
2. Poor Internet Connection:
VoIP is reliant on your internet connection to complete the to-and-fro transfer of voice packets to deliver its functionalities. The speed, availability and efficiency of your internet connection play a huge role in ensuring the quality of your VoIP calls. The bandwidth of your internet connection has to be substantial enough to support good voice quality.
A bad internet connection can also become the root cause of latency and jitter related issues.
VoIP delay or latency can be referred to as the time during which the speech exits the mouth of the sender and reaches the ear of the listener. There are 3 types of delays:
#1 Propagation delay:
It takes time for packets of voice data to reach the receiver. Ideally, this delay is negligible because fiber networks transmit data at high speeds but when it gets complemented by handling delays (explained below), it can become noticeable.
#2 Queuing delay:
Queuing delay can be defined as the time wasted after the packets get assigned to a transmission queue and before they actually start getting transmitted. During this period, the packets are kept in a buffer while other packets in the queue are sent on their ways.
#3 Handling delay:
Handling delay is caused by the devices present in the network that are responsible for forwarding the data to its desired location.
How to remove said latencies?
Delays will always be somewhat present in a network that’s always saturated with data, but they can always be minimized. An easy way of doing that is to prioritize the VoP traffic. Bandwidth reservation, MPLS or multi-protocol label switching, Class of service, type of service and policy dependent network management etc. are all efficient techniques to give VoIP traffic the top priority in a network. Prioritizing VoIP traffic can result in a significant decrease in jitters and latencies.
4. Bad Router:
A router with inadequate routing or buffering or scheduling capabilities can also result in voice quality being bad. The solution for this is the installation of special VoIP routers that can cater to the needs of a VoIP network.
This is a common issue as businesses often try to use the same network connection for both voice and data transfer. This is okay as long as you are prioritizing your VoIP traffic but if you haven’t got a router that can achieve this, then you can end up in trouble. For instance, if a user A on the network is currently making a call and another user B starts downloading a huge file on the same network, the quality of the call can get affected if VoIP prioritization isn’t in place. However, if a specialized router is being used, it prioritizes user A’s call over the download initiated by user B and doesn’t let the call quality get affected.
5. Internet Connection Saturation:
Bandwidth saturation can occur when a network tries to transmit data that exceeds its capacity/bandwidth. When it takes place, the available bandwidth for every initiated call can drop significantly, leading to diminished call quality.
However scary it may sound, bandwidth saturation can easily be controlled. In order to do so, you need to be aware of the amount of people that are connected to your network and the type of applications they are running on their computers along with their expected usages. Depending on these factors, a network architect should ensure that the network bandwidth is capable enough to entertain all the users and all the requests without letting the network get too saturated.
6. Unreliable Internet Service Providers:
Another reason for poor voice call quality can be an undependable internet service provider. While choosing a vendor, make sure that you get a trial network set up where you can run connectivity tests to ensure the reliability and suitability of the connection for VoIP related traffic. For excellent call quality, the value of jitter should be 1 milliseconds (or maximum 2ms) and there should be no packet loss. If your jitter values go up to 7ms and your packet loss is around 1 percent, you will still get acceptable quality in your voice calls.
The Final Word:
The voice call quality is arguably the most important facet of a VoIP system. In order to ensure high levels of quality, the above mentioned usual suspects need to be removed from a system. An internet connection with substantial bandwidth along with sophisticated VoIP routers can go a long way in ensuring that there is low jitter and latency in transmitted traffic, eventually leading to excellent voice call quality.