If you are contemplating a VoIP telephone system for your company or agency, then you are probably already aware of the advantages that VoIP telephone system can provide you.
A VoIP PBX saves you time, has flexible modern functionality, allows your employees to work remotely, is extremely user-friendly, and can be scaled easily as your company expands or downsizes. However, to get the most out of a VoIP system, you need to have great internet speed and bandwidth. The relationship between the quality of your internet and VoIP quality is undeniable.
The Relationship between VoIP and the Internet
To experience all the benefits that a modern VoIP system has to offer, you need to have a stable internet connection with good speed and bandwidth. Instead of using conventional phone lines, VoIP utilizes the internet to send and receive calls.
Therefore, the speed of your internet is one of the primary factors that will decide how pleasant (or unpleasant) your VoIP experience will be. In the article below we talk about ways you can optimize your network for VoIP. Some of the factors that affect the quality of your VoIP calls include the following.
1. Latency
Latency (delays) may happen when the natural audio is compressed or when the compacted audio travels through the original network of providers. The delay is due to the time taken by a data packet to reach one point from another. Latency does not affect the audio output but will disregard the delivery-time. The level of latency will impact the performance of communication interfaces and/or devices. In the most severe instances of latency, the participants of the call will “talk over” with the audio reaching the computer of the receiver at sporadic and uncomfortable intervals.
2. Jitter
This is a disruption in the flow of the voice traffic. Say what? Basically, it means that the time it takes a given voice packet to reach the other end is drastically different, longer and shorter, then the previous voice packet. And different from the next as well. What causes it? It is the result of electromagnetic interference (EMI) and intermodulation distortion (IMD) from signals on the path between the sources and destination. Jitter can trigger loss of data, audio clicking, and other audio anomalies.
The providers of a VoIP service may supply built-in jitter buffers, but often latency far exceeds the capacity of the buffer. When this occurs, you will hear distorted or “skipped” sounds, or silent lapses.
3. One-Way Audio
One of the most common problems to affect VoIP systems, one-way audio is generally caused by firmware or configuration issues. Firewalls with old and outdated firmware, poorly made VoIP phones, and routers can trigger one-way audio issues. Also, if the different devices making up the VoIP telephone system are not configured properly, then this can lead to one-way audio—especially if the call path involves Network Address Translation (NAT). At the end of it all, one-way audio means something is blocking audio from being sent or received.
4. Mean Opinion Score (MOS)
MOS rates a call’s quality between 1 (unacceptable) and 5 (excellent). According to VoIP-info—an information resource on all things VoIP—the average rating or MOS of VoIP calls is between the 3.5 and 4.2 range. Anything lower would probably constitute a poor call experience. The following is the complete chart that lists different MOS ranges and shows how they are categorized:
Businesses must track the MOS of their calls to achieve optimum call quality and to minimize the number of disruptions or calls being dropped. Here, it is important for you to know that the quality of your voice calls and thus its MOS score can be affected by a range of factors including packet loss, bandwidth, latency, codec, jitter, and hardware. Codecs are software/hardware processes that encode or decode sound to and from digital packets. You also need to be aware of the following things affecting MOS scores:
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