There are several factors that are important to get good and stable VoIP connections and therefore calls:
1) A stable connection.
If you often have (little)hiccups on your line, you will certainly notice this with VoIP calls, calls may even break off when that happens.
2) Enough speed.
You can check your internet connection speed here: Check my current Internet speed
This does depend on the codec (audio <> digital <> audio translation) you use though, the better the quality you want and use, the more data it needs.
For 1 VoIP Call with the normal, standard codecs installed you would need a (playing it safe) speed of 100k In and Out.
Every simultaneous call is extra so, just multiply and see what number you need.
3) A good enough quality Internet line.
In this case, defined by jitter, packet loss and latency in particular.
These are technical terms that describe certain values concerning the quality of your internet connection.
Ideally, all these should be as low as possible.
Jitter:
When certain packets of information are dropped, or sent out of order, leading to a jumbled conversation.
Jitter should be 30 milliseconds or less for a VoIP connection and the lower the number, the better.
Packet loss:
Packet loss is when “packets” or pieces of data travelling across a network do not reach their destination.
The result of these packets failing to reach the opposite end of the network leads to missing data.
Ideally, packet loss should be 0% (or at least under 1%), however, it's possible for you to experience acceptable VoIP quality with packet loss as high as 5% but it also depends on which codec you use.
Latency:
This refers to a small delay as audio data moves from one phone system to another.
During a phone call, latency may cause the person on the other end to hear what you say just a few milliseconds later.
It may also cause “glitches” in audio data.
ITU-T G.114 recommends a maximum of a 150 ms one-way latency for a good quality VoIP connection.
Last updated: 01/01/2024
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