OPUS VoIP Codec

VoIP (Voice over IP) uses Codecs to transmit the voice packets between Sender (Caller) and Receiver (Callee).

A Codec handles several things whereby 1) The bandwidth usage, 2) compression rate and 3) voice quality are the most obvious.

1)  Bandwidth Usage:
A Codec is something like a Zip (compression) technology. This is important because if your bandwidth is small you want to Zip (compress) the VoIP packets more as opposed to when your bandwidth is high you need less -or almost- no compression.

2) Compression rate -Audio quality-
Most Codecs have a set compression rate. For example, G711 has a very low compression rate hence the audio quality is High Definition (HD) i.e. as if you are talking face-to-face while other codecs like GSM have a very high compression rate i.e. the audio quality is Low Definition, GSM quality.  

3) Voice quality -Smoothness-
Voice quality is influenced by the latency (connection speed) between the caller and callee and can be influenced by many factors of which, 1) the distance (hops) the VoIP packages have to travel to and from the Caller to Callee and, 2) the quality of the links between all hops. For example, if you are in the US there are likely more hops to and from Caller to Callee and/or when you are in poorly covered 3G area the quality of links between some hops might be lesser.

There are various Codecs which all handle each item differently and all have their pros and cons. For example, a highly compressible Codec uses less bandwidth but will reduce your audio quality as well as increase CPU usage, i.e. similar as if you would zip a file on your pc, the higher the compression rate the slower your pc becomes”.

However, at ORENcloud we use the OPUS Codec, which is a highly versatile audio codec which is also used by all major players in the communications space such as Google Meet, WhatsApp Voice etc.

The advantage of OPUS is that it can dynamically adjust to the circumstances, if your connection is good the compression will be low hence high definition audio quality while when connection quality decreased it will compress more and vice versa. For example if you are driving from a 5G area through a 3G back to a 4G area, while on the phone during your journey the codec will adapt to the most optimal performance.

To test your connection quality and calculate your optimum bandwidth refer to article Bandwidth Calculator and Speedtest



Article ID: 38
Last updated: 14 Feb, 2021
Revision: 6
FAQ -> OPUS VoIP Codec
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