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VoIP Codecs - Effects on Call Quality

Combination of Compression and Decompression: Codec

For converting an analog voice signal to a digitally encoded version, codecs are using. VoIP codecs differ in sound quality, the necessary bandwidth, computational requirements, and more!

As we all know, any service such as software, phone, the gateway supports many different codecs and chooses which codec they are going to use while talking to each other.

How It Effects?

A VoIP codec is a system that defines VoIP phone calls for voice quality, bandwidth, and compression. VoIP codecs use either optimization algorithms or open-source ones.

Unfortunately, the weak VoIP codec can harm your budget. Therefore, it's crucial to know how much bandwidth you have and how much you'll need before you prefer an audio codec; you can stop using a codec that has more compression ratio than you need.

The more audio data is compressed by your VoIP codec, the more audio quality you'll lose. More audio quality would be reduced by a codec that reduces audio data to one-fourteenth of the original size than a codec that reduces the data to one-eighth of the original size.

Generally, VoIP codecs use lossy compression. Lossy compression removes some audio data for compressing the data. A lossy compression codec can minimize audio data to one-eighth or one-tenth of the original size by discarding a little bit of audio data. So, VoIP codecs use lossy compression. The crucial point is that during compression, your VoIP codec does a great job of selecting the appropriate one in which audio data is discarded and does not discard too much audio data.

The Most Common Types of Codecs

The crucial point is to be able to use as much bandwidth as needed effectively. Therefore, you should prefer to calculate how much you need based on the VoIP codec you are using.

There are quite a few codec options available; therefore, you should be careful in your choices. We've listed a couple of specific codecs to suggest!


Although you get higher quality sound, the necessity for bandwidth is relatively high! Also, instead of the G.729, this codec does not properly allow multiple phone calls.

Initially, G.711 was introduced in the 1970s by Bell Systems; it was officially adopted in 1988 by the International Telecommunication Union (ITU). Today, Voice over Internet Protocol (VoIP), also known as Internet telephony, is widely used for G.711.

Via logarithmic compression, this codec can compress 16-bit samples into 8 bits. That's why the compression ratio is 1:2. Also, 128 kbit/s (64 kbit/s for a single path) is the bitrate for both directions.


In 1988, ITU licensed this codec, the patent has now expired; that's why it is free to use for all! This codec helps enhance voice efficiency. Sub-band adaptive differential pulse code modulation with bit rates of 64 Kbps is used by the whole coding system and is referred to as 64 Kbps audio coding.

G.722 is used mostly in VoIP, such as on local area networks, where network bandwidth is readily accessible and improves speech efficiency instead of G.711. Broadcasters often use G.722 for sending commentary-grade audio over single optical 64 Kbps interconnected networks.


The most effective side of the G.729 codec is the better call quality with low bandwidth! This codec requires only 12.8Kbps per line with the maximum compression rate. Generally, 16Kbps or 23.6Kbps per line is required for this codec.

In frames, the codec encodes the audio. Every frame is ten milliseconds long and includes 80 samples of audio. The single-direction bitrate of this non-HD codec is 8kbit/s. Thanks to G.729, people are able to make more calls from their network at once with a high compression rate! This codec is a very good preference if you need to connect high volumes of VoIP lines!

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