Voice over Internet Protocol technology has been around for almost 100 years. A VoIP’s job is to transfer audio from one point to another through the internet. What made us have spectacular quality internet phone calls today were VoIP codecs.
What are VoIP codecs? How do they play a part in call quality? What is the best one for your business?
Continue reading this article to learn all you need about this technology.
What are VoIP Codecs?
Simply put, VoIP codecs are computer programs that compress analog audio signals into digital signals. These signals are transmitted over the internet and decompressed back into their original analog form at the receiving end.
VoIP codecs determine the quality, compression, and bandwidth of internet phone calls depending on their type. The hardware and devices used have an effect on how well the codec works.
The word codec comes from the first letters of Compressing and Decompressing or Coding and Decoding.
When transmitting audio signals, compression is essential, and here lies the most prominent distinction between different VoIP codecs.
Transmitting audio signals requires high bandwidth. Therefore, VoIP codecs that need less bandwidth for signal compression will use less space and are more efficient.
VoIP and Audio Quality
Nowadays, a reliable VoIP system is one of the most important aspects of a business’s success as it controls call sound quality.
Below, you will find some terms related to audio quality and their explanations:
- Sample rate: audio samples recorded in one second are known as sample rate. Each audio sample indicates a waveform’s value over a specific period of time. Sample rate affects audio quality – a higher sample rate leads to better audio quality.
- Bitrate: It refers to the amount of data (bits) that is processed per second. In this case, data is transferred into audio. A high bitrate means improved audio quality.
- Bandwidth: This is the speed at which you send or receive data. Bandwidth depends on the bitrate. The transmission rate is the number of samples sent every second. High transmission rates allow you to send more samples per second.
We can deduce that low bitrates and sample rates indicate low-quality audio. Consequently, bandwidth is the main concern here, and that’s where VoIP codecs comes in; they aim to conserve bandwidth and produce the highest audio quality possible at the same time.
How VoIP Codecs Improve Call Quality
Speaking to people on the phone is different from having an in-person conversation because phones can’t capture the full range of human sound. Phone call quality is divided into two bands: narrowband and wideband. Frequencies ranging from 300 Hz to 3400 Hz are covered by narrowband, and wideband covers the ones between 50 Hz and 7000 Hz.
The notion of speaking in wideband, which is also referred to as HD Voice, enables you to hear a wider range of frequencies. This is the closest resemblance to a face-to-face conversation.
Improving call quality and conserving bandwidth are the main purposes of VoIP codecs.
VoIP codecs use something called lossy compression, which reduces data to 1 tenth of its original size. This process is possible by eliminating a small portion of audio data. However, VoIP codecs can efficiently select which parts of data to remove to preserve audio quality.
Types of VoIP Codecs
Throughout the years, many types of VoIP codecs have been introduced. They differ from one another in terms of audio quality, required bandwidth, compression, and availability.
Here are some of the most well-known VoIP codecs:
G.711
In 1972, the International Telecommunication Union (ITU) launched the G.711, a narrowband codec.
This codec compresses 16-bit samples into 8-bit ones using logarithmic compression. It has a bitrate of 64 kbit/s for a single path.
The advantages of the G.711 codec are that it doesn’t require any licensing fees and it provides high-quality audio. However, it does use a high level of bandwidth and doesn’t support multiple phone calls as well as other audio codecs.
G.722 HD
This wideband HD codec was approved by the ITU in 1988. Compared to the G.711 codec, the G.722 offers better audio quality and clarity.
It has a sample rate of 16 kHz (using 14 bits) and offers 64, 56, and 48 kbit/s bitrates. Similar to the G.711, this codec’s patent has expired, which means it’s free for use.
This codec’s audio was described as having “toll quality,” which is similar to public switched telephone network (PSTN) call quality.
G.729
The G.729 VoIP codec has rather low bandwidth requirements and good sound quality.
This codec also offers a higher compression rate than the G.711 and G.722. For that reason, you will be able to make more calls from your network.
In contrast to the above two codecs, the G.729 requires a licensing fee. Nevertheless, you can purchase hardware that uses this codec, and you won’t have to pay for licensing.
OPUS
PlayStation 4, WhatsApp, and Discord are some of the companies that use this VoIP codec.
OPUS supports sample rates ranging from 8 kHz to 48 kHz and bitrates between 6 kbit/s and 510 kbit/s. This fact makes it more adaptable to numerous applications.
In addition to being a royalty-free codec, OPUS provides you with low latency levels on your internet phone calls, which is very crucial to most businesses. This makes it very useful in VoIP systems.
VoIP Codecs FAQs
Below, we have answered some of the most frequently asked questions regarding VoIP codecs.
What’s the best codec for VoIP?
Since OPUS is royalty-free, has adjustable bitrates, and has low latency levels, among many other features, it may be considered the best VoIP codec.
What are the best VoIP codecs for businesses?
In order to determine the best option for your business, you would have to do a quick VoIP codec comparison.
You need to know your bandwidth abilities and identify your desired audio quality. You should also take licensing fees into consideration when choosing the right codec for you.
For example, the G.722 might be considered by many as the best VoIP codec given they possess the necessary bandwidth for it.
What is HD Voice?
HD Voice is another name for wideband audio. It is a high-definition audio technology that decreases background noise to deliver crystal-clear sounds.
What is MOS?
MOS stands for Mean Opinion Score. MOS is a scale used to grade the audio, voice, and video quality of a call.
It has a scale of 1 to 5 as follows:
1: Bad
2: Poor
3: Fair
4: Good
5: Excellent
Summary
There is no doubt about the essential role VoIP codecs play in a professional environment. They were made to improve the quality of phone calls, which in turn creates the opportunity for smooth communication.
There is also no doubt that each VoIP codec has advantages and disadvantages. While some codecs fixate on delivering the highest levels of audio quality, others work on low bandwidth requirements.
Choosing the ultimate VoIP codec for your business depends on your capabilities and your needs.
Reach out to ULTATEL, and we will assist you in learning more about VoIP systems, codecs, and cloud phone systems.
