Voice over Internet Protocol (VoIP) allows for crystal-clear phone calls and requires only an internet connection. VoIP codecs make it possible.
Continue reading to learn more about codecs and how to choose the right one for your company’s VoIP phone system.
VoIP codecs are technologies that control the quality, bandwidth, and compression of Voice over Internet Protocol phone calls. Open-source and proprietary algorithms are used in VoIP codecs.
The Codec is a combination of Compression and Decompression.
Codecs allow you to download movies in minutes instead of hours. Codecs can be used to capture images (JPEG), encryption (AES), streaming media (H.264), or music. Audio recording software MP3 codecs, for example, determine the quality of your video on YouTube or Netflix.
A VoIP codec converts analog voice signals to digital packets or a compressed digital form for transmission. Then it returns to an uncompressed audio signal. Since the conversation takes place over the internet, VoIP codecs determine call quality and latency.
There are some possibilities. VoIP issues because calls travel over the Internet, reliability is not an issue if your VoIP provider has multiple data centers.
Compatibility with other devices is key to the success of a codec.
VoIP codecs have one common purpose. They compress data and quickly move it. VoIP for business ensures that phone calls don’t consume a lot of bandwidth and that calls sound clear and crisp.
Voice-over IP devices transform audio signals into digital packets and send them via the Local Area Network. You might wish to enable this feature in order to prioritize these data packets. Service quality (QoS). This allows VoIP data to be prioritized over less important traffic. QoS helps VoIP codecs maintain superior call quality.
The only thing that is different between VoIP codecs and each other is how they compress audio. Because audio transmission requires large bandwidth (which is limited), compression is essential. Organizations will have trouble finding enough space for files and documents if compression is not used.
To ensure that VoIP compression is effective and uses less bandwidth, business owners need to think about VoIP codec protocols.
These measures will allow organizations to maximize their capacity planning and reduce expenses. They also ensure that the company is a safer investment over the long term.
Cloud VoIP phone systems remote teams will find it especially useful. Employees can stay connected with it by allowing them to video conference, call recording, and long-distance calls.
Audio quality basics
Today, millions of businesses rely on reliable VoIP systems. Fidelity is the foundation of great business communication. This is why audio quality over the telephone is so important.
Before we get into the details of audio processing, it is important to be familiar with the terms used in audio quality.
- Rate sample. It is also known as the sample frequency. It refers to how many audio samples are taken per second. Each sample will give you the total amplitude value of the signal waveform over a specified period. The audio quality is better if the sample rate is higher.
- Bitrate. The audio bitrate is the amount of data that can be converted into audio. A higher audio bitrate means that it captures more sound information per second. A higher bitrate is generally indicative of a greater bitrate. Better sound quality.
- BandwidthBandwidth is the speed at which you can send or receive data. The transmission rate refers to the number of samples sent every second.
Low bitrates will make your sound terrible, no matter how good it is. The same goes for sample rates.
bandwidth can be described as a bottleneck. VoIP codecs are designed to save bandwidth and maintain high sound quality.
HD Voice increases call quality.
You may have noticed a difference in the sound when you talk to someone on the phone rather than talking face-to-face. The phone can’t pick up all frequencies the human voice hits, so this difference is obvious.
Human speech has a frequency range between 80 and 14,000 Hz. The deeper the sound, the lower the frequency. Lower frequencies make up a punchy beat in pop songs. Vocalists are known for their 250- to 1500 Hz.
However, phone conversations can be very different.
Two bands are common in phone audio: narrowband or wideband. Audio frequencies between 300 Hz and 3400 Hz are considered narrowband. These are for audio frequencies between 50 Hz and 7000 Hz.
Wideband audio is also called “Wideband Audio” or “HD Voice. You’ll be able to hear a wider range of pitches and most closely mimic an in-person conversation.
Wideband increases the audio spectrum and makes it sound better.
Another example is here. For optimal acoustics, automotive manufacturers design a vehicle’s exhaust system. Because high frequencies cancel out lower frequencies, luxury sports cars can be stealthy.
VoIP codecs use a similar approach in order to reduce background noise and make phone conversations more natural.
There are three types of VoIP codecs.
There are many codec options, so choosing the right one can be difficult. Below are a few codecs that you might want to take into consideration.
G.711
G.711 was introduced by the International Telecommunication Union (ITU) in 1972 for telephony. There are two versions of this codec: A-law and m-law. Japan and the United States use m-law. Europe uses A-law
Logarithmic compression allows this codec to compress 16-bit samples into eight bits. The compression ratio is now 1:2. This is quite a bit. The bitrate is 128 kbit/s in both directions (64 kbit/s if you only have one path).
Although you get superior sound quality, this codec has a high bandwidth requirement. This codec is not able to support multiple calls as well as codecs such as G.729.
G.711 can be used for any type of VoIP application as there are no licensing fees. It does not use digital compression, which is why it is considered the best VoIP codec for interface with the public switched telephone network (PSTN).
G.722 HD
The G.722 codec is high-definition, which means it’s wideband. This codec was approved by the ITU in 1988. The patent has now expired and it’s available for free.
This codec improves speech quality without any perceivable latency. HD voice can transmit twice as much data at 16 bits as G.711 and has a sample rate that is double that of G.711. The transmission speed remains constant at 64 kbit/s.
G.729
The G.729 codec is an excellent choice if you are looking for a codec with low bandwidth requirements and acceptable sound quality.
The codec encodes audio into frames. Each frame contains 80 audio samples and is 10 milliseconds in length. This non-HD codec has a bitrate of 8kbit/s for one direction. Because the compression is more powerful, you can make more calls to your network at once.
However, some VoIP providers might not be able to support the G.729 codec. Sometimes, music and other non-verbal audio may sound choppy.
How to organize your codecs
Cloud VoIP providers decide which codecs you have access to for your hardware.
VoIP providers send the data packets while IP phones must compress and decompress audio. When there is a call attempt, both the caller and the called phone negotiate the correct codec. Each caller and receiver phone have a prioritized list that they use to determine the correct codec.
Choose the codec that is most compatible with your phone system when choosing the best codec. Consider your team’s bandwidth and call volume.
If you prefer, better call quality. You should first place G.722 and then G.711. If you are concerned about bandwidth, then set G.729 higher than G.711.
G.711 is accepted by almost all VoIP providers and phones. G.722 has a much smaller range of support. G.722 is preferred by IT professionals for high-quality voice conversations that don’t place a burden on the local network.
VoIP codecs provide crystal-clear communication
VoIP systems increase productivity by ensuring seamless communication between customers and team members. Voice-over IP codecs allow you to communicate clearly and without having to use cumbersome telecom equipment.
Do not stress about trying to understand the technical details of VoIP codecs. Nextiva is a leader in cloud phone systems, and you can leverage its knowledge from the beginning.
Nextiva knows that clear audio communication is vital for your business and customers. That is why we have listed ten ways to improve the quality of your calls when using the Nextiva App. For Nextiva-specific and general tips, visit our support blog.
Jeremiah Zerby began at the bottom and is now at the top. He worked three years in technical support, troubleshooting issues with the internet and VoIP. He moved up to the technical writing and marketing content creation area. He has helped to set up hundreds of customer accounts, and he can offer advice to anyone with issues or questions regarding their voice service.
