If you want a crystal-clear call that only requires an internet connection, think about Voice over Internet protocol (VoIP). With these codecs, you can have reliable and cost-effective calling for your business.
Choosing the right codecs is vital for your enterprise. Here are a few things that you need to know about VoIP and these codecs.
Are you searching for cloud-based communication solutions for your business or enterprise? G12 Communications has several options that can help you communicate and collaborate from anywhere!
What Are VoIP Codecs?
A VoIP codec determines the bandwidth, audio quality, and compression of a phone call. These codecs are either open-source algorithms or proprietary. Within the word codec, you will notice two terms: compression and decompression.
If you have downloaded a film in a manner of minutes, you can thank those codecs. These codecs are used in various ways, including encrypting data, streaming media, capturing images, and recording audio. When you watch Netflix or Youtube, codecs determine the bandwidth and quality of the files.
In a VoIP codec, analog voice signals are converted into digital packets for transmission and then converted back into an uncompressed audio signal. These codecs determine the latency and quality so that the call can be placed from the internet.
Since these calls travel over the internet, VoIP can run into a few problems. However, many VoIP providers use multiple data centers to transmit most of the calls, and there are few reliability issues.
All codecs are created with one purpose: to compress data and move it quickly. Businesses that use VoIP applications know that these codecs will not use a ton of bandwidth and can deliver crisp and clear calls.
VoIP and Audio Quality
Millions of businesses need a reliable VoIP system, which is why audio quality is a top priority. If you want to compare audio quality, you should be familiar with these terms:
- Sample rate: Refers to audio samples taken over one second. These individual samples provide the signal waveform’s total value over a set time. A higher sample rate means better audio quality.
- Bitrate: The amount of data that can be transferred into the audio. These bitrates capture information per second. A higher bitrate means a better sound quality.
- Bandwidth: The speed at which data is sent or received. A high transmission rate means that you can send more samples per second.
Low bitrates and sample rates lead to poor quality of sound. Bandwidth is your bottleneck. With the right VoIP codecs, that can help to conserve that precious bandwidth while maintaining high-quality sound.
Improving Call Quality
The quality that you find in a typical phone call includes two bands: narrowband and wideband. With a narrowband, it covers frequencies from 300 Hz to 3400 Hz. Any audio between 50 Hz and 7000 Hz is considered wideband.
When you talk in those ranges for wideband, that is commonly referred to as HD voice. You can hear the discussion in a wide range of pitches that resembles an in-person conversation. VoIP codecs are used to help make those crystal-clear calls without draining your bandwidth.
Types of VoIP Codecs
When looking for the right VoIP codec, there are plenty of choices. Some of the most popular options include:
These codecs were developed back in 1972, and they have two variants: μ-law and A-law. μ-law is commonly found in the United States. With this codec, 16-bit samples can be compressed into 8 bits with a compression ratio of 1:2. This codec’s bitrate for both directions is 128 kbit/s.
While the bandwidth requirement is high, you do get a superior quality of sound. Since there is no digital compression, this codec is one of the best choices for those who need to interface with a public switched telephone network (PSTN).
If you want a high-definition codec for wideband, then you should look at G.722 HD. This codec was approved in 1988, and it is free to use for everyone. Without any perceivable latency, this codec can help to improve the speech quality of your calls. Its HD voice has a sample rate of G.711 at 16 bits, but the transmission rate remains the same at 64 kbit/s.
For those who want acceptable audio quality with low bandwidth requirements, the G.729 is a great choice. Audio is encoded in a frame, and each frame contains about ten milliseconds of audio with about 80 samples. The single direction bitrate for this non-HD codec is 8kbit/s. With a higher compression rate, you can make more network calls at one time. However, not every VoIP can support those G.729 codecs.
If you need to organize those codecs, you can use cloud-based services from your VoIP provider. These providers transmit the data packet, and your IP phone can decompress and compress the audio. When it comes time to find the correct codec, you want to consider your call volume and bandwidth capabilities.
For those who want crystal clear call quality, think about G.722 and G.711. However, if you’re going to prioritize your bandwidth, then G.729 is a better choice.
Crystal-Clear Communication With VoIP Codecs
By using VoIP systems, you can boost your productivity with seamless communication between your customers and team. These codecs allow everyone to speak clearly without the need for bulky telephones and other equipment.
If you are searching for the correct codec for your VoIP system, you may want to select a cloud-based phone system to help your business communicate more clearly.
Do you need an effective way to communicate with your customers and team? G12 Communications is your next partner in everything for cloud communications. Make sure to reach out to find the best solutions for your organization, business, or other enterprises!