VoIP Codecs | How Audio Codec Improve VoIP Call Quality?
How VoIP Codecs Work with Audio Codecs to Improve Voice Clarity & Efficiency in VoIP Systems
VoIP is more like a blessing for businesses and personal users looking to enhance their communication with their clients. It provides a smart method enabling users to connect via the internet. One of the fundamental parts of this latest technology is VoIP codecs.
It is like software that converts your voice into digital data and transmits it through the internet. Once it reaches the recipient, it converts this data into voice to clarify what the person is trying to say. In short, it is the heart of VoIP, which manages all the VoIP backend functions to ensure smooth communication.
If you want to know more about this tiny software, we have your back. In this guide, we will discuss in detail what VoIP codecs are. How it works, why it is used, VoIP codec list, what features & drawbacks it offers, and lastly, evaluate how it reshapes the overall communication. Stick to last and uncover all the information you need about VoIP codecs.
What is VoIP?
Before jumping straight to VoIP codecs, you must know about VoIP technology. VoIP stands for Voice Over Internet Protocol and is the latest technology used to connect people worldwide over the Internet. Let’s make it easy to understand.
Suppose you have a friend living in Russia, and you want to call him. Unlike traditional phones, This is a highly achievable task without any complexity, where you must go through many processes to establish a link between others.
That’s where VoIP technology comes in. It lets you make calls to your friends by using an internet connection. Moreover, you can talk to him as long as you want without fearing the cost of international calls.
In the past, people used old traditional phones, which were limited to one place and required high costs to manage or to dial calls. But then a change came, which we all know as VoIP.
This intelligent technology removes the limits of using the internet to make calls. This means you need a VoIP phone and a robust internet connection, and you’re all set regardless of your location.
Numerous examples of VoIP technology are available today. Some common examples are Skype, Zoom, Google Meet, Telegram, and WhatsApp. All these apps follow the exact VoIP mechanism and let you connect with your loved ones overseas without hesitancy. Furthermore, you can send them images and videos to tell them about you.
What is a VoIP Codec?
A VoIP codec is a coder-decoder. It is a software or hardware used in the VoIP technology. It processes the main functions to make smooth calls. For example, When you call through a VoIP phone, a codec first converts your voice into digital data. This conversion is crucial for the easy transmission of your message.
Conversion
Compression
Hardware Codecs
Software Codecs
Conversion
Once it converts the voice it sends it by using the internet connection. At the recipient end, the codec acts as a decoder, meaning that after digital data transmission, the VoIP codec works again to convert the digital data into audio. This conversion lets the user hear the voice clearly at the receiving end. In simple terms, it enhances call quality with limited bandwidth.
Let’s understand it by going deeply into the VoIP codecs. The first conversion is where voice signals are converted into digital data. This process is called encoding. This process is known as decoding when it again transforms into voice. The VoIP codec performs both of these tasks.
Compression
In short, the VoIP codec compresses digital data during encoding. This action is done to save bandwidth and ensure crystal-clear calls. After receiving the compressed data, the codec decompresses it and transforms it into a voice at the destination.
Lastly, let me tell you that multiple VoIP codecs are used in VoIP technology. Each codec has its way of handling data compression, which affects audio quality and bandwidth requirements.
Hardware Codecs
Hardware codecs are those codecs that have physical chips. They are used to record, transmit, and play audio and video files. Hardware codecs include computers, cell phones, tablets, laptops, headsets, and VoIP phones.
Software Codecs
Software codecs are offered by the cloud-hosted PBX providers. These software codecs are designed to transmit digital signals across networks. In addition, all the audio and video-based programs use software codecs.
VoIP Codec Comparison Chart
Here is a list of the best VoIP codecs comparison charts.
Codec
Frequency
Processing
Cost
Sample rate
Audio Quality
Strengths
Bitrate
Weaknesses
Latency
Best For
G.711
300-3400 Hz
Low
Free
8,000
High
Simple, low-latency
64 Kbps
High bandwidth, less detail
125 µs
Communication between VoIP and PSTN
G.722
50-7000 Hz
Low
Royalty-free
16,000
High HD
HD audio, natural sound
32 Kbps
High bandwidth, limited devices
4 ms
Good voice quality with low latency
G.722.2
50-7000 Hz
Low
Royalty-free
16,000
High
HD audio, natural sound
48-64 Kbps
High bandwidth, limited devices
25 ms
Compromise between voice quality and flexible bandwidth
G.729
300-3400 Hz
Moderate
Royalty-free
8,000
Good
Low bandwidth, error-tolerant
8 Kbps
Moderate quality, higher latency
15 ms
Low bandwidth usage
Opus
50-20,000 Hz
Moderate
Open Source
48,000
Excellent
Versatile, high quality
6-510 Kbps
Variable quality, complex
26.5 ms
HD voice and clear sound
How VoIP Codecs Works
The working mechanism of VoIP codecs is quite simple and easy to understand. Let’s break it into simple steps.
When you make calls using VoIP phones, your spoken words become analog voice signals.
Sampling
After that, the codec takes “samples” of your voice at specific intervals. This step turns your voice into data that can be processed digitally.
Encoding
Then, the codec compresses this data to reduce its size. The goal is to make the data small enough to travel across the internet quickly.
Transmission
After the compression of digital data, it is sent over the network to the recipient.
Decoding
The receiving device uses the same codec to decompress the data and convert it into an audio signal.
Output
Finally, the decompressed voice signal is played through the receiver’s speaker. This way, you can swiftly hear another person’s voice on your phone.
Different Types of VoIP Codecs
VoIP Codec List is essential to enable voice communication through the Internet. They are primarily used to enhance the call quality, provide fast data transmission, and minimize bandwidth. Here are some common types of Voice Codec List that perform the same function with specific intervals.
Lossy Codecs
Lossless Codecs
Wideband Codecs (HD Voice)
Variable Bitrate Codecs
Narrowband Codecs
Proprietary Codecs
Lossy Codecs
One of the most common types of VoIP codec is Lossy codecs. This is used to compress the digital voice data. It only removes specific parts of the audio which are less noticeable.
Moreover, it also reduces the bandwidth by sending data via the internet. This makes lossy codecs widely used codec in VoIP as it performs both functions swiftly.
Common examples of Lossy Codecs:
G.729 Codec:
The G.729 codec compresses voice data to 8 kbps. This makes it ideal for low-bandwidth environments. It offers less enhanced audio quality but is highly efficient in saving VoIP Codec Bandwidth.
Opus Codec:
This codec is well known for its ability to operate at various bitrates. If you’re a streaming buff, opus codec is best for you as it offers high-quality audio even in low bitrates.
Lossless Codecs
Another popular type is lossless codecs. Its function is slightly different from lossy codecs. This codec type compresses the voice data but preserves the original audio signals for crystal-clear communication.
This way, you can go with lossless codecs if you prioritize high-quality audio. Still, they require more bandwidth than lossy codecs.
Example of Lossless Codec:
G.711 Codec
Voice codec g711 is one of the oldest and most commonly used VoIP codecs. This old-used type encodes voice at 64kbps, ensuring the near-CD audio quality. However, it requires more bandwidth, which may not be ideal in companies with limited network capacity.
Wideband Codecs (HD Voice)
Thirdly, we have wideband codecs, often called HD voice codecs. As the name shows, it provides higher audio quality by transmitting a broader range of frequencies than narrowband codecs.
Narrowband codecs usually transmit frequencies between 300 Hz and 3.4 kHz, while wideband codecs extend this range to 50 Hz – 7 kHz. This enhances the audio more clearly.
Examples of Wideband Codecs:
G.722 Codec
It is a wideband codec that offers higher audio quality than narrowband codecs like G.711. It operates at a 64 kbps bitrate but provides better sound clarity due to its more comprehensive frequency range.
AMR-WB Codec:
AMR-WB, Adaptive Multi-Rate Wideband, is an upgraded AMR codec version widely used in 3G and 4G mobile networks. It also provides a wide range of frequencies, ultimately ensuring high-quality calls.
Variable Bitrate Codecs
The variable bitrates codecs operate using changeable bitrates. Some VoIP codecs, like Opus and SILK, operate using variable bitrates. The codec can adjust the bitrate depending on bandwidth and network conditions. These codecs are helpful in environments with variable network capacity, such as mobile or Wi-Fi networks.
Examples of Variable Bitrate Codec:
VoIP Codec Opus
It is a highly versatile codec that can operate in narrowband and wideband modes but depends on the available bandwidth. It can ensure high-quality audio at both low and high bitrates. This is why it has become one of the most flexible and widely used VoIP codecs.
Narrowband Codecs
These codecs are designed to transmit voice signals within a limited frequency range, such as 300 Hz to 3.4 kHz. While this range is sufficient for essential voice communication, the audio quality is lower than wideband codecs. It is often used with limited bandwidth, and high-quality audio is not a priority.
Examples of Narrowband Codecs:
G.729 Codec
As mentioned earlier, G.729 is a narrowband codec that offers moderate audio quality while maintaining bandwidth. It is commonly used in low-bandwidth environments, such as mobile networks.
Voice Codec G711 Codec
Although G.711 provides excellent audio quality, it is still considered a narrowband codec because it transmits audio within the 300 Hz to 3.4 kHz frequency range.
Proprietary Codecs
Last but not least, we have a proprietary codec. Some VoIP systems use these codecs because they are designed for a particular platform. These also perform the same function along with the ability to better integrate with the system’s infrastructure
Example of Proprietary Codec:
iLBC
iLBC, also known as Internet Low Bitrate Codec, is designed for VoIP applications where packet loss is common. It provides good audio quality even when data packets are dropped during transmission.
In summary, VoIP codecs perform the central functions of VoIP. This makes them the backbone of VoIP technology, also known as internet-based voice communication. Each code has its specialty and features that cater to different needs.
Some save bandwidth as their top priority, while others ensure high-quality audio, making communication more seamless. Understanding these different types of codecs helps you choose the best one based on your communication needs.
Yes, VoIP codecs are designed in a manner to encode and decode voice signals for the tramsission of the data over the network. These codecs specialize in compressing the audio and video signals to meet the minimum bandwidth requirements without disturbing the audio and video quality.
What is Bandwidth in VoIP Codecs?
Bandwidth in VoIP codecs is the amount of data transmitted over a network to carry voice communication. It is measured in kilobits per second (kbps) and directly impacts the quality of a VoIP call.
In simple terms, VoIP codec bandwidth determines how much space or capacity is available for sending voice data. The more bandwidth a codec uses, the higher the audio quality it can provide.
For example:
A codec like G.711 uses 64 kbps of bandwidth to provide high-quality audio but requires a strong internet connection. In contrast, G.729 uses only 8 kbps, making it more efficient in low-bandwidth environments, which moderates sound clarity.
Balancing bandwidth usage with call quality in VoIP is crucial, especially in networks with limited data capacity. For instance, always choose a codec like G.729 or Opus when you have limited bandwidth. Conversely, if you have enough bandwidth, you can go with the G.711 codec, which provides superior audio quality.
What is Latency In VoIP Codecs?
Latency is a delay between when a voice signal is transmitted and when it is received. In short, it slows down the encoding and decoding of audio signals, which produces interruptions & issues such as echo and overlapping in communication. In high latency, you’ll face delays in hearing, while in low latency, you’re all set.
Furthermore, this means low latency can cause smooth conservation. Some codecs are designed to minimize latency by processing data faster and using intelligent compression algorithms. For example, the G.711 codec has a very low latency rate because it does not perform complex compression.
Choosing The Right VoIP Codec
Selecting the right VoIP codecs, consider these factors carefully before deciding. By following these factors, you can evaluate your priorities and make the best decision for your company. Here are some factors you should look at:
Bandwidth Availability
Audio Quality
Network Conditions
Latency
Compatibility
Bandwidth Availability
Start with enough bandwidth. You can go for high-quality codecs like G.711 if you have high bandwidth. In low-bandwidth situations, you may need to choose a codec like G.729 or Opus, which are more efficient.
Audio Quality
Call clarity is a top priority in your business. In that case, you should choose a codec that provides higher audio quality to ensure smooth communication.
Network Conditions
If your network conditions are variable, it’s wise to use adaptive codecs like Opus to adjust the bitrate depending on available bandwidth.
Latency
Real-time communication is essential. Select a codec with low latency to avoid delays in conversations.
Compatibility
The codec must be supported by both the sending and receiving devices to ensure compatibility.
Benefits of Using VoIP Codecs
Using codecs ensures smooth and crystal-clear communication and benefits individuals or a company more. Here are some fascinating features of using VoIP codecs.
Improved Call Quality
Remove Background noise
Efficient Bandwidth
Reduced Latency
Adaptive Performance
Cost-Effective Communication
Multiple Platforms Support
Improved Call Quality
One of the most significant benefits of using codecs is enhancing voice quality during calls. It uses high-frequency ranges, which makes sound more natural.
Remove Background noise
Codec also compresses the audio signals more efficiently, ensuring the removal of background sound or any distortions. This makes it easy to understand conservation.
Efficient Bandwidth
Usage Bandwidth plays a vital role in the smoothness of communication. Codecs help to reduce the amount of bandwidth needed to transmit voice data. This means codec is very beneficial for an environment with limited bandwidth.
Reduced Latency
Codecs help minimize latency to ensure the audio signals reach their destination within time and with precise data. This allows businesses to communicate effectively with clients.
Adaptive Performance
Some modern codecs, like VoIP codecs opus, can work in different network conditions. For example, when internet bandwidth drops or changes, these codecs adjust the compression level to maintain a stable call.
Cost-Effective Communication
Using codecs can also save costs. Since they compress voice data, less bandwidth is used, leading to lower data consumption.
Support for Multiple Devices and Platforms
VoIP codecs are compatible with various devices and platforms, including mobile phones, computers, and VoIP desk phones. Codecs like SILK (used by Skype) and Opus work seamlessly across different devices.
In summary, VoIP codecs can be a blessing for individuals and businesses. It helps to reduce their communication cost, preserving high-quality audio calls. Lastly, it minimizes latency or other glitches to ensure efficient communication.
Common Challenges of VoIP Codecs
Although VoIP codecs have numerous benefits that improve communication overall, besides their features. There are some challenges to VoIP codecs that everyone must be aware of. By resolving these challenges, you can improve the overall communication system.
Bandwidth Limitations
Packet Loss and Jitter
Compatibility Issues
Network Congestion
Security Concerns
Bandwidth Limitations
Some codecs do not work correctly in environments where low bandwidth, such as G.711, demands higher bandwidth to show its action.
Packet Loss and Jitter
Sometimes, the codecs lose the data packet during transmission over the internet, which causes jitter in sound quality (unclear voice).
Different VoIP services may use other codecs, which can lead to compatibility issues. For example, one service might prefer G.711, while another might use G.729.
Network Congestion
Most importantly, VoIP codecs rely on stable internet connections, but the call quality might drop due to network congestion.
Security Concerns
Some VoIP codecs may not provide enough encryption to survive against malware attacks. This is also considered a drawback of some VoIP codecs.
Importance Of VoIP Codecs
VoIP codecs play an essential role in determining the overall quality of VoIP audio. When it comes to cloud-based businesses, the VoIP phone system is dependent on digital real-time audio transmission. This feature allows HD-voice audio quality to eliminate jitter and latency issues during calls.
In addition to the high-quality audio transmission, the data is another key aspect when considering codecs. The data also requires a codec that can convert analog signals into digital ones. At the same time, it minimizes the bandwidth requirement.
And for this reason, A-law codecs are preferred because of their ability to transmit data instantly from analog to digital and keep the bandwidth to a minimum.
The Hopeful Future of VoIP Codecs
As technology advances, the future of VoIP codecs is evolving rapidly. The developers are continuously working with new developments that can improve call quality and efficiency more than before.
Ultra-HD Codecs
AI Integration
Low-Latency Codecs
5G Networks
Enhanced Error Correction
Focus on Energy Efficiency
Cloud-Based Solutions
Ultra-HD Codecs
The audio quality will likely be enhanced in the future, providing richer audio quality. The idea of transforming HD codecs into Ultra HD codecs is already known for its top-notch quality.
AI Integration
As you can see today, AI is integrated into our ecosystem, making things easy to access. In the future, there is a possibility of integration of VoIP phones to revolutionize VoIP codecs. AI can analyze voice data in real-time, resulting in fewer dropped calls and smoother communication.
Low-Latency Codecs
As networks improve, these low-latency codecs will further reduce delays, making conversations feel real and even faster.
5G Networks
With the global rollout of 5G networks, codecs must be optimized for faster, more reliable connections. Many engineers predict VoIP codecs will soon shift to the 5G network, reshaping VoIP’s working speed.
Enhanced Error Correction
Future codecs will improve error correction and noise reduction to ensure more precise voice communication, even in poor network conditions.
Focus on Energy Efficiency
Some VoIP consume more energy, which causes latency or other sound-dropping issues. However, in the future, VoIP codecs must optimize power consumption, especially for mobile devices.
Cloud-Based Solutions
As more VoIP services move to the cloud, future codecs may use the cloud-based technologies to offload processing tasks from local devices.
In short, the future of VoIP codecs is promising and full of exciting advancements. These innovations will continue improving the overall VoIP experience, ensuring it is more transparent and reliable. Lastly, it provides efficient voice communication for users across various industries and platforms.
Final Verdict
A perfect VoIP codec is one that offers some of the optimal sound quality and ensures minimum bandwidth usage. Moreover, to connect a call over a VoIP phone system, both users must have the same codecs in order to enjoy the best call quality. Usually, three top-notch codecs offer premium audio quality. These are G.711, G.722, and the Opus codecs.
Therefore, almost every VoIP phone service provider offers support from multiple codec features for the users. In addition, all these different codecs can seamlessly work together to provide the users with the best call quality and minimum bandwidth utilization.
Frequently Asked Questions
What is the history of VoIP?
VoIP started in the 1990s when engineers found ways to send voice signals over the Internet instead of using traditional phone lines. This impressive discovery made long-distance calling cheaper and easier than before.
Is VoIP outdated?
The simple answer is No; VoIP is a smart technology that continues its journey towards constant evolution, making it possible for the global industry to enjoy its consistent new possibilities with time.
What was the old VoIP program?
The very first VoIP program was Internet Phone by Vocal Tec. It was introduced in the mid-1990s and became overnight. This helped users to make calls over the Internet, which solved the major problem of that time.
What is the name of Microsoft’s VoIP application?
Microsoft’s VoIP application is called Microsoft Teams. It comes with the Microsoft 365 cloud platform with other security and cloud-based features.
Why is everyone switching to VoIP?
People are moving to VoIP because it saves you money and makes you flexible to connect with others. Conversely, old traditional phones lack this feature. Furthermore, it also uses intelligent technologies to provide seamless audio in your calls.
What equipment is needed for VoIP?
To use VoIP, you don’t need as much. You have a device like a computer, smartphone, a stable internet connection, and last but not least a microphone and speakers for better sound quality. With all this equipment, you’re all set to make calls.
Who invented VoIP?
Marian Croak is the person behind the development of Voice Over Internet Protocols and VoIP technology.
