Call quality is one of the main reasons why some businesses are still hesitant to fully switch from traditional landlines to modern communication technologies like Voice over Internet Protocol (VoIP).
While VoIP offers cost savings, flexibility, and advanced features, concerns about audio clarity remain a common barrier. That’s where codecs come into play as essential tools that compress, transmit, and decompress audio signals over the internet, ensuring that voice communications remain clear and uninterrupted.
They have a critical role in balancing call quality, bandwidth efficiency, and overall performance. However, not every codec is the same.
Understanding how codecs work, and which one best fits your business needs, can make all the difference in your transition to modern communication.
Let’s dive deeper into the role of VoIP codecs and how they directly impact your call quality!
What Are VoIP Codecs?
A VoIP codec is a core technology that impacts the audio quality, bandwidth efficiency, and compression level of VoIP phone calls. The term “codec” is a combination of “compression” and “decompression,” reflecting its main function: to encode and decode audio streams during a call.
There are two main types of audio codecs:
- Hardware audio codec: A physical component that handles the real-time conversion of analog audio signals into digital data, and vice versa. It’s commonly found in devices like phones, VoIP adapters, and sound cards.
- Software audio codec: A program or algorithm that digitally encodes and decodes audio streams, allowing voice data to be transmitted over the internet with minimal delay and optimal efficiency.
VoIP codecs can be built using either proprietary or open-source algorithms, depending on the provider or platform. Selecting a suitable codec directly influences the clarity of voice communications and how efficiently your internet bandwidth is used during calls.
RELATED: What is VoIP Phone Service: The Complete Guide
Why are VoIP codecs important?
VoIP codecs are crucial because they determine the audio quality and performance of calls. In cloud-based business communication, real-time audio transmission enables HD voice quality while minimizing issues like jitter, latency, and distortion.
During a call, a codec rapidly converts analog audio signals into digital data, preserving sound quality while optimizing bandwidth usage. If a codec compresses audio too slowly or requires too much bandwidth, you may experience delays or poor sound quality.
Need help with other call-related issues? Check out the video below:
How Do VoIP Codecs Work?
VoIP codecs enable real-time communication by quickly converting and transmitting audio signals over the internet. The process begins when a speaker’s analog audio signal is captured and compressed into digital data packets.
These packets travel across a VoIP telephony network and are then decompressed back into clear, audible sound on the recipient’s device. The step-by-step process of this system is as follows:
- Capturing the Speaker’s Analog Audio Signal
- Compressing the Analog Signal into Digital Packets
- Transmitting the Digital Data Across the Network
- Decoding and Delivering the Audio to the Recipient
Below, we take a closer look at how VoIP codecs work in a cloud-based phone system:
Step 1: Capturing the Speaker’s Analog Audio Signal
When a user speaks into their device, the hardware codec captures the audio as an analog signal. Human speech typically falls between 80 to 300 Hz, while other sounds can span from 20 to 15,000 Hz.
Higher-quality codecs capture a wider range of frequencies, producing high-definition voice that closely resembles natural, in-person conversation.
Step 2: Compressing the Analog Signal into Digital Packets
The codec compresses the analog signal into digital audio data packets. This step reduces the bandwidth needed for transmission. The quality of the compressed audio depends on three factors:
- Frequency: The audio range captured; wideband frequencies deliver richer sound but require more bandwidth.
- Sample rate: The number of audio samples taken per second; higher rates produce smoother, clearer audio.
- Bitrate: The amount of data contained in each sample; higher bitrates preserve more audio detail.
Each VoIP codec – such as G.711 codec or G.722 – has its own combination of frequency, sample rate, and bitrate, directly impacting call clarity and minimizing issues like jitter and latency.
Step 3: Transmitting the Digital Data Across the Network
After encoding, the caller’s VoIP provider sends the digital packets through its internet-based telephony network. Global data centers help route the audio data efficiently, ensuring minimal delay and maintaining conversation flow.
Step 4: Decoding and Delivering the Audio to the Recipient
The recipient’s VoIP provider receives the digital packets and uses the codec to decode them back into an analog audio signal. The device then plays the clear, real-time voice. Both parties must use the same codec to ensure compatibility and maintain call quality.
What Are The VoIP Codec Types?
VoIP codecs come in different types, each balancing bandwidth usage and call quality in distinct ways. Some prioritize maintaining high sound quality, while others focus on compressing audio data to maximize network efficiency.
Below is a summary of the main codecs commonly used in VoIP systems:
| Codec | Description |
| G.711 | A narrowband codec offering high sound quality without compression, requiring more bandwidth. |
| G.722 | A wideband codec that delivers higher audio quality with moderate compression. |
| G.729 | A codec optimized for bandwidth efficiency through higher compression, with some audio quality loss. |
| GSM | Originally developed for mobile communications, it offers moderate audio quality at lower bitrates. |
| T.30 | A legacy fax codec designed for traditional telephone networks. |
| T.38 | A modern fax codec optimized for transmitting faxes over IP networks without a phone line. |
Deep dive into voice codecs
G.711
G.711 is one of the earliest and most widely used narrowband voice codecs. Developed in 1972, it operates at an 8 kHz sampling frequency and a bitrate of 64 Kbit/s. Unlike other codecs, G.711 does not compress audio data, resulting in higher sound quality but greater bandwidth requirements.
There are two versions of G.711:
- G.711u: Commonly used in Japan and North America.
- G.711a: Used in most other countries.
Because it doesn’t compress audio, G.711 provides excellent clarity, though it demands more from your internet connection compared to modern compressed codecs.
G.722
G.722, introduced in 1988, is a wideband codec designed to enhance voice quality compared to G.711. It doubles the sampling frequency to 16 kHz and uses 14 bits per sample, capturing a broader range of audio frequencies for clearer conversations.
Initially, G.722 produces an uncompressed bitrate of 224 Kbit/s, but through compression techniques, it reduces the final bitrate to 64 Kbit/s.
G.729
G.729 codec is a popular Voice over IP codec known for its efficient use of bandwidth. It encodes voice into frames of ten milliseconds each, containing 80 audio samples per frame. This codec requires only 8 Kbit/s per direction, enabling many simultaneous calls over the same network without overloading it.
While G.729 offers impressive compression, the trade-off is lower audio quality compared to G.711. However, it remains ideal for bandwidth-constrained environments.
GSM
GSM stands for Global System for Mobile Communications and was originally developed for mobile phone networks. The GSM codec operates at a bitrate of 13 Kbit/s for Full Rate (GSM-FR) and 6.5 Kbit/s for Half Rate (GSM-HR).
Although it offers decent compression, GSM audio quality is considered lower compared to modern VoIP codecs, making it less common for business-grade VoIP services today.
Fax Codecs
Just like voice communication, fax transmission over IP networks also requires specific codecs.
Learn more about faxing online on the video below:
- T.30: Developed before the rise of the internet and is the traditional protocol for fax communication over Public Switched Telephone Networks (PSTN). It allowed documents to be transmitted directly between fax machines using telephone lines.
- T.38: With the shift to IP-based networks, the T.38 codec was introduced to reliably transmit fax data over the internet. Today, it enables users to send and receive faxes without a traditional phone line, even allowing integration with email for greater flexibility.
RELATED: Check our Wiki to learn which codecs VoIP.ms supports
How Do I Choose The Right Codec For My VoIP System?
Choosing the right VoIP codec is essential to ensure your business telecommunications is clear, reliable, and efficient. As we’ve already learned, the ideal codec depends on several factors, such as your internet bandwidth, network infrastructure, device compatibility, and even the cost structure of your VoIP provider.
Here are the key variants to consider when selecting a VoIP codec:
- Bandwidth Requirements
- Use Case (Voice vs. Fax)
1. Compatibility
Ensure the codec you choose is supported by your VoIP provider, hardware (phones, headsets), and software applications. Using incompatible codecs can lead to failed calls or poor audio performance.
2. Bandwidth Requirements
Evaluate your available internet bandwidth. Some codecs like G.711 require more bandwidth to deliver high-quality audio, while others like G.729 use aggressive compression to save bandwidth at the expense of some sound quality.
3. Audio Quality
Decide how important HD voice quality is for your business needs. If call clarity is critical — for example, in customer service or sales — a wideband codec like G.722 might be ideal.
4. Latency and Jitter
Low latency and minimal jitter are crucial for seamless conversations. Some codecs handle network inconsistencies better than others, impacting the real-time quality of your VoIP calls.
5. Use Case (Voice vs. Fax)
If you need to handle fax transmissions, choosing appropriate fax codecs like T.38 is necessary in addition to voice codecs. Not all codecs support fax data transmission effectively.
RELATED: Fax over IP (FoIP) using T.38 Protocol
Choosing the Perfect VoIP Codec
Choosing the right VoIP codec for your business is a crucial decision that impacts both call quality and network efficiency.
Whether you prioritize natural-sounding voice quality or need to optimize bandwidth usage, understanding these aspects will help you make a well-informed choice that supports your business communication needs.
If you’re unsure about which codec to select or need expert guidance tailored to your specific requirements, our team is here to help.
Contact our support team at [email protected] today to discuss your needs and get personalized recommendations for the best VoIP solutions for your business!
VoIP Codec FAQ
Still have questions? You’re not alone! Understanding VoIP codecs can feel like decoding a secret language, but don’t worry, we’ve got your back.
Let’s dive into these frequently asked questions:
1. Which codec is best for VoIP?
The best codec for VoIP depends on your specific needs. If you prioritize high-quality voice calls, G.711 and G.722 are excellent options, as they provide crystal-clear audio.
However, if you need to conserve bandwidth, G.729 is a great choice, offering efficient compression at the cost of slightly reduced sound quality. For mobile use, GSM may be suitable, though it doesn’t deliver the same quality as modern codecs.
2. How do codec choices affect VoIP bandwidth usage?
Different codecs require varying amounts of bandwidth based on their compression techniques. The codec you choose will impact how many simultaneous calls your network can handle and the quality of each call. That’s why VoIP techs frequently talk about high our low bandwidth for calls.
3. How to fix issues with codecs in VoIP systems?
Codec-related issues in VoIP systems, such as poor audio quality, latency, or jitter, can often be resolved by adjusting the codec settings. Ensure that the codec selected is compatible with both ends of the call.
If you experience frequent issues, consider switching to a codec with better compression or higher bandwidth efficiency. Also, check your network’s quality and ensure there’s no congestion or interference affecting the transmission.
4. What’s the difference between narrowband and wideband codecs?
Narrowband codecs, like G.711, transmit a limited range of audio frequencies. These codecs are more bandwidth-efficient but offer lower audio quality. Wideband codecs, such as G.722, capture a broader frequency range, providing richer, more natural sound but requiring more bandwidth.
5. How do codecs impact VoIP call quality when recording or using voicemail features?
Narrowband codecs may make recorded messages sound less clear and more robotic, while wideband codecs deliver higher-quality recordings, making them easier to understand.
If voicemail clarity is critical for your business, opting for a wideband codec may improve the overall experience.