What are VoIP Codecs: List, Benefits & Bandwidth Chart

When exploring VoIP technology, you often come across the term “codec.” But what exactly is a VoIP codec, and why is it crucial for your communication needs? In this blog, we have covered different VoIP codecs, giving you the knowledge to optimize your VoIP setup.

Understanding VoIP codecs will help you make informed decisions about which ones best suit your specific requirements, balancing factors like audio quality, bandwidth usage, and network conditions.

VoIP codec list 

1. G.711
2. G.722
3. G.723.1
4. G.726
5. G.729
6. Opus
7. Speex
8. iLBC
9. SILK
10. AMR-WB
11. GSM

What are VoIP Codecs 

VoIP codecs, short for “coder-decoders,” are integral components in the VoIP communication protocol structure. They are specialized algorithms designed to encode and decode audio signals for transmission over Internet Protocol (IP) networks. 

The main role of a VoIP codec is to enable the efficient transmission of audio data by compressing it into digital packets for transfer over a network and then decompressing it at the receiving end to maintain audio quality. This involves converting analog voice signals into digital forms, enabling clear and reliable voice communication.

Different codecs vary in their encoding techniques, compression efficiency, bandwidth requirements, and quality of audio output. A deep understanding of how these codecs work and their specific characteristics can significantly impact the performance and quality of your VoIP system.

Key Parameters of VoIP Codecs

1/ Bitrate

The bitrate of a VoIP codec refers to the amount of data transmitted per second during a call and is typically measured in kilobits per second (kbps).

Higher bitrates typically translate to better sound quality but require more bandwidth. Balancing bitrate with available bandwidth is crucial for optimizing both audio clarity and network performance.

2/ Sample Rate

The sample rate of a VoIP codec refers to the frequency at which audio signals are sampled per second and is usually measured in Hertz (Hz).

Some of the sampling rates in VoIP codecs are 8,000 Hz (narrowband), 16,000 Hz (wideband), and 32,000 Hz (fullband). Higher sample rates enable a more accurate representation of the original sound, enhancing audio quality.

3/ Compression Techniques

Compression techniques decrease the data volume needed to transmit voice signals, making VoIP calls more efficient. Common methods include:

• Subband Coding: Divides the audio signal into multiple frequency bands and compresses them independently.
• Linear Predictive Coding (LPC): Uses a model of the human vocal tract to predict future audio samples.
• Vector Quantization (VQ): Compresses audio data by converting it into a limited set of predefined vectors.

4/ Latency

Latency in VoIP codecs refers to the delay between the moment an audio signal is spoken and when it is heard on the receiving end. This delay, measured in milliseconds (ms), can be influenced by several factors, including encoding and decoding times, packet transmission time across the network, and buffer delays.

Minimal latency is crucial for maintaining natural, real-time conversation flow, as high latency can lead to noticeable lags and interruptions, hampering the overall communication experience.

VoIP  Codecs benefits 

VoIP codecs offer several benefits, including:

1/ Improved Call Quality: 

One of the primary advantages of VoIP codecs is their ability to significantly enhance your call quality. By efficiently encoding and compressing audio signals, codecs ensure that your voice communications are clear and audible, even over fluctuating network conditions.

Higher bitrate codecs deliver superior sound clarity, creating a more professional communication experience. This leads to better customer service and overall higher efficiency in your communication processes.

2/ Efficient Bandwidth Utilization: 

Codecs reduce the data needed for transmission across your network by compressing audio signals, greatly enhancing efficiency.

This means that even with limited bandwidth, you can maintain high-quality voice communications without experiencing call drops or quality degradation. 

3/ Better Flexibility:

VoIP codecs offer exceptional flexibility, allowing you to choose from various codecs based on your specific communication needs and existing network infrastructure. Therefore, you can select codecs that best balance audio quality and bandwidth consumption, ensuring you achieve optimal performance. 

Additionally, as your business grows and your communication requirements change, you can easily adapt by switching to more advanced codecs or integrating new technologies without significant disruptions. This flexibility ensures that your VoIP system remains reliable and capable of scaling with your business’s demands.

4/ Cost Savings:

Utilizing VoIP codecs can result in significant cost reductions for your business. By reducing the amount of data needed for high-quality voice transmissions, you can reduce the need for extensive bandwidth and expensive network upgrades. You can support more simultaneous calls without investing in costly infrastructure.

Functionality of Codecs in VoIP

VoIP codecs operate by transforming analog voice signals into digital data, which can then be efficiently transmitted across IP networks.

The process starts when you speak on the call, generating an analog audio signal. This signal is then captured by the VoIP system’s analog-to-digital converter (ADC), which samples and quantizes the audio signal, turning it into a stream of digital data.

Once the voice data is in digital form, the codec’s compression algorithm comes into play. Depending on the specific codec used, this algorithm compresses the data by removing repetitions and using predictive models to encode the audio information more efficiently. 

For example, subband coding divides the audio signal into various frequency bands, each of which is separately compressed, whereas linear predictive coding (LPC) utilizes a model of the human vocal tract to forecast future audio samples, reducing the amount of data required.

After compression, the digital voice packets are encoded into a format suitable for transmission over the IP network. These compressed audio packets are transmitted over the internet to the receiver’s VoIP system.

On the receiving end, the reverse process happens: the digital packets are decoded, decompressed, and converted back into analog audio signals through a digital-to-analog converter (DAC). Finally, the signal is played back through the recipient’s speaker, allowing them to hear your voice.

Types of VoIP Codecs 

Understanding the different types of VoIP codecs is crucial for selecting the right one for your needs. Here are some of the most commonly used VoIP codecs:

1/ G.711: 

G.711 is a narrowband codec that is used in VoIP systems. It operates at a standard bitrate of 64 kbps and uses pulse code modulation (PCM) to encode the analog voice signal. G.711 collects the voice data by sampling the audio signal at an 8 kHz rate and quantizing each sample into 8 bits.

This method provides a high level of audio quality that is almost similar to traditional public switched telephone network (PSTN) calls.

Because it does not use compression, G.711 requires more bandwidth compared to other codecs but offers the advantage of low latency.

Related Reading: The difference between VoIP and PSTN

2/ G.722

G.722 is a wideband codec that provides higher audio quality compared to narrowband codecs. It operates at a bitrate of 64 kbps but can also function at lower bitrates of 56 kbps and 48 kbps, depending on the system requirements.

G.722 achieves its high quality by sampling the audio signal at 16 kHz, which covers a broader range of frequencies and results in clearer and more natural sound.

This wideband codec uses Adaptive Differential Pulse Code Modulation (ADPCM) to encode the audio signal. 

3/ G.723.1:

G.723.1 is another narrowband codec, but unlike G.711, it uses compression to significantly reduce bandwidth usage.

It operates at two different bitrates: 6.3 kbps and 5.3 kbps. The 6.3 kbps mode implements a high-quality compression algorithm called MP-MLQ (Multipulse Maximum Likelihood Quantization), while the 5.3 kbps mode uses an ACELP (Algebraic Code Excited Linear Prediction) algorithm.

This dual-rate capability makes G.723.1 highly adaptable to various network conditions. Despite the lower bitrate, G.723.1 maintains reasonable audio quality but with increased latency and complication compared to G.711.

It is particularly valuable in scenarios where network bandwidth is limited yet dependable voice communication remains necessary.

4/ G.726:

G.726 offers a balance between audio quality and bandwidth efficiency. It operates at multiple bitrates, typically 16 kbps, 24 kbps, 32 kbps, and 40 kbps, allowing flexibility in choosing the best balance between quality and bandwidth. G.726 uses Adaptive Differential Pulse Code Modulation (ADPCM) to encode the analog voice signal.

This approach predicts the next audio sample’s value from previous samples and transmits only the difference between the predicted and actual values. As a result, G.726 ensures compression while maintaining reasonably good audio quality. 

5/ G.729:

G.729 operates at a bitrate of 8 kbps, G.729 uses Conjugate-Structure Algebraic Code Excited Linear Prediction (CS-ACELP) for its compression algorithm.

This method effectively reduces the amount of data needed to represent the voice signal by predicting and encoding only the essential components of the audio. Due to its low bitrate, G.729 significantly saves

bandwidth, making it suitable for VoIP applications in environments with bandwidth limitations. 

6/ Opus:

Opus can handle both narrowband and wideband audio with bitrates ranging from 6 kbps to 510 kbps. This means it can deliver clear sound in different network conditions, from low to high quality.

Opus is utilized in a range of applications, including video conferencing, streaming services, remote collaboration tools, and voice-over IP (VoIP).

Due to its adaptability and superior audio performance, it has become a better option for current communication requirements.

7/ Speex:

Speex is a codec designed specifically for compressing voice data. It operates efficiently in low-bitrate scenarios, making it suitable for VoIP applications. Speex supports a range of bitrates from 2.15 kbps to 44 kbps, allowing flexibility depending on the required audio quality and available bandwidth.

It uses variable bitrate (VBR) which adjusts the bitrate based on the complexity of the voice signal, ensuring optimal performance even when network conditions change.

Speex also includes features like echo cancellation and noise suppression, enhancing the clarity of the transmitted voice.

8/ iLBC:

iLBC is a codec that is designed to compress voice data effectively at low bitrates. It works efficiently at bitrates of 13.33 kbps and 15.2 kbps.

This codec is useful in networks with limited bandwidth, ensuring clear voice communication even when data capacity is limited. iLBC maintains voice quality under different network conditions, making it reliable for VoIP applications.

9/ SILK:

SILK is a wideband codec developed by Skype, known for its high audio quality and efficiency. It operates at bitrates from 6 kbps to 40 kbps and is designed to handle different network conditions with ease.

SILK adjusts its bitrates dynamically to ensure clear and smooth voice calls, even when the network is not stable. This codec is widely used in VoIP and online communication services because of its superior performance and adaptability.

10/ AMR-WB:

AMR-WB provides great audio quality and uses a flexible bit rate to adapt to different network conditions.

It operates at a sampling rate of 16 kHz, which records more details of the voice, making it sound clearer and more natural. This codec is often used for high-definition voice calls.

11/ GSM:

GSM is a standard codec for mobile phones, ensuring clear voice calls. It uses a fixed bitrate of 13 kbps. GSM provides reliable voice quality by efficiently using available network resources.

It is extensively utilized in mobile networks globally, making it a universal choice for mobile voice communication.

Factors Influencing Codec Selection 

1/ Bandwidth Requirements:

The available bandwidth significantly influences codec choice. High-bitrate codecs like G.711 require more bandwidth but offer superior audio quality, while low-bitrate codecs like G.729 are suitable for bandwidth-limited environments.

2/ Voice Quality:

Voice quality is essential in business communications. Wideband codecs such as G.722 and Opus provide higher audio accuracy, making them ideal for applications where clarity is essential.

3/ Latency:

For real-time communications, low latency is critical. Codecs like Opus and SILK are designed to minimize delay, ensuring smooth and natural conversations.

4/ Packet Loss and Error Resilience:

Packet loss and error resilience are crucial factors in codec selection. In environments where network stability is a concern, choosing a codec that can handle packet loss effectively is vital.

Codecs like Opus and iLBC have strong error durability, ensuring clear and understandable voice transmission even when packets are lost or network conditions fluctuate.

Applications of VoIP codecs

VoIP codecs are used in various applications across different industries. Some of the common applications include:

I. Enterprise VoIP Systems

Many businesses implement VoIP systems to improve communication efficiency and reduce costs. Codecs like G.711 and G.729 are commonly used in enterprise VoIP systems to balance audio quality and bandwidth usage.

II. Mobile VoIP

Mobile VoIP applications enable voice communication over mobile data networks. Codecs like AMR-WB and SILK are used to provide high-quality audio while optimizing bandwidth usage on mobile devices.

III. Consumer VoIP Services

Consumer VoIP services, such as Skype and WhatsApp, rely on codecs like Opus and SILK to deliver clear and reliable voice communication over the Internet.

Related Reading: Differences between business and residential VoIP

VoIP codec bandwidth chart

VoIP codec comparison chart

This table provides a comparative overview of various VoIP codecs based on, audio quality, bandwidth usage, and common use cases.

Frequently Asked Questions 

Q1) What are the best codecs for VOIP? 

Ans: When it comes to selecting the best codec for VoIP, G.722 is a top choice. G.722 offers a bitrate range of 48-64 kbps and provides wideband audio for clearer and more natural sound. It is ideal for business meetings and customer service calls where voice clarity is essential.

Q2) Which Codec is better G711 and G729? 

Ans: if you have sufficient bandwidth and prioritize superior audio quality, G.711 is the better choice. On the other hand, if you need to optimize bandwidth usage and can adjust slightly on voice quality, G.729 would be more suitable.

Q3) What is the difference between g722 and g711 codecs? 

Ans: The main difference between G.722 and G.711 codecs is that G.722 is a wideband codec that operates at bitrates ranging from 48 to 64 kbps and provides superior audio quality with a broader frequency range. On the other hand, G.711 is a narrowband codec with a fixed bitrate of 64 kbps, offering high audio quality that is suitable for traditional telephony but without the extended frequency range provided by G.722.

Q4) What is the best free VoIP codec? 

Ans: The Opus codec is the best free VoIP codec available. It supports an extensive range of bitrates, ranging from 6 to 510 kbps, providing better flexibility and audio quality. Opus is developed to handle both narrowband and wideband communications, making it suitable for various applications such as video conferencing, VoIP calls, and streaming services.

Q5) What is the audio codec in VoIP? 

Ans: An audio codec in VoIP is a software or hardware-based tool that compresses and decompresses digital audio data packets transmitted over a network.

Q6) What is the best VoIP Codec for high latency? 

Ans: The best VoIP codec for high-latency environments is Opus. Opus is highly flexible and responsive, offering excellent audio quality even under challenging network conditions with high latency. It can efficiently manage jitter and packet loss, ensuring that voice communications remain clear.