VoIP Codecs: Everything You Should Know

A codec is a combination of two algorithms that compresses and decompresses voice signals to ensure efficient transmission over networks. The term "codec" is formed by merging the words "coder" and "decoder".

VoIP codecs convert analog voice signals into digital data, compress the data for transmission, and then decompress them on the receiving side, ensuring the voice is heard clearly.

As previously noted, the VoIP codec converts audio signals into digital data packets that are transmitted over the Internet before decoding them back into sound.

Here is a breakdown of the procedure.
 

1. Converts of Analog Data to Digital: This is the first step in the process. Here, analog signals, such as sound waves or video signals, are converted into digital signals. This process is called encoding.
    
2. Compress and transfer digital data into small packets: The encoded data is then divided into packets, making it easier to transfer it over a network to the receiver.
    
3. Converts Digital Data Back to Analog: When it reaches the receiver’s end, the digital data is converted back into analog signals so the receiver can hear the voice. This process is called decoding.

Some popular VoIP codecs include narrowband, wideband, and opus codecs.

As the name suggests, narrowband codecs use a narrower range of frequencies and typically offer lower sound quality than other variations of codecs. Because they require less data to be transmitted, they are best suited for the Internet with limited bandwidth. This codec's sample rate is 8,000 samples per second, or 8kHz.

Furthermore, they are the ITU Telecommunication Standardization Sector standard speech codecs that use Adaptive Differential Pulse Code Modulation (ADPCM), operating within the frequency between 300Hz and 3.4kHz. However, there’s an exception: G.711 covers frequencies between 300Hz and 3.4kHz using standard Pulse Code Modulation (PCM).
 

G.711:

Known as a standard codec, G.711 offers audio of 64 kbps. It uses either A-law or μ-law, a compression algorithm that reduces the required bandwidth for transmission and delivers high-quality sound due to its high bitrate. However, G.711 consumes more bandwidth and is considered the best codec for interacting with PSTN (Public Switch Landline Network).
 

G.723:

Introduced in 1988, G.723 is a low-compression codec used for VoIP calls in low-bandwidth situations. It compresses the audio at a low bit rate (5.3 kbps or 6.3 kbps). G.723 is considered a "dual-rate" codec, as it can operate at two different bitrates, i.e., 5.3 and 6.3.
 

G.729:

Popular for its balance between audio quality and bandwidth usage (typically 8 kbps), G.729 has a low bitrate, which means this codec consumes less bandwidth. Many VoIP apps use this codec when bandwidth needs to be preserved.
 

G.726:

G.726 offers compression at 40, 32, 24, and 16 kbps. It's used in situations requiring bandwidth efficiency but at the cost of some sound quality. G.726 is a waveform speech coder, which means that it encodes the waveform of the speech signal directly.

Wideband codecs provide higher sound quality by using a broader frequency range, capturing more of the audio spectrum, and making voices sound more natural. Its sample rate is double than that of a narrowband codec, i.e., 16,000 per second or 16kHz.

Wideband codecs usually transmit frequencies ranging from 50 Hz to 7 kHz, but they also can transmit frequencies ranging from 20 Hz to 20 kHz for high-definition audio.
 

G.722:

G.722 is an ITU-T standard audio codec that provides 16 kHz audio, delivering a richer, more natural-sounding voice. It is commonly used in high-quality VoIP and video conferencing systems. The codec has a high bitrate and high bandwidth requirements.
 

G.722.2 (Adaptive Multi-Rate Wideband):

G.722.2 is designed for mobile communications. It provides high-quality audio over narrowband and wideband channels and can efficiently handle varying network conditions. It provides a wider audio bandwidth, typically ranging from 50 - 7,000 Hz.

Opus Codec is a highly versatile audio codec that supports both narrowband and wideband audio. It adapts well to different network conditions, offering high-quality voice and low latency, making it ideal for VoIP and real-time communication. It supports bandwidth ranging from 6 kbps to 510 kbps. Likewise, its sampling rate ranges from 8 kHz up to 48 kHz per second.

Codecs improve VoIP call quality by compressing and decompressing audio data, optimizing the balance between sound quality and bandwidth usage. Different codecs offer varying compression levels, allowing VoIP systems to adapt to network conditions and ensure clear, low-latency communication, even on bandwidth-constrained connections. This adaptability helps maintain high call quality, minimizing issues like distortion, jitter, and packet loss.
 

To select the highest quality VoIP codec, consider your bandwidth requirements, bitrate, sample rate, and mean opinion score.

The speed of receiving and sending data is called bandwidth, and different codecs have different bandwidth requirements. For instance, narrowband codecs like G.729 require less bandwidth, while wideband codecs like G.722 require more. Hence, choosing a codec that aligns with your available bandwidth is crucial for maintaining call quality.

Bitrate refers to the amount of data transmitted per second. A higher bitrate typically translates to better sound quality, but it also uses more bandwidth. For example, G.711 operates at 64 kbps, providing excellent sound quality but consuming more bandwidth, whereas G.729 operates at eight kbps, with lower quality but more bandwidth efficiency.

The sample rate is the number of samples taken per second when converting analog audio to digital form. Higher sample rates allow for a wider range of frequencies to be captured, which improves the sound quality of the call. Wideband codecs typically have higher sample rates (e.g., 16 kHz for G.722) than narrowband codecs (e.g., 8 kHz for G.711).

MOS is a numerical rating used to evaluate the perceived quality of a call. It ranges from 1 (poor quality) to 5 (excellent quality). Higher MOS scores usually indicate better codec performance. Wideband codecs tend to provide higher MOS scores than narrowband codecs, as they capture a broader range of frequencies and produce crystal clear sound.

VoIP codec plays a pivotal role in determining the quality, efficiency, and overall user experience of VoIP calls. However, you must understand your bandwidth capability, alongside bitrate, sample rate, and MOS requirements, to choose the best VoIP codec, whether for personal calls or business communication.

With Calilio, you do not have to worry much about technical details regarding VoIP codecs. Calilio uses advanced codec technologies to deliver clear and stable audio, even in challenging network conditions. Our telephony system easily adapts to bandwidth fluctuations to maintain high call quality and ensure continuous connectivity all the time.

G.711 offers better audio quality as it transmits uncompressed audio at 64 kbps. However, G.729 is more bandwidth-efficient (8 kbps), making it better for environments with limited bandwidth but compromising some call quality.

The best VoIP codec depends on the use case. Wideband codecs like G.722 are ideal for high-quality calls. However, G.729 is a good choice for environments with limited bandwidth due to its lower bandwidth requirements.

Opus can adapt to different network conditions by adjusting its bitrate across a wider range, while G729 typically operates at a very low, fixed bitrate.

An audio codec converts audio signals from an analog format to digital data for transmission and then back to audio at the receiver’s end.