Explained: VoIP Audio Codecs

NOTE: Real Bandwidth figures take into account packet overheads when running over an Ethernet network. These are approximate measures and can change depending on the type of network the codec runs over. 

The G.722 codec is a standard audio compression algorithm defined by the International Telecommunication Union (ITU) in its G.722 recommendation. It is commonly used in Voice over Internet Protocol (VoIP) applications, video conferencing, and telephony systems to encode high-quality audio signals.

G.722 is often referred to as a "wideband" codec because it captures a broader frequency range compared to traditional "narrowband" codecs like G.711.

Key features of the G.722 codec include:

• Bit Rate: G.722 operates at a higher bit rate than narrowband codecs, providing better audio quality. It has a fixed bit rate of 64 kbps.
• Audio Quality: G.722 is known for delivering high-fidelity audio with a frequency range of 50 Hz to 7 kHz. This wider frequency response contributes to clearer and more natural-sounding voice communication.
• Application: G.722 is suitable for applications where enhanced audio quality is desired, such as in professional communications, conferences, and situations where clarity is crucial.
• Compatibility: G.722 is widely supported in VoIP hardware and software, making it a popular choice for organizations seeking improved audio quality in their communication systems.
• Frame Size: G.722 typically uses 20-ms frame sizes, meaning that the audio signal is divided into frames, each lasting 20 milliseconds. This helps in maintaining low latency for real-time communication.

While G.722 provides superior audio quality, it's important to note that the PSTN uses the G.711 codec. This means that the benefits of G.722 are not experienced for calls running over the PSTN.

The G.711 codec is a standard audio compression algorithm defined by the International Telecommunication Union (ITU) in its G.711 recommendation. It is widely used in Voice over Internet Protocol (VoIP) applications, telephony systems, and other voice communication technologies. G.711 is a "narrowband" codec, meaning it captures a limited frequency range compared to "wideband" codecs like G.722. The two main variants of G.711 are μ-law (pronounced "u-law") and A-law. U-law is commonly used in North America and Japan while A-law is used in Europe and other parts of the world. The difference between the two is negligible.

Key features of the G.711 codec include:

• Bit Rate: G.711 operates at a fixed bit rate of 64 kbps (kilobits per second). This bit rate ensures a constant and predictable transmission rate for voice signals.
• Audio Quality: G.711 provides good audio quality and is often referred to as "PCM" (Pulse Code Modulation). It is commonly used in traditional telephony systems and is known for its simplicity and reliability.
• Application: G.711 is suitable for applications where bandwidth is not a limiting factor, such as in environments where dedicated circuits or ample network capacity are available. It is commonly used in traditional landline telephony and is supported by many VoIP systems.
• Compatibility: G.711 is a widely supported codec in the VoIP industry, making it interoperable with various hardware and software devices. It is often used for communication between different VoIP systems.
• Frame Size: G.711 typically uses 20-ms frame sizes, where the audio signal is divided into frames, each lasting 20 milliseconds. This helps in maintaining low latency for real-time communication.

The GSM (Global System for Mobile Communications) codec is a standard audio compression algorithm used in telecommunications and Voice over Internet Protocol (VoIP) applications. It is designed to encode and compress speech signals for efficient transmission over digital networks. The GSM codec is commonly used in mobile communications and is also implemented in VoIP systems.

Key features of the GSM codec include:

• Bit Rate: The GSM codec operates at a bit rate of 13 kbps (kilobits per second). This relatively low bit rate allows for efficient use of network bandwidth.
• Audio Quality: GSM is a "narrowband" codec, optimized for speech compression within a limited frequency range. While it may not provide as high audio quality as some wideband codecs, it is suitable for voice communication in mobile and VoIP applications.
• Application: GSM is widely used in mobile networks for cellular voice communication. In the context of VoIP, it is often employed for scenarios where conserving bandwidth is crucial.
• Compatibility: The GSM codec is well-supported in VoIP equipment and software, making it interoperable across various platforms.
• Frame Size: GSM typically uses 20-ms frame sizes, where the audio signal is divided into frames, each lasting 20 milliseconds. This frame size helps maintain low latency for real-time communication.

The iLBC (internet Low Bitrate Codec) is a VoIP codec designed for narrowband speech compression, suitable for real-time communication over networks with limited bandwidth. iLBC was developed by Global IP Solutions (now part of Google) and is standardized by the Internet Engineering Task Force (IETF) in RFC 3951 and RFC 3952.

Key features of the iLBC codec include:

• Bit Rate: iLBC operates at two bit rates: 15.2 kbps (Framesize 30 ms) and 13.33 kbps (Framesize 20 ms). These bit rates provide flexibility for different network conditions and application requirements.
• Application: iLBC is particularly well-suited for VoIP applications where bandwidth is constrained. It is commonly used in scenarios such as video conferencing, online gaming, and VoIP telephony.
• Frame Size: iLBC uses either 20-millisecond or 30-millisecond frame sizes, allowing for adaptability to different network conditions and latency requirements.
• Packet Loss Resilience: iLBC is designed to handle packet loss effectively. It employs techniques such as PLC (Packet Loss Concealment) to mitigate the impact of lost packets on perceived audio quality.
• Free and Open Source: iLBC is an open-source codec, meaning it can be used without incurring licensing fees. This characteristic has contributed to its adoption in various VoIP applications.

Due to its ability to perform well in adverse network conditions and its open-source nature, iLBC has found applications in real-time communication systems where bandwidth conservation is crucial.

The G.729 codec is a widely used audio compression algorithm designed for efficient voice communication over digital networks. It is part of the ITU-T G-series recommendations and is specified in the G.729 standard. G.729 is commonly used in Voice over Internet Protocol (VoIP) applications, enabling the transmission of high-quality voice signals with a relatively low bit rate.

Key features of the G.729 codec include:

• Bit Rate: G.729 operates at a bit rate of 8 kbps (kilobits per second). This relatively low bit rate allows for efficient use of network bandwidth.
• Audio Quality: G.729 provides good audio quality within the constraints of its low bit rate. It is classified as a narrowband codec, optimized for speech compression within a limited frequency range.
• Application: G.729 is widely used in VoIP applications, especially in scenarios where bandwidth conservation is crucial. It is often chosen for international calls and networks with limited capacity.
• Compatibility: G.729 is well-supported in VoIP hardware and software, making it interoperable across various platforms. However, it is important to note that G.729 is a proprietary codec, and its use may be subject to licensing fees.
• Frame Size: G.729 typically uses 10-millisecond frame sizes, where the audio signal is divided into frames, each lasting 10 milliseconds. This frame size helps maintain low latency for real-time communication.

In most cases, sticking with G711 as the default is a wise course of action. In situations where bandwidth is at a premium, GSM is a great free option which offers impressive audio quality for both speech and music considering its bandiwdth usage. G729 is the most efficient in terms of bandwidth, however is not open source and requires a licence fee.