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Audio encoding

In this article

Audio encoding, also known as voice encoding or codec (compression/decompression), is the process of converting an analog audio signal (your voice) into a digital format that can be transmitted over an IP network, such as the internet. Similarly, at the receiving end, the digital audio data is decoded back into an analog signal for playback through speakers or headphones. Audio encoding is a crucial component of VoIP technology, as it affects the quality of voice calls and the efficiency of data transmission.

When it comes to SIP (Session Initiation Protocol) for VoIP (Voice over Internet Protocol) communications, several audio codecs (compression/decompression algorithms) are commonly used and widely compatible across various SIP devices and platforms. These codecs ensure interoperability and facilitate efficient voice communication over IP networks. Here are some of the most commonly supported audio codecs in SIP:

G.711 (u-law and a-law)

G.711 is one of the most widely supported codecs in SIP. It offers high-quality, uncompressed audio but consumes a relatively large amount of bandwidth (typically 64 kbps per voice channel).

u-law is commonly used in North America, while a-law is prevalent in Europe and other regions.


G.729 is a popular codec known for its good voice quality while being more bandwidth-efficient compared to G.711. It typically consumes 8 kbps per voice channel.

G.729 is often favored in situations where bandwidth is limited.


Opus is a versatile and adaptive codec that offers excellent voice quality across a wide range of bitrates. It can adapt to varying network conditions, making it suitable for a variety of SIP implementations.

Opus is increasingly supported in modern SIP applications.


G.723.1 is another codec that offers good voice quality while being bandwidth-efficient. It typically consumes 5.3 kbps per voice channel.

It’s suitable for scenarios with limited bandwidth.


G.726 provides several bit rate options (16 kbps, 24 kbps, 32 kbps, and 40 kbps) to balance voice quality and bandwidth consumption.

It’s commonly supported in SIP systems.

iLBC (Internet Low Bit Rate Codec)

iLBC is designed for VoIP and videoconferencing applications, offering good voice quality with moderate bandwidth requirements.

It’s often used in SIP-based communication platforms.

AMR (Adaptive Multi-Rate)

AMR is commonly used in mobile VoIP applications and supports various bitrates, adapting to different network conditions.

It’s compatible with many SIP clients on mobile devices.


Speex is an open-source codec optimized for low-bitrate voice encoding. It’s suitable for applications where bandwidth conservation is critical.

Speex support can be found in some SIP implementations.

The choice of codec in SIP communications often depends on factors like available bandwidth, desired call quality, and compatibility with SIP equipment and services. Many SIP devices and platforms offer support for multiple codecs, allowing for negotiation between endpoints to find the best codec based on network conditions and quality requirements.

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