Why libopus is Preferred Over Speex for VoIP

Modern Voice over IP (VoIP) applications require high-quality audio, low latency, and adaptability to varying network conditions. While the Speex codec was once the industry standard for voice compression, it has been officially deprecated and superseded by libopus (the Opus codec). This article explains why libopus is the superior choice for modern VoIP, detailing its advantages in audio quality, latency, adaptability, and technological integration.

Superior Audio Quality and Bandwidth Range

Speex was designed primarily for human speech, supporting narrowband (8 kHz), wideband (16 kHz), and ultra-wideband (32 kHz) audio. While effective for voice, it performs poorly with music or background sounds.

In contrast, libopus is a highly versatile codec that supports a massive range of audio bandwidths, from narrowband (8 kHz) all the way to fullband (48 kHz) stereo. It utilizes the SILK codec (developed by Skype) for high-efficiency speech compression and the CELT codec (developed by Xiph.Org) for ultra-low latency audio and music. This dual-engine architecture allows Opus to deliver crystal-clear voice and high-fidelity music within the same stream, outperforming Speex at every comparable bitrate.

Unmatched Latency Performance

Low latency is critical for natural, real-time conversation over VoIP. Speex has a fixed frame size, typically resulting in an algorithmic delay of 30 milliseconds or more.

Libopus is built from the ground up for interactive real-time communication. It supports frame sizes ranging from 2.5 milliseconds to 60 milliseconds. This flexibility allows VoIP applications to minimize delay to imperceptible levels on high-speed connections, or increase frame sizes to reduce packet overhead on congested networks.

Dynamic Adaptability to Network Conditions

Internet connections are inherently unstable, often suffering from packet loss, jitter, and fluctuating bandwidth. Speex has limited ability to adapt to these changes on the fly.

Libopus excels at real-time adaptation. It can dynamically adjust its bitrate (from 6 kbps to 510 kbps), audio bandwidth, and frame size without causing audio artifacts or requiring a session restart. Additionally, Opus features highly advanced Forward Error Correction (FEC) and packet loss concealment (PLC), allowing it to maintain clear, uninterrupted audio even when experiencing packet loss rates as high as 30%.

Obsolescence and Industry Standardization

The most definitive reason to prefer libopus over Speex is that Speex is obsolete. Jean-Marc Valin, the primary creator of Speex, also co-created Opus. Upon the standardization of Opus by the Internet Engineering Task Force (IETF) as RFC 6716, the Speex project was officially deprecated.

Today, libopus is the default audio codec for WebRTC, the technology powering modern web-based communication tools like Zoom, Discord, Google Meet, and Microsoft Teams. Using Speex in a modern application limits compatibility, whereas libopus ensures seamless integration with contemporary communication standards.