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Opus is a versatile and powerful audio codec that offers several advantages for VoIP (Voice over Internet Protocol) applications. However, it also has some limitations. Here's a breakdown of the advantages and disadvantages of the Opus codec and why it's commonly used for VoIP:


High Quality: Opus is known for its high audio quality, offering better sound fidelity compared to many other codecs, even at low bitrates. This high quality makes it suitable for transmitting voice over the internet without significant loss of audio clarity.

Low Latency: Opus has low encoding and decoding latency, which means there is minimal delay in transmitting and receiving audio packets. This low latency is crucial for real-time communication applications like VoIP, as it helps reduce communication delays and improves the overall user experience.

Adaptive Bitrate: Opus supports adaptive bitrate encoding, allowing it to dynamically adjust the bitrate based on network conditions and available bandwidth. This adaptive bitrate feature ensures optimal audio quality even in variable network conditions, such as fluctuations in internet speed or packet loss.

Wide Range of Bitrates: Opus supports a wide range of bitrates, from very low bitrates suitable for narrowband connections to high bitrates for high-fidelity audio. This flexibility allows it to adapt to various network conditions and user requirements, making it suitable for a wide range of VoIP applications.

Open Standard: Opus is an open and royalty-free audio codec standard developed by the Internet Engineering Task Force (IETF). Its open nature encourages widespread adoption and interoperability among different VoIP platforms and devices.


Compatibility: While Opus is widely supported across many VoIP platforms and software applications, there may still be some compatibility issues with older or less commonly used VoIP systems that do not support Opus natively. In such cases, transcoding may be required to convert Opus audio streams to a compatible format.

Bandwidth Requirements: Although Opus offers excellent audio quality at low bitrates, higher-quality settings may require more bandwidth, which can be a limitation for users with limited internet connections or in bandwidth-constrained environments.

Despite these limitations, the advantages of Opus generally outweigh the disadvantages, making it a popular choice for VoIP applications. Its high audio quality, low latency, adaptive bitrate support, and open-standard nature make it well-suited for delivering clear and reliable voice communication over the internet.


To enable OPUS, you will need to navigate to Settings->Tenants->edit tenant in question, and add the OPUS codec for the local and remote codec section, as per the screenshot below,

Once you enable OPUS in this section, you will need to navigate to the tenant level, and edit the extension in question on which you wish to enable OPUS as per the below screenshot,

By default, OPUS will be the prioritized codec in our mobile apps, while on your desk phone devices, you will be required to make it a priority and enable it.

You can also enable Adaptive Bit Rate for OPUS on the following path, Settings->Protocols->RTP tab, as per the screenshot below,