Opus for Asterisk

By Malcolm Davenport
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The past two years, we’ve heard from users about how great it’d be to officially integrate the Opus codec with Asterisk. Opus is the codec that’s used by millions of web browsers for WebRTC calls. It’s also known because it’s in a handful of VoIP phones, including Digium’s new D6x IP phones. Opus is better than a lot of other codecs because it provides great quality audio, even under very poor network conditions.

Until now, we’ve held off providing anything for Opus in Asterisk because of concerns about some intellectual property disclosures made against the codec. But today, we’re happy to announce that we’ve been able to resolve our concerns. So, along with Monday’s release of Asterisk 14.0, today, we’re putting out a new version, 14.0.1, that can finally and officially support Opus.

That’s the great news. The “sort of okay” news is that in order to resolve the legal concerns, we have to distribute Opus as a free binary. We can’t distribute it as source. And, once a day, it has to anonymously report back to Digium how many channels of Opus have been used. This way, we’re avoiding unlimited liability. The “slightly better” news is that it’s not like Digium’s other binary modules. It doesn’t require activation keys or registration. It’s not tied to running on a single machine. And, it’s even something you’ll be able to install directly when you build Asterisk, so you won’t have to download it from a website later.

Opus for Asterisk, it’s available now in Asterisk 14.0.1, and we’ll have a version of it available for Asterisk 13 in a couple of weeks, so that everyone using the current LTS release can take advantage, too.

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About the Author

Malcolm Davenport

Digium lifer, celebrator of 16 Digium birthdays, and Digium employee #4. "I like te-lephony and I cannot lie. You other vendors can't deny; When a call comes in with MOS so you can't hear and some echo in your ear you get angry!" - Sir Mix-a-Malcolm

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