With or without WebRTC, Opus seems like a winner.WebRTC post series or just read what WebRTC is all about.]
Opus is a relatively new codec. Less than 2 years in existence. If you want to compare it to any other codec, then Speex is probably the best one:
- Both are considered open source codecs, distributed under the BSD license
- Both support narrow and wideband
The difference? Speex raised a white flag, as stated on its official website:
The Speex codec has been obsoleted by Opus. It will continue to be available, but since Opus is better than Speex in all aspects, users are encouraged to switch
Opus had a great potential when it started. It shows merit and superior technical capabilities compared to commercial, royalty based codecs as well:
But it was still debatable if this will be enough for it.
That said, recent events indicate that Opus is being adopted widely and not only within the open source community. Here are two such instances.
UberConference adopted Opus as part of their voice conferencing service just last week. While it may seem like an easy feat, you should remember that most media servers today still can’t deal with Opus, so having an audio mixer capable of dealing with this codec required an extra effort on top of the vinyl distribution coming from the web browsers.
The other story is far more interesting. It is the one about Audio Codes and their recent developments around IP phones. In order to support WebRTC, they added… Opus to IP phones. Not WebRTC, mind you, just the voice codec. The idea behind it is to make it easier to integrate WebRTC with existing VoIP deployments while maintaining HD voice capabilities – and doing that without the need to transcode between codec versions. I believe this is a very smart move on their part and not an obvious one.
Do you know of adoption of the Opus codec in non-WebRTC environments? Speak up – I want to hear about these stories to see how far this push of WebRTC and its technologies is affecting different regions of the VoIP value chain.