WebRTC Live #42 – “Asterisk, WebRTC, and DialogFlow,” Dan Jenkins, Nimble Ape

On April 22nd, WebRTC.ventures produced Episode #42 of WebRTC Live! Formerly known as WebRTC Standards, WebRTC Live is a webinar series about the latest use cases and technical updates to the popular coding standard for live video.

For this episode, we were joined by guest Dan Jenkins, founder of Nimble Ape. Dan discusses WebRTC and Asterisk, as well as how to use Asterisk as a connector into speech-to-text services and DialogFlow.

Check out Episode #42!

WebRTC Live #42 – “Asterisk, WebRTC, and DialogFlow,” Dan Jenkins, Nimble Ape

The next episode of WebRTC Live will premiere on BigMarker on Wednesday, May 6th, 2020 at 12:00pm Eastern Time (US). Tim Panton, CTO of |pipe|, will join us to discuss the magical things that you can do with WebRTC when you apply it to niche problems. Click here to register for the webinar!

Never miss an episode of WebRTC Live! Follow us on BigMarker (our new WebRTC Live broadcasting platform) and watch past episodes on YouTube and on our blog. Using the form in the right sidebar, join our mailing list to be the first to know about upcoming WebRTC Live episodes and the latest news in WebRTC!

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