Hello everyone. I have a question. So, of course I use Vmix 18 and now would like to study in his live interview over the phone. So that when I have a live broadcast that we can contact people by phone live. How is it best to do. Divider to some awards and so on, I have to realize the live contact with your viewers. What do you do and what it takes to do it. And what is the condition for good.

I pay $3 a month and Skype lets you dial cell/landline phones.

Hi

As a follow up, with phone calls -- and skype video etc, and playing it all out on YouTube live (for example) how do we deal with latency.

I've seen a YouTube live show delay as much as 15 minutes.

Or is this a non-issue, as long as the caller is told to turn down his/her computer audio???? Like the instructions you hear in some radio talk shows????

Rowby

I do this every week for a hangout/podcast that I co-produce (http://www.vuc.me.) We're a bit unusual in that we're all telecom geeks of one sort or another.

While I like some telephone hybrids, they get costly quick and usually deliver marginal audio quality. I prefer to use something that lets me take advantage of HDVoice where I can, and without using too much hardware.

HDVoice delivers an audio channel that's 50Hz - 7KHz. It sounds dramatically better than a typical phone call. To achieve HDVoice you must have control of both ends of the call and avoid the PSTN altogether. It's a long topic that I've covered on my blog over the years. (http://www.mgraves.org)

The short story is that you can use a SIP soft phone (Bria, but there are many) and a software mixer (VoiceMeeter Banana) to create a soft hybrid. That's what I do every week.

We interconnect a Hangout to the ZipDX conference server. In this way we can have many people connected via telephone; some in HDVoice via SIP, others via traditional dial-in number.

We actually have a sponsor (Voxbone.com) that provides us with dial-in numbers in 50+ countries. We have participants in the US, Canada, UK, France, Germany, Switzerland, Austria, Hungary, even Tunisia now and again.

Skype is ok, but the quality delivered varies a lot these days.

I'm more a fan of leveraging free, WebRTC-based services that anyone can access using Chrome. They deliver great audio quality using the Opus codec, and there's nothing to install. Just distribute a URL for the "room." Jitsi Video Bridge is a favorite.