Setting Up A Phone In Radio Station - Airtime Support Discussions on Sourcefabric Forum

Ed Pressman · Post edited by Ed Pressman at 2013-05-07 06:41:47

Hello -

My first post here after having spent some time reading the Airtime Manual and posts on the forum. I am in the preliminary stages of setting up an online radio station - a field in which I am almost totally ignorant. Airtime looks as though it will do much of what is required.

I currently have available an Icecast streaming server and have Aitime and Mixxx on my list of applications needed to do the job well. However I am unsure if this arrangement will work. Following is a shortlist of the main functionality I need to achieve. Any comments, suggestions and pointers on this will be of great value. I'd like to get the roadmap reasonably accurate before I set out on the journey.

The station will focus on "live" shows with anywhere from 1 up to 3 or 4 "hosts". When there is more than 1 host it is likely that the hosts will be spread around the globe. Skype or another VOIP service would likely be the preferred method of connecting multiple hosts into the show.
Hosts will have pre-defined "shows" and timeslots. When 1 host hands over to the next host, I want them to be able to talk to one another live on air as part of the "handover" process and to create continuity.
Typical format for a show would be 2 hours where the first hour has a guest being interviewed by the host (or hosts) and then in the second hour we would take calls in from listeners which would go out live on air.
We would/could provide a Skype ID for listeners to call in on. We would also provide multiple call in telephone numbers (ie a US number, a UK number and Canadian number etc etc) and receive those calls in on a laptop etc running a softphone app.
Each primary show host would need a decent mic and a simple mixer to take in the calls. An example of how this could be done is shown here: http://www.youtube.com/watch?v=6JHuf2TKKJg Is this workable with Airtime and if so, how would Mixxx be used to handle the input from the physical mixer?
I have Mixxx down as it runs on Windows, Mac and Linux. So my thought was that each host would install Mixxx on their local pc and then use that to connect to their Airtime slot, with Airtime running on the streaming Icecast server. Basically I need to provide a host with a basic blueprint of the kit they need to make it work.
I will need detailed listener stats and the means to run scheduled ad breaks during live programmes - hosts will therefore need some kind of alert (could be ad break music fading in slowly for example) so that they can move from the interview to the ad break in a controlled manner.
I also want to send the station to tunein.com so as to get listed/featured on their mobile app player etc.

Simply put, I want to set up a full blown live chat radio station where phoners can call in with emphasis on good sound quality for the hosts.

Can I do this with a combination of Icecast, Airtime, Mixxx, Skype and VOIP? Or do I need to use BUTT or some other tools to better handle the running of the station?

--Ed

Helmut Müller

In Roadmap Version 2.5.0 there should be a new feature
[CC-2616] - Live shows with voice using Twilio or user-to-user communication using Twilio

Maybe this would be what you want.

Ed Pressman

I see - so at the moment, using Airtime and Mixxx will not give me the solution I am looking for.

Is it the idea that version 2.5.0 will be the only app I would need to get all this handled? And if so, doea anyone have any idea of when version 2.5.0 is likely to be released?

Thanks Helmut.

--Ed

Lisa Forbes

I use Butt and Skype. I'm not that familiar with Mixx but all you really need from it is an encoder to stream to Airtime. That part is easy what's not so easy is setting up Skype with the mix minus to send both sides of the conversation but with a little time and a LOT of YouTube tutorials it can be done.

Ed Pressman

Hello Lisa -

Thank you for the positive pointer. So a shortlist of items required would be:

Streaming server using Icecast
Airtime installed on streaming server
BUTT
Skype
Mixing desk (small Behringer or Mackie mixer etc)
2 PC's - 1 running skype on which to receive incoming calls, the other running BUTT
Quality Mic and popguard
Calls come into PC1 running skype - output goes to mixer - mic goes to mixer - mixer output goes to PC2 - PC2 uploads audio using BUTT to Icecast server running Airtime

Is this close ?

And the YouTube video tutorials you referred to, would you have any specific links that would be worth watching?

Again, thanks for your response on this.

-Ed

hoerich

add for 6. and 8.

If you use 2 soundcards and have knowledge to set it up correctly, you only need one PC.
With BUTT you can use any SoundDevice available. Not only the default one.
f.e. with edcast you can only use the SoundDevice currently set as default. at least in windows.

greetz

Helmut Müller

Just to let you know. I use Teamspeak and SAM Broadcaster for this Scenario. There are surely much more and better solutions.

Ed Pressman

This video gives a good summary of the type of setup I'm looking to create.

http://www.youtube.com/watch?feature=player_detailpage&v=Z8dNQq-b2k0

Thanks for the pointers everyone.

--Ed

Lisa Forbes

Hi Ed,

Sorry I just noticed your question. Hoerich is right if you're using a mixer and you have 2 soundcards then you only need 1 PC to run Skype.

And yes those are the things you would need for hardware minus 1 PC and plus 1 soundcard.

I also use a compressor/limiter/gate to cut out some of the background noise. The most helpful tutorial I found for setting this up was Cliff Ravenscraft's (Podcast Answerman) video .

Ed Pressman

Hi LIsa -

Thx for confirming the basic list. I stumbled across Cliff Ravenscraft's site a few weeks back and it is useful in getting a handle on the equipment needed to produce a pro sound.

I am planning on having multiple hosts who by default will be in multiple locations. The idea of them all having a mixer and a high quality mic is appealing but not financially viable in the early stages. I've also been looking into the use of broadcast quality headphones with mics. The Audio-Technica BHPS-1 (see HERE ) looks good. The sound quality will not be at the level of say an Electrovoice RE20 but from a cost and ease of use point of view it could be a good option.

Ed

Lisa Forbes

I dont really see a need for broadcast quality headphones. Unless you are a musician and producing music its just not needed for what we do. Now as far as multiple hosts because of finances my hosts call in via skype by phone or skype to skype. Audio quality is not that bad but skype can itself be iffy. The ideal when possible is for each host to have similar setups so quality is consistant. It also allows for them to be able to set their own audio levels and "tune" their mics to enhance their voices. I have decided that I will let anyone interested in joining my team know up front the minimum financial investment required and give them a time frame in which they must have a suitable set up or be dropped.

Lisa Forbes

Sorry Im on my phone and didn't see your link to know you were referring to a headset and not just headphones so please disregard that part.

Ashworth Payne

ed i've been thinking about your problem and i believe i have a solution. its a bit complex, but should do exactly what you want, and not-so-expensive to implement.

quick rant: the video you posted is one of many slick tutorials that fail to mention a key problem: head-phone out should not be plugged into line-in unless a dubbing cable is used.

also, since you're working across multiple pieces of software, you might consider doing i/o with behringer uca222's. they're cheap. and,, they output line-level... perfect for going into/outof mixers. and, unlike expensive audio interfaces with multiple i/o, you aren't limited by asio drivers (with asio devices you get multiple i/o on one device, but only when apps are specifically written to use asio... and windows desktop is not one of those apps. nor is skype)

i'd setup a desktop pc as your command-central. all voice apps would run on it. each app would get its own uca222 for i/o. on this pc i'd run skype for callers into one channel on the mixer. i'd run teamspeak for my hosts into another channel. and i'd run airtime on yet another channel... airtime would merely play-out jingles etc through its sound card (uca222 again ).

teamspeak and skype would make good use of aux-send or aux busses. this will allow you to bridge the audio between the two applications. off the top of my head, you'll need at least two aux's on your mixer. maybe three? ugh wheres my pen and pad?

this setup would allow your hosts to connect using teamspeak (the connection is always available). they could also log into airtime to see exactly when a jingle is to be played. also, you can put multiple rooms in teamspeak, so the hosts could jump to another room during breaks to talk without going out over the stream.

you'll need to have someone at the mixer in case a skype caller needs to be kicked or some piece of hardware dies.

lastly, you'd need to encode everything coming from the mixer and send it wherever its going... a streamhost somewhere.

if you need assistance feel free to contact me
studio at backroomradio <- and thats a dawt calm.

and please, folks, don't connect speaker-level outputs to line-level inputs. :)

Ruth Smithy

Hi Ed

I hope you can see this message.

I am trying to do exactly what you are trying to do. If you were successful, can you please contact me? I need urgent help with the exact situation as you described.

Thank you

Ruth

Stephen

We have telephone numbers and skype users dial directly into live talk shows. check www.a2zen.fm This was developed 18 months ago and has been working great thanks to the sourcefabric guys!!!

We used DID/SKYPE ---> FREESWITCH --> AIRTIME 1.9.4 --> ICECAST

Mediacore is used to archive all the shows for RSS playback.

It all installs on one server. A custom SwitchBoard to handle a greenroom, screening users, prerecords, and mute/kick was developed with chatroom.

We are currently upgrading to Hi-DEF voice and will be supporting WEB-RTC so user can join via browser using opus 48k. I'm looking at 2.4 airtime now. (You guys have done great work) if their is community interest i could add a live telephony module. I'll look at the twilio enhancement request.

Regards,

Bill Burton

Hello,

This sounds very interesting. For live programming, I run a relay into
Airtime via edcast and bridge FreeConferenceCallHD.com using Skype. This
service can also connect with FreeSwitch but I haven't tried it.

Many people call in just to listen and during a Q&A, callers can ask
questions.

At certain other times, audio is relayed the other way from Airtime/Icecast
to the phone conference for listening only as some don't have computers or
smart phones.

Bill
On Jun 27, 2013 8:04 AM, "Stephen"
wrote:

> We have telephone numbers and skype users dial directly into live talk
> shows. check www.a2zen.fm This was developed 18 months ago and has been
> working great thanks to the sourcefabric guys!!!
>
> We used DID/SKYPE ---> FREESWITCH --> AIRTIME 1.9.4

Ed Pressman · Post edited by Ed Pressman at 2014-12-03 07:44:32

Hi -

After much time looking for a good solution I have found it in using skype/voip, BUTT and Voicemeeter.

If you do live discussion talk shows and want to take calls
in from your listeners (skype, voip etc) then Voicemeeter, from
VB-Audio, (donateware) is an excellent working solution. More info along
with links to some excellent video tutorials can be found on their
facebook page here:

https://www.facebook.com/vbaudiosoftware

I
use it in conjunction with BUTT, running 2 skype instances
simultaneously with the guest on one channel and another channel for
callers in.

Means you can run the whole thing, including system sounds intro outro music auido clips etc from just 1 decent PC.

Ed

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