LUCI Knowledge Base

Add / Edit station

By choosing Add / Edit station you will see the dialog for setting up a new station, or changing the selected station.

Your new station will at first use the same settings as the currently loaded station.

name : Fill in the name of your station. Always start by giving the new station a name, otherwise it will not be saved. In this case we will name it Studio 1 and alter the settings.

protocol : The following supported protocols can be set:

RTP : send and receive a symmetric RTP stream with integrated signalling.
SIP : You can use SIP in combination with formats like G711, G722 to connect to standard VOIP equipment. Or use other codecs like AAC-HE or MP2 to connect to other SIP compliant codecs.

destination : The connection details of the server/studio codec.

For protocol RTP
Use an IP-address or URL, combined with a : and the IP port number,
examples: or
For Protocol SIP *
Use CODEC@SIPSERVER or CODEC (when there is no SIPserver) where SIPSERVER is the URL or IP-address of the SIPserver.
examples: or 102409@ or

stun : Only valid for the SIP protocol, when needed.
Use an IP-address or URL.
examples: or

default : Only valid for the SIP protocol, when needed. Switch on if you want to use the credentials of the default user that you have set in ‘Options > SIP > Defaults’

user : Your username is the name you use to login at your server. Only fill in username if you want to log in to a SIP, Shoutcast or Icecast server.

password : Your password is your personal log on credential belonging to your username. Passwords are used only together with usernames for logging in to SIP, Shoutcast or Icecast servers.

jitter buffer : Default value 100 ms (WiFi) / 200 ms (3G). Supported buffer length WiFi and 3G: 10 ms to 500 ms. This is the jitter buffer that LUCI uses for the RETURN stream. This will NOT have any influence on the outgoing stream to the studio.

jitter buffer Dyn. : This is a setting that you can choose so the software determines the buffer that is needed to get a drop free connection automatically while you are streaming. It takes the normal jitter buffer setting ( say 50 ms ) as the lowest possible, and the Jitter Buffer Dyn. setting ( say 200 ms ) as the possible range ( so buffer automatically set is between 50 and 250 msec ). 0 ms will set this feature off.

format : Select here one of the following audio codec formats for the outgoing stream to the studio:

stereo : Default is Mono. Only Android devices that *support the latest audio driver AAudio and have 2 internal microphones, stream in real stereo, all other will stream the mono input on both channels.

bitrate : Default value is set on 64 kbps. Set the bitrate of the codec format you select.
For the outgoing stream.
Supported bitrates depend on the chosen codec.

Samplerate : Default value 48KHz. Set Sample-rate of the codec format you selected for the outgoing stream.

Talk Back mode ( Push to Talk ) : Kind of Walkie-Talkie mode to be able to only listen, and only talkback by pressing a button. Similar to a Talk button on a studio mixing console. For live studio music mix / voice-over producers reviews or other applications were you do not want to use headphones.

Overview for the supported codecs and bitrates

Samplerate : Default value 48KHz. Set Sample-rate of the codec format you selected for the outgoing stream.

Overview for the supported codecs and bitrates
G.722 mono > 64 kbps ULCC mono > 252 kbps

*Note: AAudio support is only available on devices with Android 9 ( PIE ) and later.

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