Custom station > New Station

stations_menuBy choosing “Custom station” you will see the screen for setting up a new station.stations_addnewstation

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

> Title

stations_newstation_titleUntitled station
From here you will be prompted to ‘Title’.

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.

Streaming

protocol
The following supported protocols can be set:
RTP: send and receive a symmetric RTP stream with integrated signalling.
SC-RTP: send and receive a symmetric RTP stream with integrated signalling. Send the same stream on all available active network-interfaces. So when you have a 3G and a WIFI connection, LUCI will send 2 streams. This way, if 1 connection breaks up, the studio will still receive the other. In addition to this, LUCI will send also streams via IPV4 and IPV6 if the network-interface and the studio support it.
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.
Shoutcast: LUCI can be a source for a Shoutcast internet radio station you operate.
Icecast: LUCI can be a source for an Icecast internet radio station you operate.

streaming_codeccodec
Select here one of the following codec formats for the outgoing stream to the studio:
AAC, AAC-ELD, AAC-LD, AAC-HE or AAC-HE v2
G.711 A-Law or G711 u-Law
G.722
L16 or L24
MP2
Opus Audio, Opus Low Delay or Opus Voice
ULCC, ULCC-24 or ULCC-S

streaming_bitratebitrate
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.

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.

Overview for the supported codecs and bitrates
AAC mono > 56 – 256 kbps L24 mono > 1280 kbps
AAC stereo > 96 – 384 kbps L24 stereo > 2116 kbps
AAC-ELD mono > 18 – 64 kbps MP2 mono > 40 – 192 kbps
AAC-ELD stereo > 32 – 128 kbps MP2 stereo > 112 – 256 kbps
AAC-HE mono and stereo> 12 – 64 kbps Opus Audio mono > 18 – 192 kbps
AAC-HE v2 stereo > 18 – 64 kbps Opus Audio stereo > 64 – 384 kbps
AAC-LD mono > 50 – 192 kbps Opus Low Delay mono > 18 – 64 kbps
AAC-LD stereo > 76 – 384 kbps Opus Voice mono > 18 – 64 kbps
G.711 A-Law mono > 64 kbps ULCC mono > 252 kbps
G.711 u-Law mono > 64 kbps ULCC stereo > 492 kbps
G.722 mono > 64 kbps ULCC-24 mono > 276 kbps
L16 mono > 705 kbps ULCC-24 stereo > 540 kbps
L16 stereo > 1058 kbps ULCC-s mono > 51 kbps

destination
From here you will be prompted to ‘Host’ for filling in the IP address of your station, the Port number and the Stun (use only for SIP protocol if needed).

username
From here you will be prompted to ‘Credentials’ to fill in your login credentials.

channels
From here you will be prompted to ‘Audio’ to fill in the audio stream settings.

Host

destination
The connection details of the server/codec:
For protocol RTP
Use an IP-address or URL,
examples: 83.163.21.56 or echo.lucilive.com
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: echo@iptel.org or 102409@167.45.7.234 or 102.78.90.45
For Protocol Shoutcast or Icecast
Use an IP-address or URL.
examples: 83.163.21.56 or shoutcast.lucilive.com
port
Fill in the port number you want to use.
stun
Only valid for the SIP protocol, when needed.
Use an IP-address or URL
examples: 83.163.21.56 or stun.iptel.org

Credentials

default user
Switch on if you want to use the credentials of the default user that you have set in ‘Settings > Defaults’
username
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.

Audio

channels
Default on Mono. Select Stereo if you want to broadcast in Stereo, and the chosen codec ( ‘Stations > New station > Streaming’) supports it.
bit depth
Default value is 16bit. Supported bit depths: 8bit, 16bit or 24bit.
auto record
Default Off. If Activated you will automatically record your outgoing stream when you are connected live with your station.
sample rate
Default value 48KHz. Set Sample-rate of the codec format you selected for the outgoing stream.