Your new station will at first use the same settings as the currently loaded station.
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.
@@ 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.
@@ RTP Multicast Source : LUCI can be a source for multiple Multicast listeners.
@@ RTP Multicast Receiver : LUCI can be set as a Multicast listener.
@@ 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.
Select here one of the following codec formats for the outgoing stream to the studio:
@@ AAC(-LC), AAC-ELD, AAC-ELDv2, AAC-LD, AAC-HE or AAC-HEv2
@@ G.711 A-Law or G711 u-Law
@@ L16 or L24
@@ Opus Audio, Opus Low Delay or Opus Voice
@@@ ULCC, ULCC-24 or ULCC-S
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|
From here you will be prompted to ‘Credentials’ to fill in your login credentials.
From here you will be prompted to ‘Audio’ to fill in the audio stream settings.
For protocol RTP
Use an IP-address or URL,
examples: 18.104.22.168 or echo.lucilive.com
For Protocol SIP *
Fill in the destination number here.
For Protocol Shoutcast or Icecast
Use an IP-address or URL.
examples: 22.214.171.124 or shoutcast.lucilive.com
Fill in the port number you want to use.
Only valid for the SIP protocol, when needed.
Use an IP-address or URL
examples: 126.96.36.199 or stun.iptel.org
Switch on if you want to use the credentials of the default user that you have set in ‘Settings > Defaults’
Your username is the name you use to login at your server. Fill in the username and SIP server like these examples: firstname.lastname@example.org or email@example.com.
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.
Default on Mono. Select Stereo if you want to broadcast in Stereo, and the chosen codec ( ‘Stations > New station > Streaming’) supports it.
Default value is 16bit. Supported bit depths: 8bit, 16bit or 24bit.
Default Off. If Activated you will automatically record your outgoing stream when you are connected live with your station.
Default value 48KHz. Set Sample-rate of the codec format you selected for the outgoing stream.