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.
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.
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. As LUCI Live Lite has only 2 codec formats that both have only 1 bitrate option, the value shown is only informative and can not be set.
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
|G.722 mono > 64 kbps||ULCC mono > 252 kbps|
From here you will be prompted to ‘Credentials’ to fill in your login credentials.
The connection details of the server/codec:
For protocol RTP
Use an IP-address or URL,
examples: 188.8.131.52 or echo.lucilive.com
For Protocol SIP *
Use CODEC@SIPSERVER or CODEC (when there is no SIP server) where SIPSERVER is the URL or IP-address of the SIP server.
examples: firstname.lastname@example.org or email@example.com or 184.108.40.206
Fill in the port number you want to use.
Only valid for the SIP protocol, when needed.
Use an IP-address or URL
examples: 220.127.116.11 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. Only fill in username if you want to log in to a SIP.
Your password is your personal log on credential
belonging to your username. Passwords are used only together with usernames for logging in to SIP.