Connect LUCI using SIP

SIP (Session Initiation Protocol) is a protocol used in VoIP communications allowing users to make voice calls. Unlike the RTP protocol you need an external SIP server (like where both the receiver and transmitter can login (in SIP terms “REGISTER”) with a username (can also be a number) and password. Both devices login to the SIP server and then call each other by “dialing” the other username. The SIP server is only used to connect, the audio data stream is directly send from SIP client to SIP client.

This small How-to explains a step by step procedure to setup LUCI STUDIO or LUCI LIVE as a SIP client. That is to be able to make or receive a SIP call.

First you need a SIP account at a third party SIP server service like or On registering you will have the following account details:

  • a username:  johndoo or
  • a password: jd2019
  • SIP server domain:
  • Optional STUN server:

These account credentials are used to register your LUCI to the SIP server. Only when registered you can wait for a SIP call or make a SIP call to any other SIP client.

Method 1: Direct Station Call

To be able to make or wait-for a call you must create a Station in LUCI by selecting the Add/Edit new Station function. You will see the Station profile editing screen.

Here you enter :
name : Station name or person you call with this station
protocol: SIP
destination: the sip account username, optional with the @ sip server domain name or IP address if it differs from your account server.
stun: the optional stun server
user: your SIP account username with @ sip server domain name or IP address
password: your SIP account password

When this station is selected in LUCI and you press the register button , LUCI will be registered at with the johndoo user credentials. On success the button will stay red without any error message. As it is registered, it is now ready to receive SIP calls from any other SIP client.

To make the call to Jane your press the MIC button , and wait for Jane to answer your call. You can hangup by pressing the MIC button again.

Method 2: using the SIP user default settings

If you want to make more stations with other SIP destinations it is preferable to make use of the SIP default settings. Here you can put in your account credentials once, making the Station profiles simpler. You can set your default credentials in the Options -> SIP section.

Here you enter :
user: your SIP account username without the server domain
password: your SIP account password
server: sip server domain name or IP address
displayname: this name will show up on the callee’s display so they know it’s you.

All Stations can now be set with only the callee username as destination and a selected Default option.


Advanced SIP settings

In the Options SIP section you can change some advanced SIP features :

Use extra identities for incoming calls : If you have more accounts on other SIP servers and you want to be able to receive calls on these accounts also, then you can add more SIP user credentials here :

name: just an identifier
user and password: your SIP account credentials for this sip server
server: the SIP server domain name.
New : press to get an empty form to fill in
Remove: press to remove the current entry

On pressing the register button, LUCI is registered with all set SIP accounts on their SIP servers and can therefore be called by any clients of these servers. Note: making a call is done through the main default SIP server.

Use IPv6 when available : IPv6 is the new internet addressing scheme but not available on every network. If your confident that your internet path to the SIP server is IPv6 capable you can select this option.

Auto Answer : When your LUCI is used to receive calls and wants to connect automatically upon being called, select this option.

rtp port : The SIP server is only used to make the call connection, once connected the audio data is send through a different port defined here. Normally port 5004 is used, but if your internet provider blocks this port or for safety reasons you want to use a different data port, change it here.

Framing timeout : If for some reason no audio data is received you can set a timeout here to break of the call.

Different SIP server port

The standard communications port for SIP is 5060. If for some reason your SIP Server provider uses or prefers a different port you can change it by adding the port number to the server domain name, separated by a colon ‘:’ sign , so as Stations destination or at the SIP default server

Wait for call Station

When your LUCI needs to be in a constant  waiting to be called mode, it is a good practise to define a special Station for this purpose:

As destination you can either fill in


or if you also want to automatically answer incomming calls:



LUCI STUDIO has the ability to answer a call with the same codec format of the callee instead of a fixed set codec. For this select “Follow caller” as format.

Note: waitforcall cannot be used in LUCI LIVE SE and LUCI LIVE Lite as they have only one Connect button. They can make a phone call, but cannot wait to receive one.

Known SIP problems/difficulties

  • There are some mobile internet providers that block SIP traffic on their cheaper subscriptions. The only solution is to change the subscription or provider.
  • Some combinations of routers and firewalls generate NAT (Network address translation) faults, causing the SIP Server to negotiating a wrong audio data port. This is noticeable by having no audio signal at either the caller or the callee. Using a stun server can help but not in all cases. For more info :
  • Some SIP client apps support only one or a few audio codecs like G722  or AAC-HE or OPUS, and not all bitrates or sample frequencies.
  • As SIP uses a third party “in between” Server your SIP connections can be monitored.
  • Check your office phone equipment. If this is a Voip Server changes are it will eatup all SIP communication on your network, leaving your LUCI in the dark. A common solution is to get a SIP account on this Voip server.
  • More and more big radio stations implement their own SIP server systems to have full control and monitoring their Codec park. Benefits are avoiding NAT issues, block unknown origins, monitor calls, activly switch to free codecs, generate and distribute phonebooks etc.