the documentation is updated 03.4.2017
Ready interfaces for i3 pro to work as SIP intercom / intercom panel:
| intercom (smartphone)
intercom (tablet, PC)
| simplified intercom (smartphone)
simplified intercom (tablet, PC)
| intercom panel (smartphone)
intercom panel (tablet, PC)
| Mobotix intercom panel (smartphone)
Mobotix intercom panel (tablet, PC)
- use it for free in test mode of i3 pro (via iRidium Transfer)
- purchase any iRidium 3.x license, the "SIP Telephony" addon will be available there for free
iRidium for SIP-telephony is a set of tools for creating interfaces of call panels working by the SIP protocol. iRidium can make calls and get audio- and video-calls from other call panels. SIP driver doesn't work on iRidium Server.
To make calls you are required to have a set up a SIP server (it is also called a PBX server or IP-АТС) commutating calls between call panels. iRidium does not supply any SIP server so you need to use some ready solution, for example:
- linphone (linphone.org) - the free SIP server in the Internet, the registration is required
- 3cx (Windows) - for installation on PC
- AsteriskNow (Linux, Windows) - for installation on PC or hardware solution
- FreePBX (Linux) - for installation on PC
- Elastix (Linux) - for installation on PC
- Mobotix - the SIP server built in the call panel of Mobotix T24, T25 IP-intercom
Recommended intercom - call panels. They can be connected to SIP servers as subscribers to make audio and video calls:
- Mobotix T24, T25 (audio and video)
- GUINAZ (audio, support of H264 video in iRidium is planned for future version)
- 2N (audio and video)
- BAS-IP (audio and video)
- True-IP (audio and video)
The supported audio-codecs (it is recommended to deactivate the rest in the settings of the SIP server):
- PCMU (M-Law)
- PCMA (A-Law)
Background mode is not supported (it is not possible to receive call when the app minimized or closed).
Equipment configuration and settings to connect iRidium to the SIP server and to make calls.
SIP server - it can be hardware or software. The server function is to commute calls between the connected subscribers.
SIP subscriber - it is iRidium on smart phones, tablets or PC, IP phone, video-intercom. Subscribers can call each other via the server only.
Setting up of the SIP server
The SIP server (the PBX server) stores the list of subscribers which can call each other. It makes calls to subscribers possible. Via the SIP server you can set up voice connection between iRidium panels, hardware and software IP phones, video-phones and video-intercom in a flat, house or office.
Select the required SIP server and set up several subscribers on it. Each subscriber is a call panel or intercom.
Instructions on setting up servers:
- SIP server 3CX PhoneSystem
- SIP server Mobotix T24 / T25
- SIP server linphone.org (sip.linphone.org): create accounts on the web site - each account has the login you must use as a subscriber's number. The server address: sip.linphone.org
You can find information about setting up of other SIP servers on the web sites of their manufacturers.
Setting up of Call Panels
Download the call panel interface and set it up in iRidium Studio for connection to the SIP server.
1. In Project Device Panel indicate the properties of the SIP server and the subscriber's number which iRidium will be for the SIP driver.
- Name - the driver name. Do NOT change it in the ready interface * Driver Type - the driver type
- Host - the IP-address or domain name of the SIP server
- Port - the port of connection to the SIP server. It is usually 5060, for Mobotix T24/T25: 5061
- Password - the abonent's password for connection to the server
- Caller ID (Name) - the abonent's name (it can be the same as SIP ID)
- SIP ID (Number) - the abonent's number
- First RTP Port - the beginning of the range of ports of voice exchange. It is usually 40000, for Mobotix T24/T25: 7078
- Last RTP Port - the end of the range of ports of voice exchange. It is usually 41000, for Mobotix T24/T25: 9078. This UDP range has to be forwarded on the router when working via the Internet
- Use SIP TONE - to generate SIP messages of tone dialing
- USE DTMF TONE - to generate standard messages of tone dialing (recommended)
- Extrenal IP - the external IP-address of control panel. It is indicated only if external IP address of the remote SIP server used in Host field. To fill this field automatically use the instruction
- Codec PCMU, Codec PCMA, Codec GSM, Codec G722 - SIP audio-codecs. You can turn on 1 codec or all of them if you are not sure which codec supported by the other abonents. Do not turn off all the audio-codecs
- Codec H263, Codec H263-1998 - SIP video-codecs. You can turn on 1 codec or both if you are not sure which codec supported by the other abonents
- Sound correction and filtration. Correction modes should be switched in during a call. Some of modes can be switched with SIP deiver commands (see the ready projects). The needed corrections depends on working conditions:
- Echo Cancellation - echo cancellation
- Volume Control - automatic volume adjustment
- Noise Suppression - noise suppression during a call
- Use Decline - send the DECLINE command together with CANCEL when the call is terminated (required for correct termination of the call when working with some SIP servers, for example 2N)
2. Launch the interface in Emulator to check its work. The call panel status when connecting to the server: On Hook ... (SIP > Feedback > STATUS)
3. To use the call panel without changes, download the project on your control panel.
4. To add the call panel in your ready project, follow the instructions in the video.
Commands and messages of the SIP driver
- ANSWER - to answer an incoming call. Assign the command to the button
- Send Number: 0 - reply with the audio-call
- Send Number: 1 - reply with the video-call
- CALL - to make a call. The number is taken from Feedbacks > NUMBER, where it should be written in advance as a string. Assign the command to the button:
- Send Number: 0 - the audio-call
- Send Number: 1 - the video-call
- CANCEL - to cancel a call. Assign the command to the button, Send Number: 0
- SEND TONE - to send a tone value. Assign the command to the button, Send Number: 1-9, * = 10, # = 11
- REGISTER - to register on the SIP server. Assign the command to the button, Send Number: 0
- UNREGISTER - to disconnect from the SIP server. Assign the command to the button, Send Number: 0
- SET AUDIO VOLUME - to set the volume of loudspeakers (0-100%)
- SET ECHO CANCELLATION - to set the echo cancellation, on/off (1/0)
- SET MIC VOLUME - to set the microphone volume (0-100%)
- SET NOIZE SUPR - to set the noise suppression, on/off (1/0)
- SET SPEECH SEND - to set the voice activation: to send sound if there is speech, otherwise - no sound, on/off (1/0). It works only with activated SET VOLUME CONTROL
- SET VOLUME CONTROL - to set volume control: to lower the volume of the microphone and loudspeakers at speech. SET SPEECH SEND does not work without it.
- AUDIO VOLUME - the volume of loudspeakers (0-100%)
- CODEC - the data type of sound and video: Payload Type (PT) see the full list here
- ECHO CANCELATION - the echo cancellation mode, on/off (1/0)
- FIRST RTP PORT - the first port of the RTP range
- INCOMING CALL - if there is an incoming call: 0 - no, 1 - yes
- INCOMING CALL NAME - caller name, text
- INCOMING CALL NUMBER - the caller’s name, text
- JITTER - (ms) the connection quality, the channel jitter. Values hopping for more than 30 ms can be the indicator of the unstable connection
- LAST RTP PORT - the last port of the RTP range
- MIC VOLUME - the microphone volume (0-100%)
- NOIZE SUPR - the noise suppression mode, on/off (1/0)
- NUMBER - the subscriber's number. Values are written in this Feedback via the dialog window "Send to Project Token > Send Number" in the macros editor to be able to call the indicated subscriber at activation of the CALL command
- STATUS - the current connection status, text. Possible status:
- On Hook... - connected to the SIP server, ready for work
- Connected... - made a connection, getting ready to register
- Disconnected... - forced disconnection from the server (by the command to the driver)
- Failed... - an error at connection (unrelated to the SIP server)
- Trying... a call request is being processed
- Ringing... - the location of the user being called to is determined. The signal about the incoming call is sent
- Incoming Call... - the incoming call
- Talking... - the talk
- Not Found... - the subscriber being called to is not found, there is no such SIP number
- Not Acceptable... - the connection with the server is established but some parameters, such as the type of requested information, bandwidth and addressing type are not available
- Not Available... - the subscriber being called to is not available for calls
- Declined... - the subscriber being called does not want to take calls, without indicating the reason
- Request Pending... - the request is received when the server hasn’t finished processing the other request for this dialog
- Service Unavailable... - the server cannot process the call at the moment because of the overload or maintenance
Emulator: project testing
Emulator - an application launched in iRidium Studio to test the project work.
Emulator work modes see in iRidium Studio > Tools > Options > Emulator:
- Client Fullscreen - start the Emulator in Full Screen mode (press Alt+F4 to Exit)
- Client Sound On - switch sound in app
- Show log at Emuator Start - open the log window automatically at Emulator start (you can also use F4)
- F4 - open the debug log
- F5 - launch Emulator
- F7 - open the app Menu to manage account and download projects from iRidium Cloud
- F8 - open the settings (password: 2007)
Synchronization with control panels
Upload and launch of the iRdidium project on the control panel is performed with the help of the iRidium Transfer application, installed on your PC. You can also upload the project on the panel from the editor via Transfer.
Use i3 pro for iOS, Android, Windows, Mac in Test Mode by downloading projects via iRidium Transfer (the possibility is available for integrators):
iRidium Cloud can be set up only by a registered integrator. After the setting up he can invite end-users to control the automation object.