Open the VoIP Settings dialog to edit the settings required for making VoIP (Voice over IP) phone calls to other devices (e.g. SIP phones or soft phones), i.e., for placing phone calls using network connections instead of the telephone network. You can also use these voice connections for remote-controlling the camera. Likewise, the camera can place (voice) phone calls to other SIP phones, soft phones (computer applications emulating a SIP phone), but also to regular telephones and mobile (cell) phones when using SIP providers.
The term SIP telephony describes a mechanism for establishing voice connections using data networks instead of standard (analog) telephone networks. This method (also known as Voice over IP or short VoIP) allows establishing cheap phone connections (or free connections when connecting two SIP devices) using existing data networks. When using phone servers, VoIP also provides for easy integration of telephone and network services on the same network.
The MOBOTIX cameras support the SIP protocol (Session Initiation Protocol) that two SIP devices use to "negotiate" the parameters of the voice connection. The voice data itself is exchanged using the Real Time Protocol (RTP); it is encoded using the audio codec both devices have agreed on.
The simplest case involves two devices that are using their IP addresses or DNS names to establish a SIP phone connection.
This means that the calling device has to know the SIP address of the device it would like to call. A valid SIP address is
composed of a user name, the "@" character and the IP address of the called device (e.g. fred@123.231.222.213
). If this IP address also has a DNS name (this could be a router with an external IP address in larger networks), this address
could look like this: fred@mysamplecompany.net
.
If SIP providers are used and the SIP accounts have been assigned telephone numbers, the devices can reach each other using the assigned phone numbers, just like calling regular telephones. SIP providers generally also provide connections to "regular" (land-line) telephone numbers and mobile (cell) phones.
To test the SIP phone functionality, using a soft phone on your computer has proven invaluable. Soft phone applications are
nowadays furnished free of charge by good SIP providers. If this is not the case, you can easily find one by searching "free soft phone
" on Google.
Note that you will need two SIP accounts from your SIP provider (one for the camera, one for your SIP phone or soft phone). Creating SIP accounts and placing phone calls between SIP devices should be free of charge.
Make sure that the SIP provider you intend to use also transfers the phone touch tones according to the DTMF standard. If this is not the case, you will not be able to remote-control the camera using the touchtone phone keys. Likewise, you will not be able to receive voice messages from the camera, if the camera requests entering a PIN upon connecting.
Some providers require entering a "geographical" telephone number when trying to call a regular phone from a SIP device. For
example, the phone number of the called party has to start with the country code 0049
for a phone number in Germany. The "0" of the prefix is dropped in most countries: 0049631xxxxxxx
. Note that this depends on the country you are living in and make yourself knowledgeable at your SIP provider, for example.
You have two different options for configuring VoIP: The Quick Setup mode activates VoIP automatically, allowing you to place the first VoIP phone calls with a minimum of manual intervention. The Expert mode allows setting all parameters manually; you can thus use the extended VoIP features and adapt the camera to special network environments.
This section contains the general settings for VoIP telephony.
Parameter |
Description |
---|---|
VoIP |
Use this option to either Enable or Disable the VoIP functionality. NoteIf you deactivate VoIP, the settings in this dialog are still stored in the configuration file of the camera and will remain available if you need them again. |
Hangup on Outgoing Calls |
If this option is activated, the MOBOTIX camera will close an ongoing call if another call has been triggered and will play a sound file on its speaker if Call Status on Camera Speaker has been activated. If this is not the case, the camera will signal the ongoing call by blinking with all LEDs. |
Parallel Dialing |
If this option is enabled, the camera can dial several numbers simultaneously. The camera will establish the connection with the receiver that first accepts the call. NoteWhen using SIP servers that are checking the source ports of the call requests against the source ports of the registration, this feature cannot be used as the ports will differ from the registered ones. |
This section contains the parameters that the camera uses for making SIP connections. Using these parameters, the camera can establish SIP connection using SIP providers or direct connections to a SIP device with an IP address (or DNS name). See the Introduction to SIP Telephony section in this topic for more information.
The SIP accounts used in this example are fictitious. Register with a SIP provider and use your own account information for testing and using SIP telephony.
When entering a SIP address in the MOBOTIX camera dialogs, it is not necessary to add the ("sip:
") protocol prefix at the beginning of the address.
To use a SIP provider with differing registrar and proxy servers, enter it twice. Put the address of the registrar server in the first entry and activate "Use as Registrar". Put the address of the proxy server in the second entry and activate "Available as Proxy". Set the other fields to identical values in both entries. The proxy server can then be selected for outbound calls in the Phone Profiles dialog.
Parameter |
With SIP Provider |
Without SIP Provider |
---|---|---|
User Name (SIP Address) |
Enter a name for the user that should be called on the camera. The name is the left part of the SIP address (the part to the
left of the "@" sign). If the camera's SIP address is This entry usually corresponds to the User Name (Authentication) setting. |
Similar to the scenario with SIP provider, but the SIP domain would be the IP address of the camera ( |
Domain (SIP Address) |
The SIP domain (to the right of the "@" sign) contains the IP address of the device where the user can be found. If the camera's
SIP address is |
Similar to the scenario with SIP provider, but the SIP domain would be the IP address of the camera ( |
User Name (Authentication). |
Enter the user name that you have registered or that you have been assigned during the registration process with your SIP provider. This entry usually corresponds to the User Name (SIP Address) setting. |
Leave this field empty as it is not used. |
Password (Authentication). |
Enter the password for the user name that you have set during the registration process with your SIP provider. |
Leave this field empty as it is not used. |
Hostname / Address (Server) |
Enter the server address for the camera to log on as SIP user. |
Leave this field empty as it is not used. Registration with the SIP server is disabled. |
Port (Server) |
Enter the server port for the camera to log on as SIP user. This will usually be port |
Leave this field empty as it is not used. |
Available as Proxy |
If this option is enabled, the Registrar Server is used as the default SIP proxy for all outgoing SIP connections. The account specified in this entry is available for selection in the field "SIP Proxy" in the Phone Profiles dialog. It can be connected to a profile for outbound calls. |
Disable this option. |
Use as Registrar |
Check this option to have the camera log into the Registrar Server using the given authentication data. |
Disable this option. |
Registration Expires After |
The registration with the registration server expires after this time. In this case, the server will return a "user not reachable" message if you are calling the camera from your soft phone, for example. The camera will automatically renew the registration after this time. |
Since registration has been deactivated anyway if no registration server has been specified, this parameter may be set to any value. |
These settings should only be changed if your network environment involves special mechanism for security (firewall, NAT routers, ...).
Parameter |
Description |
---|---|
NAT Traversal |
Enable this option if you are trying to establish connections via a router or firewall that performs NAT (Network Address Translation). Make sure that you set the NAT Address or STUN Server according to the selected option (see below). |
External router address or STUN server address |
Enter the external IP address or the DNS name of the router or firewall or the STUN server address in this field. When contacting other SIP devices, the camera will name this address or the address received using STUN as its reply address instead of the camera's own (private) IP address. The router/firewall has to redirect the voice and video data to the corresponding port of the camera. |
SIP port |
Enter the SIP port that is to be used e.g. behind a firewall. Only change the standard port |
Audio RTP port |
This is the port that is used for receiving the audio data. You may need to change this setting for some SIP providers or firewall scenarios. If a router or firewall are used that perform NAT (Network Address Translation), the router/firewall will have to redirect this port to the camera. |
Video RTP Port |
This is the port that is used for receiving the video data. You may need to change this setting for some SIP providers or firewall scenarios. If a router or firewall are used that perform NAT (Network Address Translation), the router/firewall will have to redirect this port to the camera. |
Audio Data Timeout |
The camera hangs up the call if there is no incoming audio data during the time specified in this parameter. |
The parameters of this section control the output of voice messages via the phone connection and the camera speaker.
Parameter |
Description |
---|---|
Welcome Message for Inbound Calls |
Switches the output of a welcome message when calling the camera on or off. |
Welcome Message for Outbound Calls |
Switches the output of a welcome message when the camera calls on or off. |
Delay Before Welcome Message |
The camera waits for the specified time after the recipient has accepted the call before playing back its welcome message. |
DTMF Key Confirmation for Inbound Calls |
Switches the voice output of the phone key pressed by the caller on or off. |
DTMF Key Confirmation for Outbound Calls |
Switches the voice output of the pressed phone key when the camera is calling on or off. |
Call Status on Camera Speaker |
Controls the output of sounds to signal the call status on the camera speaker (ring on inbound call, announcement on hangup or disruptions). |
Phone Tones on Camera Speaker |
Upon outgoing calls of the camera, switches the output of the dial tone on the camera speaker on or off. If the output via early media has been enabled, this process gets priority if the camera receives early media data. |
Early Media on Camera Speaker |
Upon outgoing calls of the camera, switches the output of early media (sounds sent from the call recipient before actually accepting the call) on the camera speaker on or off. |
You can set the properties of the On-Screen display in this section.
Parameter |
Description |
---|---|
Auto Hide OSD |
Actives/deactivates automatic hiding of the On-Screen display. |
Setup OSD |
Actives/deactivates the configuration menu of the On-Screen display. |
Home View |
Controls the availability of the Home View in the live image of the On-Screen display. If this parameter is activated, you can use the "0" key to switch between the regular view and view 11. |
The camera can use one of the following audio codecs for transferring voice data. When the camera is making a SIP call, it will try to use the codecs in the order from top to bottom as listed in the dialog. If you would like to use only one codec, deactivate the other ones.
Parameter |
Description |
---|---|
Use G.722 Codec |
The G.722 audio codec creates compressed audio data with a frequency range of nearly 7 kHz (50 to 7,000 Hz) a sampling rate of 16 kHz and 14 bits quantization. The actual bandwidth required by this codec is 64 kbps (8 kByte/s). |
Use PCMA Codec |
The PCMA audio codec creates uncompressed audio data with 8 kHz and a-Law encoding. The actual bandwidth required by this codec is 86 kbps (10.4 kByte/s). |
Use PCMU Codec |
This audio codec corresponds to PCMA/G.711a except for encoding, which uses μ-Law instead of a-Law. |
Use GSM Codec |
This audio codec uses a lossy compression to reduce the size of the audio data; audio quality corresponds to GSM connections. The actual bandwidth required by this codec is 35 kbps (4.2 kByte/s). |
When using video telephony, the camera can adjust the size of the image and the data throughput rate of the video stream.
Parameter |
Description |
---|---|
Video |
This option activates the transmission of video. If deactivated, the camera can only place or receive audio phone calls. |
Video Bit Rate |
Use this parameter to adjust the maximum data throughput rate of the transferred video data. By dynamically adjusting the image quality, the camera tries to stay below the selected value. If the video bit rate drops below 250 kbit/s, the camera automatically reduces the video resolution. In this case, the On-Screen display is not available. |
Preferred Video Codec |
If possible, the camera will try to use the selected video codec for video telephony. This selection does not appear if the camera supports only one codec. |
Click on the VoIP Messages and Call Log link to get detailed information on the status and the connections of the VoIP system of the MOBOTIX camera. For additional information on this dialog, see the VoIP Messages and Call Log help page.
When using the VoIP interface of the camera, the MOBOTIX camera's LEDs signal the following information, if the LED Setup dialog has been set to the default settings.
SIP configured: VoIP LED flashes negatively at 2 Hz.
Camera successfully registered with SIP server: VoIP LED flashes at 2 Hz.
Error: VoIP LED flashes (about 250 msec on/off).
Listen or Speak: VoIP LED is on permanently.
Intercom: The lower three LEDs are lit if the camera's speaker is active ("speaking" as seen by the camera operator). If the camera microphone is active, the upper three LEDs are lit ("listening").
Intercom without switching between microphone and speaker: The top and bottom LEDs are blinking, all others are on continuously.
When making video phone calls, the camera can display a legend for the available functions that are assigned to the phone keys (DTMF). Note that you can also use the features if the legend is not visible.
Phone key "#": Shows/hides the menu, switches between different display modes, stores selections.
Phone key "*": Changes to the next page of the legend, cancels selections and closes submenus.
Phone key "0": If not used otherwise, opens submenus.
Click on the Set button to activate your settings and to save them until the next reboot of the camera.
Click on the Factory button to load the factory defaults for this dialog (this button may not be present in all dialogs).
Click on the Restore button to undo your most recent changes that have not been stored in the camera permanently.
Click on the Close button to close the dialog. While closing the dialog, the system checks the entire configuration for changes. If changes are detected, you will be asked if you would like to store the entire configuration permanently.