Displays a list of phones registered on the SIP server.
The list will show the name of the phone being registered and the remaining validity period.

Manually register the device to the SIP server.
Clear the device information to be registered and open the device information registration dialog.

Manually edit the registration information selected in the registered device list.
The device information registration dialog will open with the device information to be modified as the initial values.

Manually delete the registration information selected in the registered device list.

Displays the display name of the registered device.
The display name is taken from the 'From' field of the REGISTER command.
If the display name cannot be obtained, it will be displayed blank.

Displays the SIP-URI registered by the registered terminal.
The SIP server searches for registered terminals based on the registered SIP-URI.
* The SIP server searches for the terminal using the registered SIP-URI, and if not found, searches using the Contact value of the registered terminal.

Displays the IP address or domain name of the registered phone.

Displays the port number of the registered phone.

The Call-ID is displayed when the phone is registered using the REGISTER command.

Displays the remaining validity period of the registered phone.
When the validity period of the registered device reaches "0", the device registration information will be automatically deleted.

Displays the 'Contact' value of the registered phone.
If there are multiple 'Contact' values, the first 'Contact' value will be displayed.

Displays a list of calls that are in progress via the SIP server.

Manually delete any connection information that remains in the connection information list.

Displays the Call-ID of the active call.

Shows the call status

Displays the display names of phone 1 and phone 2.
If there is no display name in the To and From fields of the INVITE command, a blank will be displayed.

Displays the remaining time on the session timer.

Displays the SIP-URI of phone 1 and phone 2.

Displays the duration of the call from the start.

Displays the SIP port addresses of phone 1 and phone 2.

Displays the audio RTP port address and codec type of phone 1 and phone 2.

Displays the video RTP port address and codec type of phone 1 and phone 2.

Displays a list of logs of SIP packets sent and received by the SIP server.

Clear the packet log list.

Indicates the time when the packet was sent or received.

Indicates whether the packet is sent to or received from the SIP server.

The IP address of the destination of the packet.

The port number to which the packet is sent or received.

Displays the message content of the packet.

Displays a list of user names to use for server authentication.

Register a new login user.
Clears the input fields and opens the login user registration dialog.

Edit the information of the user selected in the login user list.
Sets the selected user information in the input fields and opens the login user registration dialog.

Deletes the user information selected in the login user list.

Close the dialog.

Enter the username to use for server authentication.

Enter the password to use for server authentication.

Gets the input and closes the dialog.

Cancels the input and closes the dialog.

Enter the display name of the device you want to register on the server.

Enter the SIP-URI of the device to be registered on the server.

Enter the validity period for the server registration.

Enter the contacts to be registered on the server.

Gets the input and closes the dialog.

Cancels the input and closes the dialog.

Select the method of the packets to be notified.

Enter the address of the destination terminal to which the notification packet will be sent.

Enter the SIP-URI that indicates the destination phone.

Enter the 'Event' field when the notification method is SUBSCRIBE, NOTIFY, or PUBLISH.

If the notification method is SUBSCRIBE or PUBLISH, enter the validity period of the event to be added to the 'Expires' field.

Select the event status to be added to the 'Subscription-State' field when the notification method is NOTIFY.

Enter the data format of the message body to be added to the 'Content-Type' field.

Enter the 'Accept' field.

Enter the event message to be notified.

Loads a saved event data file.

Saves the current input to an event data file.

Close the dialog.

Displays a list of registered virtual phones.

Register a new virtual phone.
Clears the registration and opens the virtual phone registration dialog.

Edit the registration details of the virtual phone selected in the Virtual phone List.
Opens the Virtual Phone Registration dialog with the selected virtual phone registration details as the initial values.

Deletes the registration details of the virtual phone selected in the virtual phone list.

Displays the extension number or user name of the virtual phone.

Displays the IP address or domain name of the SIP port of the virtual phone.

Displays the port number of the virtual phone's SIP port.

Displays the IP address or domain name of the SIP server to which the virtual phone connects.

Displays the port number of the SIP server to which the virtual phone connects.

Displays the user name to be used when receiving an authentication request from the SIP server.

Displays the password used when receiving an authentication request from the SIP server.

Displays the IP address of the RTP port used for audio RTP.

Displays the RTP port number used for audio RTP.

Displays the codec used by the audio RTP port.

Displays output data such as files sent from the audio RTP port.

Displays the IP address of the RTP port used for video RTP.

Displays the RTP port number used for video RTP.

Displays the codec used by the video RTP port.

Displays the file to be sent from the video RTP port.

Close the dialog.

Enter the extension number or user name of the virtual phone.

Select the protocol for the virtual phone from UDP/TCP/TLS.

Enter the virtual phone IP address or domain name.

Enter the port number for the virtual phone.

Enter the IP address or domain name of the SIP server.

Enter the port number of the SIP server.

Enter the user name to be used when receiving an authentication request from the SIP server.

Enter the password to be used when receiving an authentication request from the SIP server.

Enter the IP address of the RTP port to be used for audio RTP.

Enter the RTP port number to be used for audio RTP.
For audio RTP, the RTP port number entered here will be used as follows:

Please specify an even number for the RTP port number.

Select the codec to be used for the audio RTP port.
If you leave this field blank, all supported audio codecs will be listed in the SDP when you INVITE.

Select the file to send from the audio RTP port.

Displays the file name to be sent from the audio RTP port.

Enter the IP address of the RTP port to be used for video RTP.

Enter the RTP port number to be used for video RTP.
For video RTP, the RTP port number entered here will be used as follows:

Please specify an even number for the RTP port number.

Select the codec to be used for the video RTP port.
If you leave this field blank, all supported video codecs will be listed in the SDP when you INVITE.

Select the file to send from the video RTP port.

Displays the file name to be sent from the video RTP port.

Gets the input and closes the dialog.

Cancels the input and closes the dialog.

Displays a list of registered virtual phones.
You can select multiple virtual phones.
Virtual phones that are already open will show "Available" after the phone's display name.
Phones that are already open but whose Virtual Phone dialog is hidden will have "Hidden" after their display name.

Selects all virtual phones in the phone list.

Opens a hidden virtual phone window.
When opening a large number of virtual phones at once during automated testing, etc., and the window becomes a nuisance, you can open it in hidden mode.

Opens all selected virtual phones.
For any virtual phones that are already open but hidden, the Virtual Phones dialog will be displayed.
It does nothing to virtual phone sets that are already open and visible.

Closes the dialog without opening a virtual phone.

Displays a list of open virtual phones.
You can select multiple virtual phones.

Selects all virtual phones in the phone list.

Closes all selected virtual phones.

Closes the dialog without closing the virtual phone.

Enter the phone number or SIP-URI to connect to.

Change the contents of the 'To' field of the INVITE command sent by the virtual phone.

Change the contents of the 'From' field of the INVITE command sent by the virtual phone.

Enter the phone number to connect to.
The phone number you entered will be displayed in the "Dial Input" text box.

An INVITE command will be sent to the phone number or SIP-URI entered in "Dial Input".

Disconnects the current call.

If there is an active call, it will be placed on hold.
If there is a line currently on hold, it will be taken off hold. Putting line 1 on hold will allow line 2 to be used.

Transfers a call on hold on line 1 to line 2.

It performs a registration request to the SIP server.

Sends arbitrary packets from the virtual phone to other phone.

An event packet is sent from the virtual phone to another phone.

Manually send any response to a SIP command received by the virtual phone. Intercepted SIP commands will be treated as "not received" by the virtual phone.

Displays the connection status of telephone line 1.

Displays the connection status of telephone line 2.

Change the content output from audio RTP.

Displays the file name output from the audio RTP port.

Change the content output from video RTP.

Displays the file name output from the video RTP port.

Clears the contents displayed in the message box.

Displays the message body of the NOTIFY, PUBLISH, INFO and MESSAGE commands received by the virtual phone.
The dialed number when a DTMF signal is received is also displayed in the message box.

Send SIP request commands such as NOTIFY commands according to the file created in "Creating event data."

Opens the Virtual Phone Debugging Extensions Settings dialog.

Hides the virtual phone dialog.
To make the dialog visible again, open the "hidden" terminal in the Virtual Phone Open dialog.

Stops the virtual phone and closes the dialog.
Performs deregistration against the SIP server.

Select the type of data to be sent from the RTP port from the following.
For audio RTP

Enter the frequency of the sine wave to be generated when the transmission data type is sine wave output.

Enter the repeating period of the sine wave to be generated when the transmission data type is sine wave output.

When the transmission data type is sine wave output, enter the "signal present" time of the repeating cycle of the sine wave to be generated.

Enter the file name to be sent when the data type is file transmission.

When the transmission data type is file transmission, select the file name to be sent from the file list.

If the data type is file transmission, the file will be sent repeatedly.

Captures the input and closes the dialog.

Cancels the input and closes the dialog.

Select the method of the packets to be notified.

Enter the address of the destination terminal to which the notification packet will be sent.

Enter the SIP-URI that indicates the destination phone.

Enter the 'Event' field when the notification method is SUBSCRIBE, NOTIFY, or PUBLISH.

If the notification method is SUBSCRIBE or PUBLISH, enter the validity period of the event to be added to the 'Expires' field.

Select the event status to be added to the 'Subscription-State' field when the notification method is NOTIFY.

Enter the data format of the message body to be added to the 'Content-Type' field.

Enter the 'Accept' field.

Enter the event message to be notified when the notification method is NOTIFY, PUBLISH, INFO, or MESSAGE.

Loads a saved event data file.

Saves the current input to an event data file.

Sends an event command packet.

Close the dialog.

Enter the destination address and port number of the arbitrary packet.

Enter the content of the arbitrary packet to be sent.

The message entered will be sent to the destination specified in 'Send-To'.

Close the dialog.

Sets the source phone of the SIP request packet.
Set this for SIP servers.

Set the SIP request method to intercept.

Displays the contents of the intercepted SIP request packet.

Enter the content of the SIP response packet to be sent.
As the default value, the contents of the response packet when the response is "200 OK" in response to the intercepted SIP request packet are displayed.

The contents of the entered SIP response packet are sent back to the party that sent the SIP request.

Discards intercepted SIP request packets.

Close the dialog.

Enter the IP address of the SIP server that this application will provide to your SIP phone.

Enter the port number for UDP connection of the SIP server that this application provides to the SIP terminal.

Enter the port number for TCP connection of the SIP server that this application provides to the SIP terminal.

Enter the port number for TLS connection of the SIP server that this application provides to the SIP terminal.

Enter the IP address of this application to communicate with the application "RTPSend".

Enter the port number of this application to communicate with the application "RTPSend".

Enter the IP address of the "RTPSend" application that sends and receives RTP.

Enter the port number for the application "RTPSend".

Enter the IP address of another "SubstSIP" when working in conjunction with another "SubstSIP".

* When two SubstSIPs work together, they notify other "SubstSIPs" of device registration information.
(This is used to recreate an environment where the SIP server and registrar server are separated.
In this case, the SIP server sets this setting to "blank" and the registrar sets this setting to "SIP server address.")

* Tip
This app references the contents of "substsip.ini" when it starts, so you cannot start two or more at the same time from the same folder. However, you can start them simultaneously by copying the contents of the folder to another folder and adjusting the SIP server port number.

Enter the port number of another "SubstSIP" when working in conjunction with another "SubstSIP".

Select the methods that your SIP server supports.

Enter the minimum validity period when registering a terminal with the SIP server using the REGISTER command.

The OPTIONS command is sent at regular intervals to terminals registered on the SIP server.
If you specify "0", the OPTIONS command will not be sent.

Digest authentication is performed when registering a terminal with the SIP server using the REGISTER command.

Even if the device is registered with the SIP server, an authentication request is required when starting a connection.

Specify a certificate file with the .crt extension.

Specify the certificate file with the .crt extension from the file selection dialog.

Specify a certificate file with the .key extension.

Specify the certificate file with the .key extension from the file selection dialog.

The SIP server adds a 'Record-Route' field to the INVITE command that is passed through it.

Select the DTMF output format from the following.

Enter the total duration of the DTMF tone, including any silent parts.

Enter the amount of silence at the start of the DTMF tone.

Enter the duration of the DTMF tone's voiced portion.

Enter the validity period for registering with the SIP server.

Enter the session timer timeout period.
If you enter "0", the session timer will not be used.

Enter the minimum value for the session timer.

Select the method that the virtual phone supports.

Enter the string for the 'User-Agent' field of the virtual phone.

List the supported audio codecs in order of priority.

Increases the priority of the codec selected in the audio codec list by one.

Decreases the priority of the codec selected in the audio codec list by one position.

List supported video codecs in order of priority.

Increases the priority of the codec selected in the video codec list by one.

Decreases the priority of the codec selected in the video codec list by one.

Specify the event data file to be sent when the function key is pressed.

Select the event data file from the file selection dialog.

Enter the T1 timer value.

Enter the T2 timer value.

Enter the T4 timer value.

Specify the folder where the log file is saved when the automated test is executed.

Select the folder where you want to save the log file from the folder selection dialog.

Prevents the SIP server from returning a response when it receives an unsupported method.

The settings are reflected and the dialog box is closed.

Cancel the settings and close the dialog.

Prevents the virtual phone from returning a response when it receives an unsupported method.

When the virtual phone puts the line on hold, it replaces "a=sendonly" with "a=inactive" that is added to the SDP information of the INVITE or UPDATE packet.

Prevents removing registration from SIP server when terminating virtual phone.

Replaces the "To" field of the INVITE request sent from the virtual phone with the specified string.

Replaces the "From" field of the INVITE request sent from the virtual phone with the specified string.

The virtual phone adds SDP to the 1xx response to the INVITE request.

INVITE requests sent from the virtual phone are sent with anonymous settings.

If an abbreviation is defined for a field in a SIP request command, the abbreviation is used.

Select the method that the virtual phone supports.

Enter the string for the 'User-Agent' field of the virtual phone.

The settings are reflected and the dialog box is closed.

Cancel the settings and close the dialog.

Displays the automated tests to be performed.

Displays a list of the status of open virtual phones.

Displays the execution log of the automated tests.

Loads saved automated test data.

Adds a new test command to the end of the current automated test list.

Changes the test command selected in the Auto Test list.

Deletes the selected test command from the automated test list.

Moves the selected test command up one position in the automated test list.

Moves the selected test command down one step in the automated test list.

Clear the automated test list.

Saves the contents of the automated test list to a file.

Starts the automated test run.

Stops any automated tests currently running.

Repeats the contents of the automated test list.

Saves the automated test execution log to a file.

Close the dialog.

Set the extension number of the virtual phone on which you want to run the test.

pecifies the button to press on the virtual phone.

Specify the time to run the test.

Specify the extension number when the call button is pressed.
The parameter is ignored for other buttons.

Select the display language of the application from the following languages.

Displays the homepage where you can purchase the license key.

Enter the user name to register the license.

Enter the license key you purchased.