Genesys Softphone configuration options
This article lists and describes, by container and then by domain, the configuration settings in the <Genesys Softphone Installation Directory>\Softphone.config file. For an example of the configuration file, see Configuring Genesys Softphone. It also describes how to configure Genesys Softphone to work with the audio devices that you use in your environment.
When you install Genesys Softphone, either by using the Genesys Installation Wizard or silently by command line, the Softphone.config and genesys_softphone.exe files are both installed. The contents of the Softphone.config file is generated by the choices you specify in the wizard or by modifications you make to the genesys_silent.ini file.
In the Softphone.config file, the setup.exe executable sets the values of the following attributes of the Connector section: protocol, port, and certificate_search_value. The enable_sessionid and auto_restart are not set by the executable; you must set these yourself. The default values of these attributes are designed to address most business deployments. However, if you want to adjust their values, follow these steps to make a custom deployment:
- Install Genesys Softphone on an administrator's machine.
- Edit the Softphone.config file to change the values of the attributes in the Connector section.
- Repackage Genesys Softphone with the custom Softphone.config file through an IT-controlled installation.
- Push the custom package to the agent workstations.
The Basic container holds the connectivity details that are required to connect to your SIP Server. This container has at least one connection (Connectivity) element with the following attributes:
<Connectivity user="DN" server="SERVER:PORT" protocol="TRANSPORT"/>
If you are using a configuration that supports Disaster Recovery and Geo-Redundancy, there can be multiple connection elements present, with each element specifying a separate possible connection. Refer to the configuration settings of that feature for details.
You must make the following changes and save the updated configuration file before using Genesys Softphone:
- user="DN": Supply a valid DN for the user attribute.
- server="SERVER:PORT": Replace SERVER with the host name where your SIP Server is deployed, and PORT with the SIP port of the SIP Server host. The default SIP port value is 5060. For SRV resolution, specify the SRV record without including the port number in the server's URI. Also see SRV Resolution below.
- protocol="TRANSPORT": Set the protocol attribute to reflect the protocol being used to communicate with SIP Server. Possible values are UDP, TCP, or TLS.
When using an SRV record for the server parameter, note the following:
- Do not specify the port in the server URI.
- Genesys Softphone does not take into account the weight field of an SRV record.
- You cannot combine IPv4 and IPv6 for a single FQDN.
- The maximum number of targets (SRV records) per service is 20.
- You can only specify SRV records in the server parameter of the Connectivity element. You cannot use SRV records for the mailbox section or the vq_report_collector setting.
|Connectivity||user||The first user's DN extension as configured in the configuration database. Included in the SIP URI. For example, <sip:DN0@serverHostName0:port0>|
|server||The SIP Server or Proxy location for the first user. Included in the SIP URI. For example, <sip:DN0@serverHostName0:port0>|
|protocol||The transport procotcol for the first user. For example, UDP, TCP, or TLS.|
|For more information, see the Basic Container description in the SIP Endpoint SDK for .NET Developer's Guide.|
The Genesys container holds configurable settings that are organized into domains and sections. You don't have to change these settings but you can customize them.
The following table describes the settings in this container and their valid values:
|defer_device_release||Integer||Specifies a time in milliseconds before releasing audio devices after the audio stream has been stopped. Deferring device release avoids potential service interruptions if the audio will be restarted quickly and if audio device operations are too slow on the user workstation or has other problems with restart. The value 0 disables deferred device release.
|gui_call_lines||Number from 1 to 7||This option controls the number of phone lines in the First Party Call Control tab.
|gui_tabs||Comma-separated list of tab names||This option controls what tabs are shown in the GUI and their order.
|include_mac_address||Number||If set to 1, the MAC address is included in the Contact header of the REGISTER message of the host's network interface in a format compatible with RFC 5626.
|include_os_version_in_user_agent_header||Number||If set to 1, the User Agent field includes the OS version the client is currently running on.
|include_sdk_version_in_user_agent_header||Number||If set to 1, the User Agent field includes the SDK version the client is currently running on.
|public_address||String||Local IP address or Fully Qualified Domain Name (FQDN) of the machine. This setting can be an explicit setting or a special value that the GSP uses to automatically obtain the public address.
This setting can have one of the following special values:
Default value: Empty string which is fully equivalent to the $auto value.
|rtp_inactivity_timeout||Number||Timeout interval in seconds for RTP inactivity.
|rtp_port_min||Number||The integer value representing the minimum value for an RTP port range. Must be within the valid port range of 1 to 65535. If the minimum and maximum values are not specified or are set to an invalid value, the default minimum (9000) and maximum (minimum value + 999) are used. Setting the minimum to a value that is larger than the maximum is considered an error and will result in a failure to initialize the endpoint.|
|rtp_port_max||Number||The integer value representing the maximum value for an RTP port range. Must be within the valid port range of 1 to 65535. If the minimum and maximum values are not specified or are set to an invalid value, the default minimum (9000) and maximum (minimum value + 999) are used. Setting the maximum to a value that is less than the minimum is considered an error and will result in a failure to initialize the endpoint.|
|tcp_port_min||Number||The integer value representing the minimum value for a TCP client-side port range. Must be within the valid port range of 1 to 65535. If set to 0 (default) or if the configured range is not valid, SIP connections over TCP and TLS use ephemeral ports, assigned by the operating system.|
|tcp_port_max||Number||The integer value representing the maximum value for a TCP client-side port range. Must be within the valid port range of 1 to 65535.
If set to 0 (default) or if the configured range is not valid, SIP connections over TCP and TLS use ephemeral ports, assigned by the operating system.
If the value is non-zero and greater than the tcp_port_min value, this value specifies the maximum value for a TCP client-side SIP port range that will be used for all outgoing SIP connections over TCP and TLS transport.
|sip_port_min||Number||The integer value representing the minimum value for a SIP port range. Must be within the valid port range of 1 to 65535. If the minimum and maximum values are not specified or are set to an invalid value, the default minimum (5060) and maximum (minimum value + 6) are used. Setting the minimum to a value that is larger than the maximum is considered an error and will result in a failure to initialize the endpoint.|
|sip_port_max||Number||The integer value representing the maximum value for a SIP port range. Must be within the valid port range of 1 to 65535. If the minimum and maximum values are not specified or are set to an invalid value, the default minimum (5060) and maximum (minimum value + 6) are used. Setting the maximum to a value that is less than the minimum is considered an error and will result in a failure to initialize the endpoint.|
|sip_transaction_timeout||Number||SIP transaction timeout value in milliseconds.
|vq_alarm_threshold||0 (default) or number from 1.0 to 5.0||Specifies Mean Opinion Score (MOS — a measure of reported network quality ratings) threshold for generating Voice Quality Alarms. The value 0 disables the alarms. The recommended threshold value is 3.5. Using values above 4.2 are not recommended as an MOS that high might not be obtainable with some codecs, even under perfect network conditions.|
|vq_report_collector||See SIP Endpoint SDK for .NET—Producing RTCP Extended Reports|
|vq_report_publish||See SIP Endpoint SDK for .NET—Producing RTCP Extended Reports|
|If set to 0, AGC (Automatic Gain Control) is disabled; if set to 1, it is enabled. Other values are reserved for future extensions. This configuration is applied at startup, after that the agc_mode setting can be changed to 1 or 0 from the main sample application.
|Enables and disables Receiving-side Automatic Gain Control (Rx AGC). [Added: 9.0.xxx.xx]
|auto_answer||Number||If set to 1, all incoming calls are answered automatically.|
|Valid values: 0 or 1 (default). If set to 1, DTMF feedback (audio tones played locally when sending DTMF signals to the remote side of the conversation) is enabled. See also: dtmf_method. (Added: 9.0.018.04)|
|Method to send DTMF when dtmf_feedback is enabled.|
|Valid values: 0 or 1. If set to 1, echo control is enabled.|
|Valid values: 0 or 1. If set to 1, noise suppression is enabled.|
|dtx_mode||Number||Valid values: 0 or 1. If set to 1, DTX is activated.|
|reject_session_when_headset_na||Number||Valid values: 0 or 1. If set to 1, the GSP rejects the incoming session if a USB headset is not available.|
|sip_code_when_headset_na||Number||If a valid SIP error code is supplied, the GSP rejects the incoming session with the specified SIP error code if a USB headset is not available.
|vad_level||Number||Sets the degree of bandwidth reduction.
|ringing_enabled||0, 1, 2, 3, 4, 5, 6, or 7||Specifies whether to enable the ringtone and on which device to play the media file.
Default value: 1
|ringing_timeout||Number||Specifies the duration, in seconds, of the ringtone. If set to 0 or if the value is empty, the ringing time is unlimited.
|ringing_file||String||Specifies the audio file that is played in the audio out device (headset) when the ringtone is enabled with the ringing_enabled option.|
Note that WebRTC does not support MP3 playback. The ringtone files for built-in ringing are RIFF (little-endian) WAVE files using one of the following formats:
Uncompressed PCM audio must be 16 bit mono or stereo and have a frequency of 8, 16, or 32 KHZ.
Sample ringing sound files are provided. Each file is adjusted to have a higher or lower level of volume than the default ringing sound file, as follows:
You can adjust the volume level of the default ringer sound by specifying the file that has the preferred level of volume.
Valid values: Empty or the path to the ringing sound file for the audio out device (headset). The path can be a filename in the current directory or the full path to the sound file.
Default value: ringing.wav
|ringback_enabled||0, 1, 2, 3, 4, or 6||Specifies how the ringback feature is enabled.
|ringback_file||Empty or a valid path to a 16-bit 8-, 16-, or 32-Khz .wav sound file.||Specifies the audio file that is played when the ringback_enabled option is configured to play a local file as the ringback tone.|
|audio_in_device||String||Microphone device name: can be either the device proper name or a regular expression.|
|audio_out_device||String||Speaker device name: can be either the device proper name or a regular expression.|
|ringer_device||String||Ringer device name: can be either the device proper name or a regular expression. Used when ringing_enabled = 4|
The name of the headset model: can be either the device proper name or a regular expression. When the value of the use_headset option is set to 1, you can set the value of this option to *, the default value, to select the default headset. If the value of this option is empty, this option is not considered as a regular expression and will fail to find a headset.
|use_headset||Number||If set to 0, the audio devices specified in audio_in_device and audio_out_device are used by the Genesys Softphone. If set to 1, the Genesys Softphone uses a headset as the preferred audio input and output device and the audio devices specified in audio_in_device and audio_out_device are ignored.
|exclude_headset||String||The name of a headset model or built-in audio device (example: Realtek). The specified device is excluded from being recognized or automatically selected as a valid headset.
Note that components of an excluded device, such as a microphone or speaker, can still be selected manually (or automatically, if Softphone does not find a better device). However, even if a component of an excluded device is selected, Softphone does not recognize the excluded device as an available headset.
Note: This option is only available for Genesys Softphone installed on Windows.
|auto_restart||Number||Valid values: 0 or 1. If set to 1 (default) the Softphone must be restarted after every client session.|
|certificate_search_value||String||The thumbprint of a valid certificate that is accessible from the Certificate Store of the workstation where Softphone is running.|
|enable_sessionid||Number||Valid values: 0 or 1. If set to 1 (default), a SESSION_ID attribute is generated in the header of the HTTP response returned to the HTTP Client (typically Workspace Web Editon (WWE) running in a browser). This option must always be set to 0 when connecting Workspace Desktop Edition (WDE) to Genesys Softphone.|
|port||Number||The port that Softphone opens at start-up time to listen to HTTP or HTTPS requests sent by the HTTP Client (typically WWE running in a browser). If sent to empty value (default) the connector is not activated and Softphone runs in regular stand-alone GUI mode.|
|protocol||String||Valid values: http or https. Specifies whether the HTTP requests sent from HTTP client (typically WWE running in a browser) are secured. If set to a non-empty value the option port must be populated with a valid port number. If set to https, the option certificate_search_value must be populated with a valid certificate thumbprint.|
codecs— See SIP Endpoint SDK for .NET 9.0.0NET—Working with Codec Priorities.
|display_name||String||Proxy display name.|
|reg_interval||Number||The period, in seconds, after which the endpoint starts a new registration cycle when a SIP proxy is down. Valid values are integers greater than or equal to 0. If the setting is empty or negative, the default value is 0, which means no new registration cycle is allowed. If the setting is greater than 0, a new registration cycle is allowed and will start after the period specified by regInterval.|
ImportantThe re-registration procedure uses a smaller timeout (half a second) for the first re-try only, ignoring the configured reg_interval setting; the reg_interval setting is applied to all further retries.
|reg_match_received_rport||Number||DEPRECATED: This setting controls whether or not Genesys Softphone re-registers itself when receiving a mismatched IP address in the received parameter of a REGISTER response. This helps resolve the case where the Genesys Softphone has multiple network interfaces and obtains the wrong local IP address. A value of 0 (default) disables this feature and a value of 1 enables re-registration.
|reg_timeout||Number||The period, in seconds, after which registration expires. A new REGISTER request will be sent before expiration. Valid values are integers greater than or equal to 0. If the setting is 0 or empty/null, then registration is disabled, putting the endpoint in stand-alone mode.|
|ice_enabled||Boolean||Enable or disable ICE.|
|stun_server||String||STUN server address. An empty or null value indicates this feature is not being used.|
|stun_server_port||String||STUN server port value.|
|turn_password||Number||Password for TURN authentication.|
|turn_relay_type||Number||Type of TURN relay.|
|turn_server||String||TURN server address. An empty or null value indicates this feature is not being used.|
|turn_server_port||String||TURN server port value.|
|turn_user_name||String||User ID for TURN authorization.|
|enable_logging||Number||Disable or enable logging.
|log_file||String||Log filename, for example, SipEndpoint.log|
|log_level||Number||Valid values: 0 – 4|
Log levels: 0 = "Fatal"; 1 = "Error"; 2 = "Warning"; 3 = "Info"; 4 = "Debug".
|log_options_provider||String||Valid values for webrtc = (warning, state, api, debug, info, error, critical). For example: gsip=2, webrtc=(error,critical)|
|logger_type||file||If set to file, the log data will be printed to the file specified by the log_file parameter.|
Number in KB,MB, or hr
|Specifies the segmentation limit for a log file. If the current log segment exceeds the size set by this option, the file is closed and a new one is created. This option is ignored if log output is not configured to be sent to a log file.
|Determines whether the log files expire. If they do, sets the measurement for
determining when they expire, along with the maximum number of files (segments) or days before the files are removed. This option is ignored if log output is not configured to be sent to a log file.
|Specifies the system in which an application calculates the log record time when generating a log file. The time is converted from the time in seconds since the Epoch (00:00:00 UTC, January 1, 1970).
|Specifies how to represent, in a log file, the time when an application generates log records. A log record’s time field in the ISO 8601 format looks like this: 2001-07-24T04:58:10.123.
|certificate||String||Thumbprint value of the Public endpoint certificate file, which is used as a client-side certificate for outgoing TLS connections and server-side certificate for incoming TLS connections. For example: 78 44 34 36 7a c2 22 48 bd 5c 76 6b 00 84 5d 66 83 f5 85 d5|
allowed disabled off elective both enabled force mandatory
|Indicates whether to use SRTP (Secure Real-Time Transport Protocol) [Modified: 9.0.005.06]
Adding either ',UNENCRYPTED_SRTCP' (long form) or ',UEC' (short form) to any value (for example, 'enabled,UEC'), adds the UNENCRYPTED_SRTCP parameter to that offer. When this parameter is negotiated, RTCP packets are not encrypted but are still authenticated.
|Enables a case insensitive comparison of the TLS host name and the certificate’s subject field during the authentication process. This option is transferred to a third-party library and describes whether it is necessary or not to check the names.
ImportantThe default value for this option (no) is only to ensure backward compatibility with previous releases. Genesys recommends to always set the value as host in production environments for security reasons to avoid any man-in-the-middle attack attempts.
|ringing_file||String||The Ringing sound filename in the current directory or the full local path to the ringing sound file. Specifies the audio file that is played in the default audio device (speakers) when the default device ringtone is enabled with the ringing_enabled option.
You can adjust the volume level of the default ringer sound by specifying the file that has the preferred level of volume.
Valid values: Empty or String filename
For more information about these options, see SIP Endpoint SDK for .NET Developer's Guide.
Audio device settings
This section describes how to set up Genesys Softphone to work with your audio devices, such as headsets.
Genesys Softphone uses the following criteria to select its audio input and output devices:
- Basic settings for audio input and output devices.
- Selection rules used to choose an audio device, auto-answer a call, and reject a call.
- Combinations of settings that affect audio device selection, auto-answer, and call rejection.
Use the following parameters to configure headsets and other audio input devices:
If none of the audio devices that are accessible to the endpoint, match the device names in the configuration file; Genesys Softphone picks up the first available devices from the WebRTC list for audio devices.
The headset_name, audio_in_device, and audio_out_device options support both device proper names and regular expressions.
Audio device selection rules
The following rules are used to select an audio device, auto-answer a call, and reject a call.
The following audio device selection procedure is applied on startup and every time any changes are made to device presence (such as when a new device is plugged in or an existing device is removed):
- The first device in the applicable list that is present in the system is selected when possible. This device (or devices) will either be specified by the headset_name parameter or by the audio_in_device and audio_out_device parameters, depending on whether the use_headset parameter has been enabled.
- If none of the configured devices are present (or if the configuration list is empty), then Genesys Softphone selects the audio devices using the priority provided by WebRTC, based on the order of the available devices in its device list.
When either of the following conditions is met, the SDK blocks the auto-answer functionality (a policy of should answer returns the value unknown; a manual answer is still possible):
- the use_headset parameter is set to 1, and none of the devices listed in the headset_name settings are present (but session rejection is not applicable, that is, the reject_session_when_headset_na parameter has been set to 0).
- Genesys Softphone was unable to find any usable microphone or speaker device (applicable to cases where the use_headset parameter is set to 0).
Finally, if the auto_answer parameter is set to 1 and the auto-answer functionality is not blocked (and the call was not already rejected), Genesys Softphone answers the incoming call automatically (the should answer policy returns the value true).
Rejecting a call
For backward compatibility with previous releases, a call can only be rejected when both of the following conditions are met (a policy of should answer returns the value false):
- Both the use_headset and reject_session_when_headset_na parameters are set to 1.
- None of the devices listed in the headset_name settings is present on the workstation.
When these conditions are both met, an incoming call is rejected with the SIP response code that is configured in the sip_code_when_headset_na setting. If the setting is missing or the value is not in the valid range of 400 to 699, then the default value of 480 (Temporarily Unavailable) is used.
In addition, when these conditions are met, Genesys Softphone refuses to initiate any new calls; it rejects all outgoing call attempts.
The availability of a fallback device (selected by Step 2 in the Audio device selection section) does not affect call rejection.
Audio setting combinations
Sometimes combinations of settings that you make can have unexpected results. Before adjusting your settings, review this section. The following combinations of settings affect audio device selection, auto-answer, and call rejection in the ways described below:
|Headset is available
Genesys Softphone considers a headset to be available if a headset is found by name in the list of headset names stored in the headset_name parameter. (The highest priority device in the list is selected).
Outgoing calls can be initiated.
|auto_answer=1||Incoming calls are answered automatically.|
|auto_answer=0||Incoming calls are answered manually.|
|Headset is not available
Genesys Softphone determines that no headset is available if a headset is not found by name in the list of headset names stored in the headset_name parameter.
An audio device is still assigned if any supported devices are present in the system, using the first available audio input and output devices from the list compiled by WebRTC.
|No auto-answer is possible in this subcase, so the auto_answer setting is not used.||reject_session_when_headset_na=1
Audio devices are configured using the names from the audio_in_device and audio_out_device settings. Genesys Softphone selects the highest-priority input and output devices from that list or, if no valid devices are found in that list, from the first available devices in the list compiled by WebRTC. Outgoing calls can be initiated.
|Both microphone and speaker are available||auto_answer=1||Incoming calls are answered automatically.|
|auto_answer=0||Incoming calls are answered manually.|
|Either microphone or speaker is not available
||No auto-answer is possible in this subcase, so the auto_answer setting is not used.||Auto-rejection is not applicable, so the reject_session_when_headset_na setting is not used.|