Configuration Options Reference
This section lists and describes, by container and then by domain, the configuration settings found in the <Genesys Softphone Installation Directory>/Genesys Softphone/GenesysSoftphone/Softphone.config file. For an example of the configuration file, see Configuring Genesys Softphone.
|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 second Container ("Genesys") holds a number of configurable settings that are organized into domains and sections. These settings do not have to be changed, but can be customized.
An overview of the settings in this container and the valid values for these settings is provided here:
|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. Default: 1.|
|gui_call_lines||Number from 1 to 7|| This option controls the number of phone lines in the First Party Call Control tab.|
Valid values: Integer between 1 and 7
|gui_tabs||Comma-separated list of tab names|| This option controls what tabs are shown in the GUI and their order.|
Valid values: Comma-separated list of tab names in any order. The tab names are status, calls,and devices. Names may be shortened to stat, call, and dev. The value is case-sensitive. This option ignores unrecognizable and duplicate tab names. If the setting is present but has an incorrect value, the value will fall back to the single tab status.
|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. Default: 1.|
| A value of IPv4 means that the application selects an available local IPv4 address; IPv6 addresses are ignored.
A value of IPv6 means that the application selects an available local IPv6 address; IPv4 addresses are ignored.
|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 may have one of the following special values:
Default Value: Empty string which is fully equivalent to the $auto value.
If the value is specified as an explicit host name, FQDN, or $fqdn, the Contact header includes the host name or FQDN for the recipient of SIP messages (SIP Server or SIP proxy) to resolve on their own. For all other cases, including $host, the resolved IP address is used for Contact. The value in SDP is always the IP address.
|rtp_inactivity_timeout||Number||Timeout interval for RTP inactivity. Valid values are positive integers. A value of 0 means that this feature is not activated. A value 1 or higher indicates the inactivity timeout interval in seconds. Default: 0. Suggested values: 1 through 150.|
|rtp_port_min||Number||The integer value representing the minimum value for an RTP port range. Must be within the valid port range of 9000 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 9000 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.|
|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. Valid values are 1 through 32000, with a default value of 4000. The recommended value is 4000.|
|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|
| Valid values:|
0—the audio layer is defined by environment variable "GCTI_AUDIO_LAYER"
| If set to 0, AGC (Automatic Gain Control) is disabled; if set to 1, it is enabled. Default: 1. Other values are reserved for future extensions. This configuration is applied at startup, after which time the agc_mode setting can be changed to 1 or 0 from the main sample application.
NOTE: It is not possible to apply different AGC settings for different channels in multi-channel scenarios.
|auto_answer||Number||If set to 1, all incoming calls should be answered automatically.|
|Method to send DTMF|
|Valid values: 0 or 1. If set to 1, echo control is enabled.|
|Valid values: 0 or 1. If set to 1, noise suppresion 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 should reject the incoming session if a USB headset is not available.|
|sip_code_when_headset_na||Number|| Defaul Value: 480|
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. Valid values: 0 – 3 — from 0 (conventional VAD) to 3 (aggressive high).|
|ringing_enabled||Number|| Valid values: 0, 1, 2, or 3.|
0 = None, disable ringtone
|ringing_timeout||Number|| Valid Values: Empty, 0, or a positive number|
Default Value: 0
|ringing_file||String|| Valid values: Empty or the path to the ringing sound file for the audio out device (headset). The path may be a file name in the current directory or the full path to the sound file.|
Default Value: ringing.wav
kWavFormatPcm = 1, PCM, each sample of size bytes_per_sample
For more information, see SIP Endpoint SDK for .NET—Audio Device Settings
|String||Microphone device name|
|audio_out_device||String||Speaker device name|
|headset_name||String||The name of the headset model|
|use_headset||Number||Valid values: 0 or 1. If set to 0, the audio devices specified in audio_in_device and audio_out_device are used by the SDK. If set to 1, the SDK 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.|
|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.|
|port||Number||The port that Softphone is opening 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 standalone GUI mode.|
|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 (currently not supported), a SESSION_ID attribute is generated in the header of the HTTP response returned to the HTTP Client (typically WWE running in a browser).|
codecs— See SIP Endpoint SDK for .NET—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|| Valid Values: 0 or 1|
Default Value: 0
|reg_timeout||Number||The period, in seconds, after which registration should expire. 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 standalone 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||Valid values: 0 or 1. Disable or enable logging.|
|log_file||String||Log file name, 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
| Valid Values:|
false: No segmentation is allowed
| Valid Values:|
false: No expiration; all generated segments are stored.
| Valid Values:|
local: The time of log record generation is expressed as a local time, based
on the time zone and any seasonal adjustments. Time zone information of the application’s host computer is used.
| Valid Values:|
time: The time string is formatted according to the HH:MM:SS.sss (hours, minutes, seconds, and milliseconds) format
|cert_file||String||Thumbprint value of the Public endpoint certificate file, which is used as a client-side certificate for outgoing TLS connection 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|
|tls_enabled||Number||If set to 1, connection with TLS transport will be registered. Default: 0.|
|Indicates whether to use SRTP|
|ringing_file||String|| Valid Values: Empty or String file name|
Defaul Value: ringing.mp3
For more information about these options, see SIP Endpoint SDK for .NET Developer's Guide.