Jump to: navigation, search

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.

Basic Container

Important
Your environment can have up to six SIP URIs (Connectivity sections) that represent six endpoint connections with SIP Server.
Domain Section Setting Default Value Description
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.

Genesys Container

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:

Domain Section Setting Values Description
policy
endpoint
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

Default value: 3

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.

Default value: status,calls,devices

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.
ip_versions IPv4

IPv6
IPv4,IPv6
IPv6,IPv4
empty

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.
A value of IPv4,IPv6 or an empty value means that the application selects an IPv4 address if one exists. If not, an available IPv6 address is selected.
A value of IPv6,IPv4 means that the application selects an IPv6 address if one exists. If not, an available IPv4 address is selected.
Default: IPv4.
NOTE: This parameter has no effect if the public_address option specifies an explicit IP address.

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.

Valid Values:
This setting may have one of the following explicit values:

  • An IP address. For example, 192.168.16.123 for IPv4 or FE80::0202:B3FF:FE1E:8329 for IPv6.
  • A bare host name or fully qualified domain name (FQDN). For example, epsipwin2 or epsipwin2.us.example.com.

This setting may have one of the following special values:

  • $auto—The GSP selects the first valid IP address on the first network adapter that is active (status=up) and has the default gateway configured. IP family preference is specified by the policy.endpoint.ip_versions setting.
  • $ipv4 or $ipv6—Same behavior as the $auto setting but the GSP restricts the address to a particular IP family.
  • $host—The GSP retrieves the standard host name for the local computer using the gethostname system function.
  • $fqdn—The GSP retrieves the fully qualified DNS name of the local computer. The GSP uses the GetComputerNameEx function with parameter ComputerNameDnsFullyQualified.
  • An adapter name or part of an adapter name prefixed with $. For example, $Local Area Connection 2 or $Local. The specified name must be different from the special values $auto, $ipv4, $host, and $fqdn.

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
webrtc_audio_layer 0
1
2
Valid values:

0—the audio layer is defined by environment variable "GCTI_AUDIO_LAYER"
1—Wave audio layer is used
2—Core audio layer is used

session
agc_mode 0

1

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.
dtmf_method Rfc2833

Info
InbandRtp

Method to send DTMF
echo_control 0
1
Valid values: 0 or 1. If set to 1, echo control is enabled.
noise_suppression 0
1
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
1 = Play ringtone through system default device only. Configure media in system.media.ringing_file.
2 = Play ringtone through communication device (headset) only. Configure media in policy.session.ringing_file.
3 = Play ringtone through both devices at the same time.
Default Value: 1
Specifies whether to enable the ringing tone and on which device to play the media file.

ringing_timeout Number Valid Values: Empty, 0, or a positive number

Default Value: 0
Specifies the duration, in seconds, of the ringing tone. If set to 0 or if the value is empty, the ringing time is unlimited.

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
Specifies the audio file that is played in the audio out device (headset) when the ringing tone is enabled with the ringing_enabled option.
Note that WebRTC does not support MP3 playback. The ringtone file for built-in ringing should be a RIFF (little-endian) WAVE file using one of the following formats:

kWavFormatPcm = 1, PCM, each sample of size bytes_per_sample
kWavFormatALaw = 6, 8-bit ITU-T G.711 A-law
kWavFormatMuLaw = 7, 8-bit ITU-T G.711 mu-law


Uncompressed PCM audio must 16 bit mono or stereo and have a frequency of 8, 16, or 32 KHZ.

device
audio_in_device

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.
codecs
— See SIP Endpoint SDK for .NET—Working with Codec Priorities
proxies
proxy<n>
display_name String Proxy display name
password String Proxy password
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.
Important
The 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
This setting controls whether or not SIP Endpoint SDK should re-register itself when receiving a mismatched IP address in the received parameter of a REGISTER response. This helps resolve the case where SIP Endpoint SDK for .NET 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 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.
nat
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
system
diagnostics
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.
log_segment false
Number
Number in KB,MB, or hr
Valid Values:

false: No segmentation is allowed
<number> or <number> KB: Size in kilobytes
<number> MB: Size in megabytes
<number> hr: Number of hours for segment to stay open
Deafult Value: 10 MB
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 logfile.

log_expire false
Number
Number file
Number day
Valid Values:

false: No expiration; all generated segments are stored.
<number> or <number> file: Sets the maximum number of log files to store. Specify a number from 1—1000.
<number> day: Sets the maximum number of days before log files are deleted. Specify a number from 1100
Deafult Value: 10 (store 10 log fragments and purge the rest)
Determines whether 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.

log_time_convert local
utc
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.
utc: The time of log record generation is expressed as Coordinated Universal Time (UTC).
Default Value: local
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).

log_time_format time
locale
ISO8601
Valid Values:

time: The time string is formatted according to the HH:MM:SS.sss (hours, minutes, seconds, and milliseconds) format
locale: The time string is formatted according to the system’s locale.
ISO8601: The date in the time string is formatted according to the ISO 8601 format. Fractional seconds are given in milliseconds.
Default Value: time
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.

security
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.
use_srtp String

disabled optional
mandatory

Indicates whether to use SRTP
media
ringing_file
String Valid Values: Empty or String file name

Defaul Value: ringing.mp3
The Ringing sound file name in the current directory or the full local path to the ringing sound file. Specifies the audio file that is played in the defualt audio device (speakers) when the default device ringing tone is enabled with the ringing_enabled option.

For more information about these options, see SIP Endpoint SDK for .NET Developer's Guide.

This page was last edited on December 8, 2017, at 17:37.
Comments or questions about this documentation? Contact us for support!