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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/SipEndpoint.config file. For an example of the configuration file, see Configuring Genesys Softphone.


Domain Section Setting Default Value Description
Basic Container
Note: Your environment can have up to six SIP URIs (Connectivity sections) that represent six endpoint connections with SIP Server.
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
Policy endpoint public_address empty Specifies the local address of the machine. Set to one of the following values:
  • $auto or empty is a default value for a single local IP address.
  • $host is for multiple NICs; it selects the primary IP address assigned to that host.
  • $fqdn is for fully qualified domain name instead of an IP address in the SIP contact header.
  • IP address or host name.
ip_versions IPv4,IPv6 Specifies the IP version preference.
include_os_version_in_user_agent_header 1 Specifies whether to include the OS version in the User Agent header.
sip_port_min 5060 Specifies the minimum value for the SIP port range. Valid values range from 1 to 65535.
sip_port_max The value of sip_port_min + 6 Specifies the maximum value for the SIP port range. Valid values range from 1 to 65535.
rtp_port_min 9000 Specifies the minimum value for the RTP port range. Valid values range from 1 to 65535.
rtp_port_max 9999 Specifies the maximum value for the RTP port range. Valid values range from 1 to 65535.
rtp_inactivity_timeout 0 Specifies the timeout interval for RTP inactivity.
vq_report_publish 0 Specifies whether to publish the Voice Quality extended report.
vq_report_collector collector@<SIP Server IP Address>:<SIP Server Port>;transport=udp Specifies whether to publish the Voice Quality extended report according to the Connectivity section of the Basic Container.
sip_transaction_timeout 4000 (4 seconds) Specifies the SIP timeout, in milliseconds.
webrtc_audio_layer 2 Specifies the type of audio layer to use. If set to 0, the audio player is defined by the GCTI_AUDIO_LAYER environment variable. If set to 1, the wave audio layer is used. If set to 2, the core audio layer is used. If a value is not specified, the core audio layer is used.
session agc_mode 1 Specifies whether to use the Automatic Gain Control.
dtx_mode 1 Specified whether to use Discontinuous Transmission.
vad_level 1 Specifies whether to use Voice Activity Detection.
reject_session_when_headset_na 0 Specifies whether to reject the session of a headset is not available.
sip_code_when_headset_na 480 Specifies whether to reject the session with the specified SIP error is a headset is not available.
auto_answer 0 Specifies whether to automatically answer incoming calls.
auto_accept_video Specifies whether to automatically accept video stream with audio. Note that this parameter only works if auto_answer is set to 1.
dtmf_method InbandRtp Specifies the type of dual-tone multi-frequency to use.
noise_suppression 0 Specifies whether to control the background noise.
ringing_enabled 1 Specifies whether to enable the ringing tone. If set to 1 or the value is empty, the ringing tone is enabled. If set to 0, the ringing tone is disabled.
ringing_timeout 0 Specified the duration, in seconds, of the ringing tone. If set to 0 or the value is empty, the ringing time is unlimited.
device

Note: The device priority depends on the position of the device in this section of the configuration file.
The higher the position, the higher the device priority. To disable a device, comment out, or remove it from the file. If not configured or empty,
the first listed system devices are selected.

use_headset 0 Specifies whether to use a headset with the Genesys Softphone.
headset_name Specifies the name of the headset.
audio_in_device Specifies the name of the audio input device—for example, Microphone1.
audio_out_device Specifies the name of the audio output device—for example, Speaker1.
capture_device Specifies the name of the capture device—for example, Camera1.
Codecs

Note: The codec priority depends on position of the codec in configuration file.
The higher the position, the higher codec priority. To disable a codec, comment out or remove it from the file.

For the list of supported codecs, see SIP Voice. fmtp Valid values annexb = yes, annexb = no Specifies what Flight Message Transfer Protocol to use with the g729 code. If set to annexb = yes, voice activity detection is enabled. If set to annexb = no, voice activity detection is disabled. For more information, see RFC 3555.
Proxies proxy<n> reg_timeout 1800 Specifies the time, in seconds, after which the registration expires.
reg_interval 3 Specifies the time, in seconds, after which the endpoint starts a new registration cycle when a SIP proxy is down.
display_name Genesys0 The name of the proxy.
password empty The proxy password using WWW-Digest authentication if applicable.
System diagnostics logger_type file Specifies that the log data is to be written to the file specified by the log_file parameter.
log_file SipEndpoint.log Specifies the name and path to the SipEndpoint.log file. You can create log files several ways:
  • Default path—for example, InstallationFolder\logs\SipEndpoint.log.
  • Path using the Environmental variable—for example. %HOMEPATH%\GSP\SIPEndpoint.log.
  • Full path with drive—for example, C:\users\user\GSP\SIPEndpoint.log.
  • Network path—for example, \\network\drive\users\user\GSP\SIPEndpoint.log.

If the setting is left empty, log files are created in the default path. For example, InstallationFolder\logs\SipEndpoint.log.

enable_logging 1 Specifies whether to collect log data.
log_level 3 Specifies the level of detail displayed in the log file. The valid values are:
  • 0="Fatal"
  • 1="Error"
  • 2="Warning"
  • 3="Info"
  • 4="Debug".
log_segmentation 1 MB Specifies the size, in MB, of the log file.
log_expiration 2 Specifies the number of segmented log files to keep.
security cert_file Specifies the certificate thumbprint. This option is valid only if tls_enabled is set to 1.
tls_enabled 0 Specifies whether to the Transport Security Layer (TLS) is enabled.
use_srtp disabled Specifies whether to use the Secure Real-Time Transport Protocol (SRTP).
media ringing_file <Genesys Softphone Installation Directory>/ringing.mp3 Specifies the audio file that is played when the ringing_enabled option is set to 1.


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

Retrieved from "https://docs.genesys.com/Documentation:SP:dep:Options:8.5.0 (2019-06-20 05:58:32)"
This page was last modified on November 6, 2014, at 13:10.

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