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:
|
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. |
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. |
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:
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:
| ||
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.