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Default SipEndpoint.config Settings

Using the Default Configuration File

You can find the default configuration file in the following location:

<installation folder>/Configuration/SipEndpoint.config

This file contains XML configuration details that affect how your SIP Endpoint SDK application behaves. The inital settings are the same as those specified for use with the QuickStart application that is included with your SIP Endpoint SDK release.

Configuration settings are separated into two containers: the Basic Container holds the connectivity details that are required to connect to your SIP Server, while the Cp Container holds a variety of configuration settings.

Basic Container

The first Container ("Basic") holds the basic 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 may be multiple connection elements present with each specifying a separate possible connection. Refer to the configuration settings of that feature for details.

You will have to make the following changes and save the updated configuration file before using the SIP Endpoint SDK:

  • 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.)
  • protocol="TRANSPORT" - Set the protocol attribute to reflect the protocol being used to communicate with SIP Server. Possible values are UDP, TCP, or TLS.

Cp Container

The second Container ("Cp") 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 to take full control over your SIP Endpoint SDK applications.

A full overview of this container, and the settings that are included by default, is provided here:

Domain Section Setting Description
audio
headset audio_in_agc_enabled Set to true to enable AGC for audio via the headset (outgoing audio stream).
incoming use_agc True to apply AGC to incoming streams.
vad continue_sending_from_last_activity_in_milliseconds Sets the amount of time after the point where no voice is detected before audio is actually not sent. It adds a delay to allow a little bit of audio to continue after no voice is detected. This setting is only for codecs that do not have built in support for VAD/DTX, and only if DTX is enabled.
system
qos audio Specifies the type of QOS supported for audio, and if so, whether bandwidth is to be reserved
dtmf force_send_in_band Set as described in RTP:2833:enabled.

One scenario in which it might be advisable to send in band is if you own your gateways, and:

  • One or more of these gateways does not support 2833 or does not handle it well.
  • Your gateway is using codecs that reproduce DTMF tones well.

In this case, setting this setting to true will ensure that DTMF tones get through (because the DTMF tones will bypass the gateway) and that they reproduce accurately at the receiving end.
Another scenario is:

  • One or more of these gateways does not support 2833 or does not handle it well.
  • Your gateway is using codecs that do not reproduce DTMF tones well (because they are designed to handle human voice rather than artificial sounds).

In this scenario, setting this setting to true will not help ensure DTMF tones get through. There is in fact no solution to this scenario.

minimum_rfc2833_play_time If system:dtmf:force_send_in_band is false, specify the minimum play duration for DTMF tones.
network dtx_enabled When DTX is enabled, transmission to the remote party is suspended when the application detects that the local user is not speaking. True means DTX is enabled; silence is not transmitted. False (the default) means silence is transmitted.
diagnostics enable_logging Enables logging is set to 1; disables logging if set to 0.
log_level None, Critical, Error, Warning, Info, Debug, MaxDetails
general add_OS_version_to_user_agent_header Set to true to include the OS version in the SIPUserAgent header field.
indialog_notify enable_indialognotify Set to true to enable in-dialog NOTIFY.
rtp
2833 enabled Set to true to enable local support for RFC 2833 out-of-band DTMF. This setting (a) works with system:dtmf:force_send_in_band setting (b) as follows:
(a) (b) Result
1 1 Send out-of-band 2833; if that is not acceptable, fall back to in-band.
1 0 Send out-of-band 2833; if that is not acceptable, fall back to INFO .
0 1 Send in-band DTMF.
0 0 Send out-of-band INFO.

In-band means the application will encode the DTMF signals in the audio stream as regular sound. Typically, DTMF is not sent in band, and is only used in specific situations. See system:dtmf:force_send_in_band for examples.

hold_over_time_in_ms If system:dtmf:force_send_in_band is false and rtp:2833:enabled is true, specifies the minimum length of time for which to send 2833 packets. This setting is useful in case the user presses a key very fast, to make sure the packet time is longer than that key press.
packet_time_in_ms If system:dtmf:force_send_in_band is false and rtp:2833:enabled is true, specifies the time between 2833 packets. During this time, only audio will be sent. This setting is useful if you cannot handle back-to-back 2833 packets.
payload_number If system:dtmf:force_send_in_band is false, specifies the payload number for DTMF.
inactivity timer_enabled Set to true to instruct the application to hangup when it detects that the RTP session is inactive.
proxies <ref>Note that settings in this section should only be present if using a proxy. Otherwise, all settings and sections should be commented out in your configuration file.</ref>
proxy0 <ref>The proxies domain may have more than one proxy section, and reflects content of the Basic container where each connectivity element has its settings in the corresponding proxy section. For instance, a configuration file with two connections may have two proxies (proxy0 and proxy1) that are equivalent to the SIP Endpoint SDK connections with connection IDs of 0 and 1. Default settings will be applied for any connection that does not have a proxy section.

</ref>

sip_port_range_enable SIP only.

Set to true to force the application to use transport ports within a specific range for RTP (as specified by sip_port_range_min and sip_port_range_max). Set to false to enable the application to use transport ports in the full standard range of 1025 to 65535.

If the port range is set to an incorrect value (for example, if the maximum value is less than the minimum value, or if values that are out of range or uncountable are used) then the default port range values will be used instead.

These settings and the settings with the "sip_" prefix work as follows:

sip_port_range _enable port_range _enable
0 0 Use the full standard range
0 1 Use the range specified by the port_range min and max for both signaling and RTP
1 0 Use the range specified by the sip_port_range min and max for signaling and use the full standard range for RTP
1 1 Use the range specified by the sip_port_range min and max for signaling and the range specified by port_range_min/max for RTP
sip_port_range_min SIP only.

If sip_port_range_enable is true, enter the port value of the lower port in the range. Note that the smallest allowable value for this setting is 10.

sip_port_range_max SIP only.

If sip_port_range_enable is true, enter the port value of the upper port in the range. This parameter must have a value that is greater than the value of the sip_port_range_min setting. Note that the largest allowable value for this setting is 65535.

port_range_enable Set to true to force the application to use transport ports within a specific range (as specified by port_range_min and port_range_max). Set to false to enable the application to use transport ports in the full standard range of 1025 to 65535.

If the port range is set to an incorrect value (for example, if the maximum value is less than the minimum value, or if values that are out of range or uncountable are used), the default port range values will be used instead.

port_range_min If port_range_enable is true, enter the port value of the lower port in the range. Note that the smallest allowable value for this setting is 10.
port_range_max If port_range_enable is true, enter the port value of the upper port in the range. This parameter must have a value that is greater than the value of the port_range_min setting. Note that the largest allowable value for this setting is 65535.
genesyslab
device<ref>See Headset Connectivity Notification for examples and additional details about the settings in the device section.</ref> use_headset Indicates whether a headset is in use. Available options are 1 (true) or 0 (false).
reject_call_when_headset_na Determines call behavior if a headset is not available. Available options are 1 (true) or 0 (false).
error_code_when_headset_na Specifies an error code to return if the headset is not available.
error_message_when_headset_na Specifies an error message to return if the headset is not available.
headset_name Device name of the headset to use.
manual_audio_devices_configure Indicates whether audio devices are manually configured using the audio_in_device and audio_out_device settings. Available options are 1 (true) or 0 (false).
audio_in_device Specifies the audio in device.
audio_out_device Specifies the audio out device.
ringer_device Specifies the ringer device.
system export_settings Specifies file name to store exported from SDK settings if the setting enable_export_settings is true.
enable_export_settings Determines whether the specified export_settings options are used to export system logs. Set to true = 1 to allow the SDK to export and store SDK settings in the file that is specified in setting export_settings.
log_level_AbstractPhone Available values for log level settings are:
  • None
  • Critical
  • Error
  • Warning
  • Info
  • Debug
  • MaxDetails
log_level_Audio
log_level_Auto
log_level_CCM
log_level_Conferencing
log_level_Contacts
log_level_DNS
log_level_GUI
log_level_Jitter
log_level_Licensing
log_level_Media
log_level_Privacy
log_level_RTP
log_level_STUN
log_level_Security
log_level_Storage
log_level_Transport
log_level_USB Devices
log_level_Utilities
log_level_Video
log_level_Voice Quality
log_level_XMPP
log_level_Endpoint
beeptone play_locally Set to true = 1 to allow the SDK to play the beeptone locally, and the agent will hear the beeptone.
enable_beeptone Set to true = 1 to enable the SDK beeptone feature to be executed.
beeptone_file Specifies audio file name to be played as a beeptone.
beeptone_timeout Specifies, in milliseconds, the period of playing a beeptone.
dtmf play_locally Set to true = 1 to allow the SDK to play DTMF locally, and the agent will hear the tone.
pause_start_stop_dtmf Specifies, in milliseconds, the time interval of playing DTMF tone.
control auto_answer Set to true = 1 to allow the SDK to answer incoming calls automatically.

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Additional Configuration Options

The default configuration file does not contain all settings that may be used with the SIP Endpoint SDK; additional settings can be added to change certain behaviors. Check the topic on configuration for a discussion of these additional settings.

This page was last modified on July 26, 2016, at 15:15.

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