Create base settings for a PureCloud softphone

AI-generated summary

Genesys Cloud requires administrators to establish comprehensive base settings configurations for PureCloud softphones through the Admin interface, accessible via Telephony > Phone Management > Base Settings, which necessitates Telephony > Plugin > All permissions. These configurations define operational parameters across four expandable sections: General, Media, Network, and Custom. The General section enables Dynamic Reload functionality for dynamic phone configuration reloading. Media settings allow administrators to configure Quality of Service through DSCP values placed in IP header TOS fields, define RTP Audio Port Start Range for UDP packet transmission (valid range 1024–65,535; default 4000), and establish Preferred Codec Lists in MIME format. DTMF configuration options include specifying payload type values (96–127 range; default 101) and selecting between RTP Events for out-of-band processing per RFC 4733 or In-band Audio for codec stream detection and synthesis, with RTP Events as the default method. Network provisioning settings enable phones to obtain configuration data from Edges or third-party URIs with TLS Authority ID specification for secure connections. Signaling configuration includes DSCP value selection for SIP packet Quality of Service, with values ranging from 00 to 3F in IP header TOS fields. The Custom section permits Genesys Cloud Product Support personnel to modify configurations for troubleshooting using property name, data type, and value parameters. Base line appearance configuration requires Key Label entry and customization across General, Signaling, and Interface sections. General settings control calls per line capacity and persistent connection features with configurable timeout values. Signaling settings specify SIP protocol selection (UDP, TCP, or TLS), local listen port, registration period, maximum bindings, and SIP server assignments, with first line appearance settings locked to ensure Edge routing. Administrators can configure call control and outbound request routing through the Edge or custom SIP and proxy servers with prioritized ordering. Digest authentication supports administrative access control to physical phone menus with toggleable password visibility. UDP settings include configurable T1 and T2 timeout values for packet retransmission and maximum retry limits for SIP messages and INVITE requests. Auto-Conference functionality requires PIN entry for conference access with multiple language options for voice prompts. Recording capabilities include toggleable call recording with automatic level control, selectable audio codecs (G726-32, GSML, Opus, PCMA, PCMU, Truespech, and L16), and dual-channel recording that separates local and remote audio streams, with recording and auto-conference defaulting to disabled and dual-channel recording defaulting to enabled for supported formats. After establishing baseline settings through the Save Base Settings function, users can create a PureCloud softphone. The platform distinguishes PureCloud Softphone configuration from Genesys Cloud WebRTC phones and provides access to related resources including softphone s, managed phone configuration, WebRTC functionality, phone trunk creation, and migration guidance for transitioning to generic SIP softphone alternatives. Community support channels are available for users requiring additional assistance

PrerequisitesClick to expand

Before you can create a PureCloud softphone, you create a base settings configuration. The base settings configuration contains a group of settings found on the Base Phone and Base Line Appearance tabs that define how a PureCloud softphone is to operate in Genesys Cloud. Once you create a base settings configuration, you can save it with the default settings or you can customize the settings.

Note: The PureCloud Softphone is different from the Genesys Cloud WebRTC phone. For more information, see .

Create the base settings

Click Admin.
Under Telephony, click Phone Management.
Click Menu > Digital and Telephony > Telephony > Phone Management.
Click the Base Settings tab.
Click Add Base Settings.
Type a name in the Base Settings Name box.
From the Phone Make and Model list, select PureCloud Softphone.
Choose one of the following steps:
- To use the default base phone settings, click Save Base Settings and proceed to the Configure the base line appearance section of this article.
- To customize the base phone settings, proceed to the Customize the base phone section of this article.

Customize the base settings

In the Phone Configuration section of the Base Phone tab, there are four expandable sections: General, Media, Network, and Custom that allow you to change the base settings for the Genesys Cloud WebRTC phone.

To customize the base settings:

Select the appropriate settings from each expandable section.

GeneralClick to expand

Setting	Description
Dynamic Reload	Enable the dynamic reloading of the phone configuration.

MediaClick to expand

Setting	Description
DSCP	Use the list to choose the Differentiated Services Code Point (DSCP) value of Quality of Service (QoS) for RTP packets. The system places this value in the upper 6 bits of the TOS (Type Of Service) field. The TOS field is in the IP header of every RTP packet. The range of values available is 00 (0,000000) through 3F (63, 111111).
RTP Audio Port Start Range	Defines the UDP port of the remote computer where the system sends the recorded packets. The valid range is 1024–65,535. The default port is 4000.
Preferred Codec List	Use the list to select and build a list of preferred media codecs in mime format. Use the arrows next to the codec name to change the order in which the codecs in the list are used.
DTMF Settings
DTMF Payload	Specify the payload type value to use when the DTMF Method type is set to RTP Events. Valid range is 96–127. Valid only when DTMF Method value is set to RTP Events. The default value is 101.
DTMF Method	Use the list to select the method to use to transmit dual-tone multifrequency (DTMF) signaling. Select RTP Events to enable out-of-band processing of events from the RTP stream (RFC 4733). Select In-band Audio for the processing, detection, and synthesis from the audio codec stream. The default value is RTP Events.

NetworkClick to expand

Setting	Description
Provisioning	Use the Provision Source list to select the source that the phone uses to obtain the provision configuration data.
From Edges within the Site	Configure the phone to obtain provision configuration data from an Edge on your site.
From a third party URI	Configure the phone to obtain provision configuration data from a third-party URI.
Provisioning Third Party URI	When you select From a third party URI in the Provision list, this box is enabled. Enter the URI to the third-party resource that the phone uses to obtain provision configuration data.
TLS Authority ID	Specify the trusted certificate authority to validate connections when using TLS.
Signaling
DSCP	Use the list to choose the Differentiated Services Code Point (DSCP) value of Quality of Service (QoS) for SIP packets. The system places this value in the upper 6 bits of the TOS (Type Of Service) field. The TOS field is in the IP header of every SIP packet. The range of values available is 00 (0,000000) through 3F (63, 111111).

CustomClick to expand

The Custom option is designed to allow Genesys Cloud Product Support personnel to alter a phone configuration for troubleshooting or special circumstances. You should only enter custom property settings as directed by Genesys Cloud Product Support.

Setting	Description
Property Name	The name to assign to the custom property.
Data Type	The data type for the custom property.
Value	The value to assign the custom property.

To use the custom base settings, click Save Base Settings and proceed to the Configure the base line appearance section of this article.

Configure the base line appearance

Click the Base Line Appearance tab.
Type a name in the Key Label box.
Choose one of the following steps:
- To use the default base line appearance settings, click Save Base Settings. You can now Create a PureCloud softphone.
- To customize the base line appearance settings, proceed to the Customize the base line appearance section of this article.

Customize the base line appearance

In the Configuration setion of the Base Line Appearance tab, three expandable sections appear: General, Signaling, and Interface. Each expandable section contains controls that customize the settings for the PureCloud Softphone.

To customize the base line appearance settings:

Select the appropriate settings from each expandable section.

GeneralClick to expand

Setting	Description
Calls Per Line	The number of ACD calls that this line can handle.
Persistent Connection Settings	When the persistent connection feature is disabled, Genesys Cloud must create a connection for every call. When you enable the persistent connection feature and set a timeout value, you improve Genesys Cloud’s ability to process subsequent calls. More specifically, calls that come in while the connection is still active are immediately alerted via the UI or are auto answered if the Auto Answer feature is configured for the user.
Enable Persistent Connection	Disabled (Default): Do not use persistent connection feature. Enabled: Turn on persistent connection feature.
Persistent Connection Timeout	Sets the amount of time, in seconds, that the open connection can remain idle before being automatically closed.

SignalingClick to expand

Setting	Description
Protocol	Use the list to select the SIP protocol the phone uses to register: UDP, TCP, or TLS. The default is UDP.
Listen Port	Defines the local SIP listen port for SIP messages. The listening ports are the network ports on which the station expects to receive messages from SIP peers. While you can enter a custom port number, each protocol has a default listen port.
Registration period	The periodic delay (in seconds) between sending a SIP REGISTER).
Max Bindings	Specifies the maximum number of bindings.
SIP Servers or Proxies	The SIP Servers or Proxies setting is not configurable for the first line appearance to ensure that Genesys Cloud sends all outbound requests to the Edge. More specifically, this default configuration for the first line appearance ensures that the phone uses the Edge for the default call control. As long as you did not enable the Span appearance to remaining keys setting, you can choose to define where Genesys Cloud sends outbound requests for the remaining Line Keys. You can choose the Edge or specify custom SIP server or proxy server addresses and ports.
Use Edge	Send all outbound requests to the Edge.
Use the following	Use the Hostname or IP Address and Port fields to construct a prioritized list of SIP servers or proxy servers to use to process outbound requests. Use the + to add the server to the list. Use the arrows next to the address name to change the order in which the servers in the list are used.
Digest Authentication	When outbound requests are challenged with digest authentication, use the following credentials:
User Name	The user name pushed down to the phone that changes the password on the administrative account for the physical phone menus.
Password	The password pushed down to the phone that changes the password on the administrative account for the physical phone menus. By default the password is masked, but you can select the Show Password check box to see the password in plain-text.
UDP Settings
UDP T1 timeout	The timer value that represents the initial incremental delay between UDP packet retransmission.
UDP T2 timeout	The timer value that represents the maximum incremental delay between packet retransmissions.
Max Packet Retry	Maximum number of times to retransmit a SIP message over UDP.
Max Invite Retry	Maximum number of times to retransmit a SIP INVITE request over UDP.

InterfaceClick to expand

Setting	Description
Auto-Conference Settings
Enable Auto-Conference	Off (Default): Do not use auto-dial settings. On: Use the following fields to specify the auto-dial settings.
Auto-Conference PIN	Number required to join a conference.
Language	Speech selected to use for the conference voice menu.