Create a PureCloud softphone

AI-generated summary

Genesys Cloud softphones are customized through comprehensive base and phone profiles that configure codecs, DTMF methods, TLS authorities, and Quality of Service settings to optimize call performance. Administrators create phone profiles by accessing Phone Management, entering the phone name, selecting base settings and site, and providing the PC's fully qualified domain name (FQDN) in the Hardware ID field. The Phone Configuration panel enables extensive customization of General settings including Dynamic Reload functionality and Media settings such as DSCP values for RTP packet Quality of Service, with RTP Audio Port Start Range configurable between 1024–65,535 (default 4000) and Preferred Codec List adjustable in MIME format with priority ordering. DTMF signaling operates through two methods: RTP Events (out-of-band, RFC 4733, default) or In-band Audio processing, with RTP Events payload specifications valid in the range 96–127. Line Keys support persistent connections, SIP server bindings, and digest authentication with configurable Calls Per Line capacity and connection reuse features that improve call processing efficiency through adjustable timeout values. Network provisioning sources configuration data from site Edges or third-party URIs, with TLS Authority ID specifying trusted authorities directory paths and signaling DSCP QoS values (00-3F range) applied to SIP packets in the IP header's TOS field. Signaling protocol options include UDP (default), TCP, or TLS registration with configurable Listen Port for SIP messages, Registration Period intervals, Max Bindings limits, and SIP Servers/Proxies settings that route outbound requests through the Edge. Genesys Cloud enables flexible outbound request routing for phone line keys, allowing administrators to direct traffic either through the Edge or to custom SIP/proxy servers with prioritized address lists. Digest authentication uses credentials controlling administrative phone access with password masking available, while UDP settings configure SIP message retransmission parameters including T1/T2 timeout values and maximum retry attempts. The Media section provides call recording capabilities with configurable audio codecs (G726-32, GSML16, Opus, PCMA, PCMU, and Truespeech), automatic level control for balanced recording volumes, and dual-channel recording options that capture the phone connection on audio channel 0 and the other party's connection on audio channel 1 by default. Additional customization options include Auto-Conference functionality requiring configurable PINs with selectable voice menu languages, language preferences, and custom properties enabling Genesys Cloud Product Support to modify settings for troubleshooting purposes. Users finalize PureCloud softphone configuration by clicking the Save Phone button and can access related resources including managed phone configuration, WebRTC phone trunk creation, and migration options from PureCloud Softphone to generic SIP or Genesys Cloud WebRTC phones.

PrerequisitesClick to expand

Once you have created an PureCloud softphone base settings profile you can create the phone profile. This profile contains a group of settings found on the Phone and Line Keys tabs that define the phone in Genesys Cloud. The majority of the settings on the Phone and Line Keys tabs are inherited from the base settings. However, you can customize the phone by altering any of the settings that it inherited from the base settings configuration-without effecting the original base settings configuration. See Inherited settings.

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

Configure the phone

Click Admin.
Under Telephony, click Phone Management.
Click Menu > Digital and Telephony > Telephony > Phone Management.
Click the Phones tab.
Click Add Phone and the Phone tab appears.
Type a name in the Phone Name box.
From the Base Settings list, select the base settings configuration you created.
From the Site list, select your Site.

Note: If you are using BYOC Cloud or Genesys Cloud Voice, do not use the PureCloud Voice – AWS site. Make sure that you select one of the sites that you created.

Type the PC’s fully qualified domain name (FQDN) in the Hardware Id field.

Locate the FQDN on a Windows computerClick to expand

Perform one of the following:
- To use the default phone configuration settings, click Save Phone and proceed to the Configure the line keys section of this article.
- To customize the phone configuration settings, proceed to the Customize the phone section of this article.

Customize the phone

In the Phone Configuration panel, click the arrow to expand the section containing the settings you wish to customize.

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
Provision	Use the Provision 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	The file path to the trusted authorities directory on the disk used for 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 your custom phone configuration settings, click Save Phone and proceed to the Configure the line keys section of this article.

Configure the line keys

Click theLine Keys tab.
Type a name in the Key Label box.
Type a name in the Line Appearance Id box.
Choose one of the following steps:
- To use the default line keys settings, click Save Phone.
- To customize the line keys settings, proceed to the Customize the line keys section of this article.

Customize the line keys

In the Configuration panel, click the arrow to expand the section containing the settings you wish to customize.

GeneralClick to expand

Setting	Description
Calls Per Line	The number of 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./td>

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.

CustomClick to expand

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.

MediaClick to expand

Setting	Description
Recording Settings
Record calls on this phone	Use this check box to enable or disable recording. The default setting is disabled.
Level the volume on both sides of the conversation	Use this check box to enable or disable automatic level control for the recordings. The default setting is disabled.
Audio Format/Codec	Use this list to select the audio codec to use for recording. The available choices are: G726-32 GSM L16 (Uncompressed Linear 16-bit PCM) Opus PCMA (G.711 A-law) PCMU (G.711 u-law) Truespeech
Dual channel	Use this check box to enable or disable dual channel recording. The Dual channel setting is only available if you select one of the following audio formats: PCMU (G.711 u-law) PCMA (G.711 A-law) L16 (Uncompressed Linear 16-bit PCM) Opus When you enable the Dual channel setting, the system saves each channel of the recording in a separate stream. The system uses audio channel 0 to save the recording connected to the phone and audio channel 1 to save the recording connected to the other party. The default setting is enabled.

To use your custom line key settings and complete the PureCloud softphone configuration procedure, click Save Phone.

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Create a PureCloud softphone

AI-generated summary

PrerequisitesClick to expand

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

Configure the phone

Click Admin.
Under Telephony, click Phone Management.
Click Menu > Digital and Telephony > Telephony > Phone Management.
Click the Phones tab.
Click Add Phone and the Phone tab appears.
Type a name in the Phone Name box.
From the Base Settings list, select the base settings configuration you created.
From the Site list, select your Site.

Note: If you are using BYOC Cloud or Genesys Cloud Voice, do not use the PureCloud Voice – AWS site. Make sure that you select one of the sites that you created.

Type the PC’s fully qualified domain name (FQDN) in the Hardware Id field.

Locate the FQDN on a Windows computerClick to expand

Perform one of the following:
- To use the default phone configuration settings, click Save Phone and proceed to the Configure the line keys section of this article.
- To customize the phone configuration settings, proceed to the Customize the phone section of this article.

Customize the phone

In the Phone Configuration panel, click the arrow to expand the section containing the settings you wish to customize.

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
Provision	Use the Provision 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	The file path to the trusted authorities directory on the disk used for 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

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 your custom phone configuration settings, click Save Phone and proceed to the Configure the line keys section of this article.

Configure the line keys

Click theLine Keys tab.
Type a name in the Key Label box.
Type a name in the Line Appearance Id box.
Choose one of the following steps:
- To use the default line keys settings, click Save Phone.
- To customize the line keys settings, proceed to the Customize the line keys section of this article.

Customize the line keys

In the Configuration panel, click the arrow to expand the section containing the settings you wish to customize.

GeneralClick to expand

Setting	Description
Calls Per Line	The number of 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./td>

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.

CustomClick to expand

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.

MediaClick to expand

Setting	Description
Recording Settings
Record calls on this phone	Use this check box to enable or disable recording. The default setting is disabled.
Level the volume on both sides of the conversation	Use this check box to enable or disable automatic level control for the recordings. The default setting is disabled.
Audio Format/Codec	Use this list to select the audio codec to use for recording. The available choices are: G726-32 GSM L16 (Uncompressed Linear 16-bit PCM) Opus PCMA (G.711 A-law) PCMU (G.711 u-law) Truespeech
Dual channel	Use this check box to enable or disable dual channel recording. The Dual channel setting is only available if you select one of the following audio formats: PCMU (G.711 u-law) PCMA (G.711 A-law) L16 (Uncompressed Linear 16-bit PCM) Opus When you enable the Dual channel setting, the system saves each channel of the recording in a separate stream. The system uses audio channel 0 to save the recording connected to the phone and audio channel 1 to save the recording connected to the other party. The default setting is enabled.

To use your custom line key settings and complete the PureCloud softphone configuration procedure, click Save Phone.

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Get user feedback about articles.

Create a PureCloud softphone

Configure the phone

Customize the phone

Configure the line keys

Customize the line keys

[NEXT] Was this article helpful?

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Create a PureCloud softphone

Configure the phone

Customize the phone

Configure the line keys

Customize the line keys

[NEXT] Was this article helpful?