Configure the Genesys Cloud WebRTC phone

AI-generated summary

Genesys Cloud WebRTC phone is a comprehensive communication platform that requires a two-step configuration process involving base settings and individual phone customization, with one WebRTC phone permitted per user. The system operates through four customization sections—General, Media, Network, and Custom—that allow individual phone modifications while inheriting base configurations, with changes taking effect after user logout and login. The General section provides persistent connection functionality with configurable timeouts ranging from 600 to 604,800 seconds to improve call processing efficiency and enable immediate alerting or auto-answering capabilities. The Media section offers dual-channel call recording, multiple audio codec options (PCMU, PCMA, L16, Opus, G726-32, GSM, Truespeech), automatic volume leveling, and Idle Media recovery with configurable reconnection attempts for handling stream disruption. Network management includes TURN Behavior for firewall control through reserved addresses via an automatically scaling fleet, and DSCP configuration (00-3F range) for Quality of Service management in RTP and SIP packets. The Attempt recovery setting enables automatic media stream recovery but cannot function simultaneously with persistent connection settings. WebRTC Media Helper supports VDI environments with automatic failback mechanisms and integrates with Microsoft Multimedia Redirection for Azure Virtual Desktop and Windows 365 Cloud PC. The Require WebRTC Media Helper setting controls failover behavior—when disabled (default), the system allows failback to VDI streaming upon disconnection; when enabled, it prevents failback but causes subsequent calls to fail until re-authentication. WebRTC functionality exclusively uses the Opus codec. Configuration requires Telephony > Plugin > All permissions. When used with Genesys Cloud Voice, only phone settings require configuration as the external trunk is enabled by default. Custom properties are reserved for Genesys Cloud Product Support personnel to address troubleshooting or special circumstances. The platform includes a feedback mechanism for documentation improvement and supports WebRTC phone configuration import via CSV files, with related documentation covering WebRTC fundamentals, media helper functionality, and VDI deployment scenarios.

PrerequisitesClick to expand

The process to configure a Genesys Cloud WebRTC phone is a two-step operation. First, create and configure the base settings. Second, create and configure the phone.

If the base settings for the Genesys Cloud WebRTC phone are already configured, you only need to create and configure the phone.

Notes:

When you configure a Genesys Cloud WebRTC phone to use under Genesys Cloud Voice, keep in mind that the Genesys Cloud Voice configuration enables the external trunk by default. So, you only need to create the base settings and phone settings for your Genesys Cloud WebRTC phones.
You can only create one WebRTC phone per user.

Create the base settings

The base settings configuration contains a group of settings that define how the Genesys Cloud WebRTC phone operates in Genesys Cloud. After you create a base settings configuration, save it with the default settings or customize the settings.

To create the base settings:

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

Customize the base settings

The Phone Configuration section of the Base Phone tab contains four expandable sections: General, Media, Network, and Custom. These sections 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
Persistent Connection	When the persistent connection setting is disabled, Genesys Cloud creates a connection for every call. When you enable the persistent connection setting and configure 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 if the Auto Answer setting is configured for the user, are auto answered. Note: Genesys recommends that you enable the persistent connection setting for WebRTC phones outside normal business hours to ensure proper configuration. If you enable the persistent connection setting for WebRTC phones, either in the base settings or on individual phones, after the agents that use those phones are logged in to Genesys Cloud, their phones do not receive the persistent connection setting unless they log out and then log back in. For more information, see Use the persistent connection feature with a Genesys Cloud WebRTC phone.
Maintain a persistent connection	Disabled (Default): Do not use persistent connection setting. Enabled: Turn on persistent connection setting.
Timeout	Set the amount of time, in seconds, that the open connection remains idle before it automatically closes. The default timeout setting is 600 seconds, or 10 minutes. The maximum timeout setting is 604,800 seconds, or seven days.

Setting

Description

Persistent Connection

When the persistent connection setting is disabled, Genesys Cloud creates a connection for every call.

When you enable the persistent connection setting and configure 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 if the Auto Answer setting is configured for the user, are auto answered.

Note: Genesys recommends that you enable the persistent connection setting for WebRTC phones outside normal business hours to ensure proper configuration. If you enable the persistent connection setting for WebRTC phones, either in the base settings or on individual phones, after the agents that use those phones are logged in to Genesys Cloud, their phones do not receive the persistent connection setting unless they log out and then log back in.

For more information, see Use the persistent connection feature with a Genesys Cloud WebRTC phone.

Maintain a persistent connection

Disabled (Default): Do not use persistent connection setting.

Enabled: Turn on persistent connection setting.

Timeout

Set the amount of time, in seconds, that the open connection remains idle before it automatically closes.

The default timeout setting is 600 seconds, or 10 minutes. The maximum timeout setting is 604,800 seconds, or seven days.

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.
Idle Media	Use the Idle Media feature to specify how long the media stream remains idle before attempting to reconnect. When a WebRTC client unexpectedly loses the media stream for an active call, Genesys Cloud immediately disconnects the call. To prevent the immediate disconnect, enable the Attempt recovery setting and specify a timeout value for recovery. Genesys Cloud then attempts to recover the media stream when the timeout value is reached. Note: If you enable the Attempt recovery setting and also have the persistent connection setting enabled, keep in mind that the persistent connection will not be able to automatically reconnect.
DSCP	From the list, choose the Differentiated Services Code Point (DSCP) value of Quality of Service (QoS) for RTP packets. The system places this value in the upper six bits of the TOS (Type Of Service) field. The TOS field is in the IP header of every RTP packet. The range of available values is 00 (0,000000) through 3F (63, 111111).
Preferred Codec List	WebRTC only works with the Opus codec.
Require WebRTC Media Helper	Note: If you are using Microsoft’s Multimedia Redirection tool with Azure Virtual Desktop or Windows 365 Cloud PC, then the Require WebRTC Media Helper setting must be set to Disabled. If the WebRTC Media Helper connection inadvertently disconnects or closes, then the WebRTC Media Helper fails back to streaming through VDI by design. This failback method ensures that an agent can continue to answer subsequent calls. However, in some VDI configurations, this failback method can degrade audio quality or cause issues with the VDI infrastructure. If you encounter issues with the WebRTC Media Helper failback method, then to prevent the failback operation from occurring, enable the Require WebRTC Media Helper setting. Disabled (Default): Do not use Require WebRTC Media Helper setting. Enabled: Turn on the Require WebRTC Media Helper setting. For more information, see Require WebRTC Media Helper. Warning: When you enable the Require WebRTC Media Helper setting on a WebRTC station, any disconnect causes subsequent inbound and outbound calls to fail. The result is a Media helper not connected error. To re-establish the WebRTC Media Helper connection, agents must access the Genesys Cloud WebRTC Media Helper app for your region and log in. For more information, see Use WebRTC Media Helper in a Virtual Desktop Infrastructure (VDI) environment.

NetworkClick to expand

Setting	Description
Signaling Differentiated Services Code Point (DSCP)	From the list, choose the value of Quality of Service (QoS) for SIP packets. The system places this value in the upper six bits of the TOS (Type Of Service) field. The TOS field is in the IP header of every SIP packet. The range of available values is 00 (0,000000) through 3F (63, 111111).

Setting

Description

Signaling

Differentiated Services Code Point (DSCP)

From the list, choose the value of Quality of Service (QoS) for SIP packets.

The system places this value in the upper six bits of the TOS (Type Of Service) field. The TOS field is in the IP header of every SIP packet. The range of available values is 00 (0,000000) through 3F (63, 111111).

CustomClick to expand

By design, the Custom option allows Genesys Cloud Product Support personnel to alter phone configuration for troubleshooting or special circumstances. 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.

Click Save Base Settings.
Proceed to the Create the phone section of this article.

Create the phone

Once you create, configure, and save the base settings for the Genesys Cloud WebRTC phone, create the phone and assign it to a user.

To create the phone, follow these steps:

Select the Phones tab.
Click Add Phone. The Phone tab appears.
In the Phone Name box, type a name.
From the Base Settings list, select the base setting configuration that you created for your Genesys Cloud WebRTC phone.
From the Site list, select your site.

Note: If you use 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.

From the Person list, select the name of the person to whom you want to assign this Genesys Cloud WebRTC phone.
Choose one of the following steps:
- To use the default phone settings, click Save Phone.
- To customize the phone settings, proceed to the Customize the phone section of this article.

Customize the phone

In the Phone Configuration section of the Phone tab, four expandable sections appear: General, Media, Network, and Custom. Each expandable section contains controls that customize the settings for the Genesys Cloud WebRTC phone. These settings are inherited from the base settings. However, you can customize a particular phone by altering any setting that it inherits from the base settings configuration without affecting the original base settings configuration. For more information, see Inherited settings.

To customize the phone:

Select the appropriate settings from each expandable section.

GeneralClick to expand

Setting	Description
Persistent Connection Settings	When the persistent connection setting 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 arrive while the connection is still active immediately alert via the UI or if the Auto Answer feature is configured for the user, are automatically answered. Note: Genesys recommends that you enable the persistent connection feature for WebRTC phones outside normal business hours to ensure proper configuration. If you enable the persistent connection feature for WebRTC phones, either in the base settings or on individual phones, agents that use those phones and are already logged in to Genesys Cloud will not receive the persistent connection feature unless they log out and then log back in. For more information, see Use the persistent connection feature with a Genesys Cloud WebRTC phone.
Maintain a persistent connection	Disabled (Default): Do not use the persistent connection feature. Enabled: Turn on persistent connection feature.
Timeout	Set the amount of time, in seconds, that the open connection remains idle before being automatically closed. The default timeout setting is 600 seconds, or 10 minutes. The maximum timeout setting is 604,800 seconds, or seven days.

Setting

Description

Persistent Connection Settings

When the persistent connection setting 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 arrive while the connection is still active immediately alert via the UI or if the Auto Answer feature is configured for the user, are automatically answered.

Note: Genesys recommends that you enable the persistent connection feature for WebRTC phones outside normal business hours to ensure proper configuration. If you enable the persistent connection feature for WebRTC phones, either in the base settings or on individual phones, agents that use those phones and are already logged in to Genesys Cloud will not receive the persistent connection feature unless they log out and then log back in.

For more information, see Use the persistent connection feature with a Genesys Cloud WebRTC phone.

Maintain a persistent connection

Disabled (Default): Do not use the persistent connection feature.

Enabled: Turn on persistent connection feature.

Timeout

Set the amount of time, in seconds, that the open connection remains idle before being automatically closed.

The default timeout setting is 600 seconds, or 10 minutes. The maximum timeout setting is 604,800 seconds, or seven days.

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.
TURN Behavior	The TURN Behavior feature reduces the number of open outbound ports on your firewall. When you use the TURN Behavior feature, an automatically scaling fleet of TURN service instances takes over and routes your outbound connections through a limited pool of reserved addresses. You must add the free TURN Behavior feature to your Genesys Cloud subscription. For more information, see Use the TURN Behavior feature. Note: After you enable the TURN Behavior feature and before agent WebRTC phones recognize the TURN Behavior feature, make sure you instruct agents using WebRTC phones to log out, and then log back in to Genesys Cloud.
TURN Behavior for WebRTC calls	Only use TURN when necessary (Default setting.) Always use TURN (Forces the use of TURN.)
Idle Media	Use the Idle Media feature to specify how long the media stream remains idle before attempting to reconnect. When a WebRTC client unexpectedly loses the media stream for an active call, Genesys Cloud immediately disconnects the call. To prevent the immediate disconnect, enable the Attempt recovery setting and specify a timeout value for recovery. Genesys Cloud then attempts to recover the media stream when the timeout value is reached. Note: If you enable the Attempt recovery setting and also have the persistent connection setting enabled, keep in mind that the persistent connection will not be able to automatically reconnect.
DSCP	From the list, 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 available values is 00 (0,000000) through 3F (63, 111111).
Preferred Codec List	WebRTC only works with the Opus codec.
Require WebRTC Media Helper	Note: If you are using Microsoft’s Multimedia Redirection tool with Azure Virtual Desktop or Windows 365 Cloud PC, then the Require WebRTC Media Helper setting must be set to Disabled. If the WebRTC Media Helper connection inadvertently disconnects or closes, then the WebRTC Media Helper fails back to streaming through VDI by design. This failback method ensures that an agent can continue to answer subsequent calls. However, in some VDI configurations, this failback method can degrade audio quality or cause issues with the VDI infrastructure. If you encounter issues with the WebRTC Media Helper failback method, then to prevent the failback operation from occurring, enable the Require WebRTC Media Helper setting. Disabled (Default): Do not use the Require WebRTC Media Helper setting. Enabled: Turn on the Require WebRTC Media Helper setting. For more information, see Require WebRTC Media Helper. Warning: When you enable the Require WebRTC Media Helper setting on a WebRTC station, any disconnect causes subsequent inbound and outbound calls to fail. The result is a Media helper not connected error. To re-establish the WebRTC Media Helper connection, agents must access the Genesys Cloud WebRTC Media Helper app for your region and log in. For more information, see Use WebRTC Media Helper in a Virtual Desktop Infrastructure (VDI) environment.

NetworkClick to expand

Setting	Description
Signaling Differentiated Services Code Point (DSCP)	Use the list to choose the value of Quality of Service (QoS) for SIP packets. The system places this value in the upper six bits of the TOS (Type Of Service) field. The TOS field is in the IP header of every SIP packet. The range of available values is 00 (0,000000) through 3F (63, 111111).

Setting

Description

Signaling

Differentiated Services Code Point (DSCP)

Use the list to choose the value of Quality of Service (QoS) for SIP packets.

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.

Click Save Phone.

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Configure the Genesys Cloud WebRTC phone

AI-generated summary

PrerequisitesClick to expand

The process to configure a Genesys Cloud WebRTC phone is a two-step operation. First, create and configure the base settings. Second, create and configure the phone.

If the base settings for the Genesys Cloud WebRTC phone are already configured, you only need to create and configure the phone.

Notes:

When you configure a Genesys Cloud WebRTC phone to use under Genesys Cloud Voice, keep in mind that the Genesys Cloud Voice configuration enables the external trunk by default. So, you only need to create the base settings and phone settings for your Genesys Cloud WebRTC phones.
You can only create one WebRTC phone per user.

Create the base settings

To create the base settings:

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

Customize the base settings

To customize the base settings:

Select the appropriate settings from each expandable section.

GeneralClick to expand

Setting	Description
Persistent Connection	When the persistent connection setting is disabled, Genesys Cloud creates a connection for every call. When you enable the persistent connection setting and configure 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 if the Auto Answer setting is configured for the user, are auto answered. Note: Genesys recommends that you enable the persistent connection setting for WebRTC phones outside normal business hours to ensure proper configuration. If you enable the persistent connection setting for WebRTC phones, either in the base settings or on individual phones, after the agents that use those phones are logged in to Genesys Cloud, their phones do not receive the persistent connection setting unless they log out and then log back in. For more information, see Use the persistent connection feature with a Genesys Cloud WebRTC phone.
Maintain a persistent connection	Disabled (Default): Do not use persistent connection setting. Enabled: Turn on persistent connection setting.
Timeout	Set the amount of time, in seconds, that the open connection remains idle before it automatically closes. The default timeout setting is 600 seconds, or 10 minutes. The maximum timeout setting is 604,800 seconds, or seven days.

Setting

Description

Persistent Connection

When the persistent connection setting is disabled, Genesys Cloud creates a connection for every call.

For more information, see Use the persistent connection feature with a Genesys Cloud WebRTC phone.

Maintain a persistent connection

Disabled (Default): Do not use persistent connection setting.

Enabled: Turn on persistent connection setting.

Timeout

Set the amount of time, in seconds, that the open connection remains idle before it automatically closes.

The default timeout setting is 600 seconds, or 10 minutes. The maximum timeout setting is 604,800 seconds, or seven days.

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.
Idle Media	Use the Idle Media feature to specify how long the media stream remains idle before attempting to reconnect. When a WebRTC client unexpectedly loses the media stream for an active call, Genesys Cloud immediately disconnects the call. To prevent the immediate disconnect, enable the Attempt recovery setting and specify a timeout value for recovery. Genesys Cloud then attempts to recover the media stream when the timeout value is reached. Note: If you enable the Attempt recovery setting and also have the persistent connection setting enabled, keep in mind that the persistent connection will not be able to automatically reconnect.
DSCP	From the list, choose the Differentiated Services Code Point (DSCP) value of Quality of Service (QoS) for RTP packets. The system places this value in the upper six bits of the TOS (Type Of Service) field. The TOS field is in the IP header of every RTP packet. The range of available values is 00 (0,000000) through 3F (63, 111111).
Preferred Codec List	WebRTC only works with the Opus codec.
Require WebRTC Media Helper	Note: If you are using Microsoft’s Multimedia Redirection tool with Azure Virtual Desktop or Windows 365 Cloud PC, then the Require WebRTC Media Helper setting must be set to Disabled. If the WebRTC Media Helper connection inadvertently disconnects or closes, then the WebRTC Media Helper fails back to streaming through VDI by design. This failback method ensures that an agent can continue to answer subsequent calls. However, in some VDI configurations, this failback method can degrade audio quality or cause issues with the VDI infrastructure. If you encounter issues with the WebRTC Media Helper failback method, then to prevent the failback operation from occurring, enable the Require WebRTC Media Helper setting. Disabled (Default): Do not use Require WebRTC Media Helper setting. Enabled: Turn on the Require WebRTC Media Helper setting. For more information, see Require WebRTC Media Helper. Warning: When you enable the Require WebRTC Media Helper setting on a WebRTC station, any disconnect causes subsequent inbound and outbound calls to fail. The result is a Media helper not connected error. To re-establish the WebRTC Media Helper connection, agents must access the Genesys Cloud WebRTC Media Helper app for your region and log in. For more information, see Use WebRTC Media Helper in a Virtual Desktop Infrastructure (VDI) environment.

NetworkClick to expand

Setting	Description
Signaling Differentiated Services Code Point (DSCP)	From the list, choose the value of Quality of Service (QoS) for SIP packets. The system places this value in the upper six bits of the TOS (Type Of Service) field. The TOS field is in the IP header of every SIP packet. The range of available values is 00 (0,000000) through 3F (63, 111111).

Setting

Description

Signaling

Differentiated Services Code Point (DSCP)

From the list, choose the value of Quality of Service (QoS) for SIP packets.

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.

Click Save Base Settings.
Proceed to the Create the phone section of this article.

Create the phone

Once you create, configure, and save the base settings for the Genesys Cloud WebRTC phone, create the phone and assign it to a user.

To create the phone, follow these steps:

Select the Phones tab.
Click Add Phone. The Phone tab appears.
In the Phone Name box, type a name.
From the Base Settings list, select the base setting configuration that you created for your Genesys Cloud WebRTC phone.
From the Site list, select your site.

Note: If you use 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.

From the Person list, select the name of the person to whom you want to assign this Genesys Cloud WebRTC phone.
Choose one of the following steps:
- To use the default phone settings, click Save Phone.
- To customize the phone settings, proceed to the Customize the phone section of this article.

Customize the phone

To customize the phone:

Select the appropriate settings from each expandable section.

GeneralClick to expand

Setting	Description
Persistent Connection Settings	When the persistent connection setting 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 arrive while the connection is still active immediately alert via the UI or if the Auto Answer feature is configured for the user, are automatically answered. Note: Genesys recommends that you enable the persistent connection feature for WebRTC phones outside normal business hours to ensure proper configuration. If you enable the persistent connection feature for WebRTC phones, either in the base settings or on individual phones, agents that use those phones and are already logged in to Genesys Cloud will not receive the persistent connection feature unless they log out and then log back in. For more information, see Use the persistent connection feature with a Genesys Cloud WebRTC phone.
Maintain a persistent connection	Disabled (Default): Do not use the persistent connection feature. Enabled: Turn on persistent connection feature.
Timeout	Set the amount of time, in seconds, that the open connection remains idle before being automatically closed. The default timeout setting is 600 seconds, or 10 minutes. The maximum timeout setting is 604,800 seconds, or seven days.

Setting

Description

Persistent Connection Settings

When the persistent connection setting is disabled, Genesys Cloud must create a connection for every call.

For more information, see Use the persistent connection feature with a Genesys Cloud WebRTC phone.

Maintain a persistent connection

Disabled (Default): Do not use the persistent connection feature.

Enabled: Turn on persistent connection feature.

Timeout

Set the amount of time, in seconds, that the open connection remains idle before being automatically closed.

The default timeout setting is 600 seconds, or 10 minutes. The maximum timeout setting is 604,800 seconds, or seven days.

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.
TURN Behavior	The TURN Behavior feature reduces the number of open outbound ports on your firewall. When you use the TURN Behavior feature, an automatically scaling fleet of TURN service instances takes over and routes your outbound connections through a limited pool of reserved addresses. You must add the free TURN Behavior feature to your Genesys Cloud subscription. For more information, see Use the TURN Behavior feature. Note: After you enable the TURN Behavior feature and before agent WebRTC phones recognize the TURN Behavior feature, make sure you instruct agents using WebRTC phones to log out, and then log back in to Genesys Cloud.
TURN Behavior for WebRTC calls	Only use TURN when necessary (Default setting.) Always use TURN (Forces the use of TURN.)
Idle Media	Use the Idle Media feature to specify how long the media stream remains idle before attempting to reconnect. When a WebRTC client unexpectedly loses the media stream for an active call, Genesys Cloud immediately disconnects the call. To prevent the immediate disconnect, enable the Attempt recovery setting and specify a timeout value for recovery. Genesys Cloud then attempts to recover the media stream when the timeout value is reached. Note: If you enable the Attempt recovery setting and also have the persistent connection setting enabled, keep in mind that the persistent connection will not be able to automatically reconnect.
DSCP	From the list, 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 available values is 00 (0,000000) through 3F (63, 111111).
Preferred Codec List	WebRTC only works with the Opus codec.
Require WebRTC Media Helper	Note: If you are using Microsoft’s Multimedia Redirection tool with Azure Virtual Desktop or Windows 365 Cloud PC, then the Require WebRTC Media Helper setting must be set to Disabled. If the WebRTC Media Helper connection inadvertently disconnects or closes, then the WebRTC Media Helper fails back to streaming through VDI by design. This failback method ensures that an agent can continue to answer subsequent calls. However, in some VDI configurations, this failback method can degrade audio quality or cause issues with the VDI infrastructure. If you encounter issues with the WebRTC Media Helper failback method, then to prevent the failback operation from occurring, enable the Require WebRTC Media Helper setting. Disabled (Default): Do not use the Require WebRTC Media Helper setting. Enabled: Turn on the Require WebRTC Media Helper setting. For more information, see Require WebRTC Media Helper. Warning: When you enable the Require WebRTC Media Helper setting on a WebRTC station, any disconnect causes subsequent inbound and outbound calls to fail. The result is a Media helper not connected error. To re-establish the WebRTC Media Helper connection, agents must access the Genesys Cloud WebRTC Media Helper app for your region and log in. For more information, see Use WebRTC Media Helper in a Virtual Desktop Infrastructure (VDI) environment.

NetworkClick to expand

Setting	Description
Signaling Differentiated Services Code Point (DSCP)	Use the list to choose the value of Quality of Service (QoS) for SIP packets. The system places this value in the upper six bits of the TOS (Type Of Service) field. The TOS field is in the IP header of every SIP packet. The range of available values is 00 (0,000000) through 3F (63, 111111).

Setting

Description

Signaling

Differentiated Services Code Point (DSCP)

Use the list to choose the value of Quality of Service (QoS) for SIP packets.

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.

Click Save Phone.

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Configure the Genesys Cloud WebRTC phone

Create the base settings

Customize the base settings

Create the phone

Customize the phone

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Configure the Genesys Cloud WebRTC phone

Create the base settings

Customize the base settings

Create the phone

Customize the phone

[NEXT] Was this article helpful?