Configure the Genesys Cloud WebRTC phone
- Telephony > Plugin > All permission
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.
- 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 Admin.
- Under Telephony, click 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
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.
| Setting | Description |
|---|---|
|
Persistent Connection |
When the persistent connection setting is disabled, Genesys Cloud must create 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 |
|---|---|
| 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 |
|
| 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 |
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.
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.
|
| 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 |
|---|---|
| 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.
- 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.
| 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 immediate 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 of 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 persistent connection feature. Enabled: Turn on persistent connection feature. |
| Timeout |
Set the amount of time, in seconds, that the open connection can remain 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 |
|---|---|
| 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 |
|
| 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 |
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.
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.
|
| 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). |
The Custom option is designed to allow Genesys Cloud Customer Care personnel to alter a phone configuration for troubleshooting or special circumstances. You should only enter custom property settings as directed by Genesys Cloud Customer Care.
| 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.
