WebRTC Phone Management

What Is a WebRTC Phone?

A Genesys Cloud WebRTC phone is a browser or desktop app-based softphone that lets users place and receive calls directly in the Genesys Cloud client — no physical desk phone required. It is the most common phone type for contact center agents, remote users, and fast deployments.

Provisioning Model

WebRTC phones are provisioned in two steps:

Step 1: Create Base Settings
        ↓
Step 2: Create the Phone object

Base Settings is a shared configuration profile that defines how the WebRTC phone behaves. The Phone is the individual user-assigned record that uses those settings. Always build Base Settings first.

Task	Path
Open Phone Management	`Admin → Telephony → Phone Management`
Create WebRTC Base Settings	Phone Management → Base Settings tab → Add
Create WebRTC Phone	Phone Management → Phones tab → Create New
Configure global WebRTC behavior	`Admin → Telephony → Global Telephony Settings`
Configure site media behavior	`Admin → Telephony → Sites → [Site] → General`
User selects WebRTC phone	Genesys Cloud client → Calls panel → phone selector

Required permission: Telephony > Plugin > All

Step 1: Create Base Settings

Step	Action
Step 1	Navigate to `Admin → Telephony → Phone Management`
Step 2	Open the Base Settings tab
Step 3	Click Add
Step 4	Enter a Base Settings Name
Step 5	In Phone Make and Model, select `Genesys Cloud WebRTC Phone`
Step 6	Enable Persistent Connection if needed
Step 7	Configure Transport DSCP Value and Media DSCP Value
Step 8	Click Save Base Settings

Base Settings Fields

Field	Description	Notes
Base Settings Name	Name for this configuration profile	Use a descriptive name e.g. `Support_WebRTC_Standard`
Phone Make and Model	Select the phone type	Choose `Genesys Cloud WebRTC Phone`
Persistent Connection	Keeps the WebRTC connection open after calls end	Improves subsequent call handling speed
Persistent Connection Timeout	How long the connection stays active	Configure based on call volume patterns
Transport DSCP Value	QoS marking for SIP signaling traffic	Align with enterprise voice network policy
Media DSCP Value	QoS marking for audio/media traffic	Align with enterprise voice network policy

Step 2: Create the Phone

Step	Action
Step 1	In Phone Management, open the Phones tab
Step 2	Click Create New
Step 3	Enter the Phone Name
Step 4	Select the Site
Step 5	Select the Base Settings profile created in Step 1
Step 6	Assign the User
Step 7	Click Save
Step 8	Have the user select the WebRTC phone in the client and test calling

Persistent Connection

Keeping the WebRTC connection open after a call ends allows subsequent calls to alert faster because the connection is already established.

Setting	Behaviour
Disabled	Connection closes after each call; next call requires fresh connection setup
Enabled	Connection stays open for the timeout period; subsequent calls alert immediately

⚠️ Important: If you enable Persistent Connection after users are already logged in, they must log out and back in for the setting to apply. Genesys recommends making this change outside business hours.

QoS / DSCP

DSCP values mark WebRTC SIP and media traffic so your network can prioritize it. Set these values to match your organization's enterprise voice QoS policy.

DSCP Field	Applies To
Transport DSCP	SIP signaling traffic
Media DSCP	Audio/RTP media traffic

Site-Level WebRTC Media Settings

These are configured on the Site, not in Base Settings. Validate these for every site that will host WebRTC users.

Field	Options	Description
Media Regions	Available Home/Core/Satellite regions	Select and prioritize regions for WebRTC / Global Media Fabric
Relay/TURN Behavior	Any media region set on this site / Lowest latency via Geo-Lookup	Controls how Genesys selects TURN relay regions for WebRTC calls that need relay services

Relay/TURN Option	Best For
Any media region set on this site	Strict control — limits TURN relay to configured regions only
Lowest latency via Geo-Lookup	Best performance — Genesys dynamically selects lowest-latency TURN region

⚠️ Forcing TURN relay can reduce resiliency and force RTP through relay services when not otherwise necessary.

User Experience

Users select and manage the WebRTC phone from the Calls panel in the Genesys Cloud client:

Choose microphone and speaker device

Adjust volume

Run audio diagnostics

Troubleshooting

Issue	Cause	Resolution
User cannot answer calls reliably	Persistent connection disabled or not applied	Enable it; have user log out and back in
No audio	Wrong microphone/speaker selected	Verify audio devices in WebRTC phone client settings
Poor call quality	DSCP/network/headset issue	Check QoS policy, headset, and local network
WebRTC phone not visible to user	Phone not created or not assigned to correct user	Recheck phone assignment
Calls do not route	Site/routing configuration issue	Validate site, number plans, and trunks
Settings change not applied	User session retained old settings	Log out and back in
Unexpected TURN/media path	Site Media Regions or Relay/TURN Behavior misconfigured	Review assigned Site's General tab

Quick Reference

Question	Answer
How do you add a WebRTC phone?	Create Base Settings first, then create the Phone
What model do you select?	`Genesys Cloud WebRTC Phone`
Why enable Persistent Connection?	Improves subsequent call handling speed
Where is Relay/TURN Behavior configured?	On the Site, not in Base Settings
What should users do after Persistent Connection is enabled later?	Log out and back in