# WebRTC Phone Management

**Navigation:** Admin → Telephony → Phone Management
**Last verified:** Genesys Cloud Resource Center — March 2026

---

## 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.

---

## Navigation

| 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

> ℹ️ Recommend using a quality headset rather than laptop built-in speakers and microphone to avoid echo and audio quality issues.

---

## 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 |

---

## Naming Convention

| Resource | Example |
|---|---|
| Base Settings | `Support_WebRTC_Standard` |
| Base Settings | `Sales_WebRTC_Remote` |
| Phone | `AgentName_WebRTC` |

Pattern: `<BusinessArea>_WebRTC_<Purpose>`

---

## See Also

- **Sites** — Media Regions and Relay/TURN Behavior are configured here
- **Topology** — confirm phone-to-edge assignments and site relationships
- **Architectural Build Order** — phones are assigned in Phase 3 (People)

---

## Screenshots

[![](https://wiki.tinod.net/uploads/images/gallery/2026-03/scaled-1680-/ewGnICTIbeARhwUP-image-1773118569449.png)](https://wiki.tinod.net/uploads/images/gallery/2026-03/ewGnICTIbeARhwUP-image-1773118569449.png)

[![](https://wiki.tinod.net/uploads/images/gallery/2026-03/scaled-1680-/JXcipyD2Go2k90dF-image-1773118604100.png)](https://wiki.tinod.net/uploads/images/gallery/2026-03/JXcipyD2Go2k90dF-image-1773118604100.png)

[![](https://wiki.tinod.net/uploads/images/gallery/2026-03/scaled-1680-/taDgKT1l74NeLLlm-image-1773118659336.png)](https://wiki.tinod.net/uploads/images/gallery/2026-03/taDgKT1l74NeLLlm-image-1773118659336.png)

[![](https://wiki.tinod.net/uploads/images/gallery/2026-03/scaled-1680-/v43pFMx9a4zxoxnB-image-1773118704788.png)](https://wiki.tinod.net/uploads/images/gallery/2026-03/v43pFMx9a4zxoxnB-image-1773118704788.png)

[![](https://wiki.tinod.net/uploads/images/gallery/2026-03/scaled-1680-/mRMr8CAgoKIcg4nA-image-1773119448802.png)](https://wiki.tinod.net/uploads/images/gallery/2026-03/mRMr8CAgoKIcg4nA-image-1773119448802.png)

[![](https://wiki.tinod.net/uploads/images/gallery/2026-03/scaled-1680-/iV9Eu1WfNwpBYsjb-image-1773119472778.png)](https://wiki.tinod.net/uploads/images/gallery/2026-03/iV9Eu1WfNwpBYsjb-image-1773119472778.png)

[![](https://wiki.tinod.net/uploads/images/gallery/2026-03/scaled-1680-/emvprxJAqLuQrMEl-image-1773119485640.png)](https://wiki.tinod.net/uploads/images/gallery/2026-03/emvprxJAqLuQrMEl-image-1773119485640.png)

[![](https://wiki.tinod.net/uploads/images/gallery/2026-03/scaled-1680-/sMn5aUzgOnWegZeA-image-1773119568820.png)](https://wiki.tinod.net/uploads/images/gallery/2026-03/sMn5aUzgOnWegZeA-image-1773119568820.png)

[![](https://wiki.tinod.net/uploads/images/gallery/2026-03/scaled-1680-/ORKzGQI0CkAcVqk2-image-1773119525331.png)](https://wiki.tinod.net/uploads/images/gallery/2026-03/ORKzGQI0CkAcVqk2-image-1773119525331.png)

[![](https://wiki.tinod.net/uploads/images/gallery/2026-03/scaled-1680-/3tFrWHvkQSLiz3bc-image-1773119604438.png)](https://wiki.tinod.net/uploads/images/gallery/2026-03/3tFrWHvkQSLiz3bc-image-1773119604438.png)

[![](https://wiki.tinod.net/uploads/images/gallery/2026-03/scaled-1680-/hUeF76ymDII8YIhF-image-1773119637506.png)](https://wiki.tinod.net/uploads/images/gallery/2026-03/hUeF76ymDII8YIhF-image-1773119637506.png)

Now Add a phone
[![](https://wiki.tinod.net/uploads/images/gallery/2026-03/scaled-1680-/un42QEAJCA8vxBEm-image-1773119656360.png)](https://wiki.tinod.net/uploads/images/gallery/2026-03/un42QEAJCA8vxBEm-image-1773119656360.png)