Voice, Video & Screen Share

Wiretalk supports voice notes in chat (all plans), plus live voice & video calls and screen sharing over secure WebRTC on Pro and Enterprise plans — no Zoom link required.

Voice notes in chat

Agents and visitors can record short voice messages directly in the chat composer. Voice notes are uploaded as audio attachments and play inline in the conversation.

• Tap the microphone icon in the chat input area
• Record, preview, and send — works on desktop and mobile browsers that support MediaRecorder
• Available on Free, Pro, and Enterprise plans

Browsers may ask for microphone permission the first time. If blocked, use the in-app help prompt to enable access in site settings.

Voice & video calls

Start a real-time call inside an active visitor chat. Calls use WebRTC peer-to-peer media with signaling through your Wiretalk socket server.

How to start a call

Plan availability: Pro and Enterprise only. Free plans do not include live voice/video calls.

Mobile: Voice and video calls work on modern mobile browsers (Chrome on Android, Safari on iOS 14.3+) over HTTPS. Allow microphone and camera permissions when prompted. For reliable calls on mobile data, configure a TURN server.

Screen sharing

Share your screen during a visitor chat so agents can see exactly what the customer sees — ideal for troubleshooting checkout, forms, or product UI issues.

• Click the screen-share icon in the chat composer (desktop browsers with getDisplayMedia support)
• Visitors can accept or decline an incoming share request
• Either side can stop sharing at any time
• Available on Pro and Enterprise

Screen sharing from mobile browsers is limited by the OS. Use desktop Chrome or Edge for the best experience.

Security

WebRTC signaling is protected end-to-end through Wiretalk’s socket layer:

• Signed socket tokens — only participants in the conversation room can send signals
• Verified identity — the server stamps each signal with the authenticated agent or visitor ID (no spoofing)
• Plan capability gating — free tokens cannot initiate WebRTC calls or screen share
• SDP sanitization — offer/answer payloads are validated and size-limited on the socket server
• Rate limiting — flood protection on signaling events

Media streams are peer-to-peer where possible. For strict corporate networks, configure a TURN server (see below).

TURN / STUN configuration

Wiretalk uses Google STUN servers by default. For production reliability (especially behind symmetric NAT or corporate firewalls), add your own TURN server in .env:

WIRETALK_TURN_URLS=turn:turn.example.com:3478
WIRETALK_TURN_USERNAME=your_username
WIRETALK_TURN_CREDENTIAL=your_credential

TURN credentials are only sent to clients when username and credential are both set. Incomplete TURN config is ignored to prevent WebRTC errors.

Troubleshooting