WebRTC calling lets a person make and take business calls from a web browser or a light desktop app, with no desk phone and no separate softphone to install. For work-from-home staff, sales teams on laptops and call centre agents, it can replace a lot of hardware. But WebRTC has its own network and security needs. This guide explains what WebRTC calling is, how it differs from a normal SIP phone, how it connects to an IP PBX, and how to set it up so calls sound clear.
What WebRTC is
WebRTC (Web Real-Time Communication) is a set of standards built into modern browsers such as Chrome, Edge, Firefox and Safari. It lets a web page use the microphone and speaker and send live audio, without plugins. When a web app or desktop app uses WebRTC to register to your PBX, the user gets a full extension: they can call colleagues, take queue calls, transfer and hold, just like on a desk phone.
Under the hood, a WebRTC phone still uses SIP to set up calls. The difference is how it travels:
- Signalling goes over WebSocket (WS) or secure WebSocket (WSS) instead of plain UDP or TCP.
- Audio is always encrypted (SRTP with DTLS keys).
- NAT traversal uses ICE, with STUN and TURN servers to find a working path through home routers and office firewalls.
WebRTC vs a SIP phone
| WebRTC (browser or app) | SIP desk phone or softphone | |
|---|---|---|
| Install | Nothing, or a light app | Desk phone or installed softphone |
| Connects over | WSS (secure WebSocket) | UDP, TCP or TLS |
| Encryption | Always on | Optional, depends on settings |
| Needs a certificate on the PBX | Yes, for WSS | Only if you use TLS |
| Works from home | Yes, with STUN/TURN | Yes, with NAT settings |
| Best for | Agents, laptop users, remote staff | Reception, desks, heavy phone users |
Many businesses use both. Reception and desk staff keep IP phones, while agents and travelling staff use WebRTC. See how to choose an IP phone for the desk side.
How WebRTC connects to Voxdial IP PBX
Voxdial IP PBX supports browser-based phones through its PJSIP settings:
- WS and WSS transports: the PBX can listen for WebSocket connections. WSS (secure WebSocket) is the recommended option for WebRTC; plain WS is unencrypted.
- Certificate Manager: WSS needs a valid security certificate, chosen in the PJSIP settings. Browsers refuse to connect without one.
- WebRTC STUN and TURN settings: separate STUN and TURN servers can be set just for browser phones. If left blank, the general STUN and TURN settings are used.
- Max Contacts on PJSIP extensions: one extension can register more than one device, so a user can have a desk phone and a WebRTC client on the same number.
In Voxdial Contact Center, the agent console is a desktop app with a built-in WebRTC softphone. Agents take queue calls, transfer, hold, park and conference from the console, and it includes STUN/TURN support and a call-quality monitor for remote agents.
Setting it up: a checklist
- Turn on WSS on the PBX and choose a valid certificate for your PBX’s domain name. Use a proper domain, not a bare IP address, because browsers check the certificate name.
- Open the right ports on your firewall for WSS signalling and the PBX’s RTP media range.
- Set STUN, and TURN for remote users. STUN helps the browser find its public address. TURN relays audio when a direct path is blocked, which is common on strict corporate or mobile networks.
- Create or edit the extension and set Max Contacts if the user will also keep a desk phone.
- Test from outside the office: a home broadband line and a mobile hotspot. Most WebRTC problems only show up off the office network.
- Give users a headset. A laptop microphone picks up the keyboard and the room. See the call centre headsets guide.
Common WebRTC problems and fixes
| Problem | Likely cause | Fix |
|---|---|---|
| Phone will not register | WSS off, wrong certificate or name mismatch | Enable WSS, use a valid certificate for the domain |
| Calls connect but no audio | ICE cannot find a path | Add a TURN server, check the RTP port range |
| Audio fine in office, bad at home | Home Wi-Fi or shared bandwidth | Wired connection or 5 GHz Wi-Fi, headset, close video calls |
| Browser asks for microphone every time | Site not trusted or permission blocked | Allow the microphone for the site in browser settings |
| Choppy audio under load | Laptop CPU or network congestion | Close heavy apps, check jitter and loss |
For deeper network issues, the VoIP one way audio guide and VoIP call quality for remote agents walk through the checks step by step.
Security points
- Always use WSS, never plain WS, on anything reachable from the internet.
- Strong extension passwords and account lockouts matter even more when phones register from anywhere. Read PBX security and toll fraud.
- Limit who can call where with class of service rules, so a stolen login cannot dial international numbers.
Want browser and app calling for your remote team on your own PBX? Talk to BitVoice about a setup that fits your network.
FAQs
What is WebRTC calling?
It is calling from a browser or light app using WebRTC, a standard built into modern browsers. The app registers to your PBX as an extension and sends encrypted audio over the internet.
Is WebRTC the same as SIP?
No, but they work together. WebRTC phones still use SIP to set up calls, carried over secure WebSocket (WSS), and always encrypt the audio.
Why does WebRTC need a TURN server?
When home routers or strict firewalls block a direct audio path, a TURN server relays the audio so the call still connects with sound in both directions.
Can one extension use both a desk phone and a WebRTC client?
Yes. On a PJSIP extension in Voxdial IP PBX, Max Contacts allows more than one device to register to the same extension.
