WebRTC Softphones: A Practical Guide for PBX Admins
If you run a PBX, you’ve probably heard people talk about “WebRTC softphones” and “browser calling” as if they’re the same thing. Sometimes they are. Often, they’re not. And in the middle of it all you still have real users, a real Asterisk or FreePBX box, and the same uptime targets as always.
This article breaks down what a WebRTC softphone actually is, how it compares to a traditional SIP softphone, and when it makes sense to plug one into your existing PBX instead of rolling out more desk phones or thick clients.
What is a WebRTC softphone?
A softphone is simply a software phone: an app that lets you make VoIP calls from a laptop, mobile, or browser instead of a desk handset. A good, neutral explanation is this guide from GetVoIP on what a softphone is and why you might use one. GetVoIP
A WebRTC softphone is a special kind of softphone that lives in the browser. It uses the WebRTC APIs in modern browsers to handle real-time media (voice and video), and usually talks SIP over WebSockets to your PBX or SBC behind the scenes. MDN Web Docs
In practice, that means:
- No separate app install for end users.
- Calls use the same browser they already live in all day.
- Audio and video are handled with standards-based encryption by default.
WebRTC softphone vs traditional SIP softphone
On the surface, both types of softphone give you a dial pad, contacts and call controls. The differences are mostly under the hood and in how you deploy them.
1. Where the app lives
- Traditional SIP softphone
- Installed as a desktop or mobile app.
- Needs packaging, updates, and OS-specific support.
- WebRTC softphone
- Runs entirely in the browser (Chrome, Edge, Safari, etc.).
- Users visit a URL and log in; updates are handled on the server side.
If you’re choosing between browser and installed clients generally, this comparison of browser-based and installed softphones is a helpful overview. Call Handling
2. How they connect
- Traditional SIP softphone
- Speaks SIP and RTP directly to your PBX, often over UDP.
- You manage NAT, firewall rules, and sometimes VPNs.
- WebRTC softphone
- Speaks WebRTC from the browser to a WebRTC-capable edge (gateway, proxy or SBC).
- That edge then bridges into SIP/RTP for your PBX.
- Uses secure WebSockets (WSS) and SRTP by design, which changes how you open ports and handle TLS. MDN Web Docs+1
3. User experience
- Traditional SIP softphone
- Feels like a classic phone client.
- Good when users live inside a dedicated app all day (for example, in a call centre).
- WebRTC softphone
- Feels like another web app.
- Great when users already work in a browser-based CRM, ticketing tool, or admin portal.
4. Maintenance and rollout
- Traditional SIP softphone
- IT needs to manage installers, versions, and device policies.
- Remote users can be tricky if they’re behind strict firewalls.
- WebRTC softphone
- Ship new features by updating the web app once.
- As long as users can reach your WebRTC edge over HTTPS/WSS, they can usually make calls.
When a WebRTC softphone makes sense
A WebRTC softphone isn’t always the right answer. But it is very attractive in a few specific situations:
- Browser-first teams
Sales, support, and remote staff who already live in browser tools and don’t want “yet another app”. - Distributed or remote workers
Staff connecting from home networks, co-working spaces, and café Wi-Fi, where SIP over UDP is often blocked but HTTPS is allowed. - Mixed devices and BYOD
When you don’t control every laptop and phone, a browser-based client saves you from supporting multiple OS-specific apps. - Tight integration with web apps
If you want click-to-call from a CRM, or auto-pop records on incoming calls, a WebRTC softphone embedded directly in that web UI is a natural fit.
If you run Asterisk, FreePBX or another SIP PBX, you still keep that investment. A WebRTC softphone simply becomes a new on-ramp into the system.
Key technical building blocks (without the jargon overload)
Under the surface, a WebRTC softphone relies on a small set of building blocks:
- WebRTC media – the browser APIs that capture mic/camera and send encrypted audio/video.
- Signalling – typically SIP carried over secure WebSockets to your proxy or PBX.
- ICE, STUN and TURN – the network plumbing that figures out how two endpoints can actually talk to each other across NAT and firewalls. MDN Web Docs+1
- Gateway or proxy – the piece that understands both WebRTC on one side and classic SIP/RTP on the other.
You don’t need to explain all of this to users. But it’s useful for PBX admins to know that a WebRTC softphone is not “just a web page”; there are real protocols and security models behind it.
Where Siperb fits into this picture
Most PBX teams don’t want to write their own WebRTC stack from scratch. That’s where products like Siperb come in.
Siperb combines:
- A browser-based WebRTC softphone that users can open in modern browsers.
- A SIP-to-WebRTC proxy layer that talks to Asterisk, FreePBX, FreeSWITCH and other SIP PBXs you already run. Siperb+1
If you’re specifically working with Asterisk, the guide to Asterisk WebRTC with Siperb shows how to wire up browser calling to your existing dialplan, without ripping out your current trunking or call flows. Siperb
In other words, Siperb gives you the WebRTC softphone and the gateway piece, while your PBX stays in charge of routing, queues, IVRs and recording. If you already have a SIP trunk and call flows you’re happy with, a setup like this lets you modernise the endpoint experience without rewriting your whole voice stack.
A simple checklist for choosing a WebRTC softphone
If you’re evaluating WebRTC softphones (commercial or open source), use this quick checklist:
- PBX compatibility – Does it support your flavour of SIP PBX (Asterisk, FreePBX, FusionPBX, etc.)?
- Security defaults – Does it use TLS/WSS and SRTP by default? How are certificates handled?
- Browser support – Which browsers and versions are officially supported?
- NAT and firewall handling – Does the solution provide TURN or guidance for dealing with restrictive networks?
- User authentication – Can it integrate with your existing login system, or does it manage its own?
- Integration options – Can you embed the softphone into a CRM or admin portal, or is it a standalone page only?
- Operational visibility – Are there logs, metrics or call traces that help you troubleshoot issues when calls fail?
Once those boxes are ticked, you can focus on user experience: audio quality, call controls, and how naturally it fits into your team’s daily workflow.
Browser-based calling doesn’t replace SIP or your PBX; it simply changes where the “phone” lives. For PBX admins and VoIP teams, a WebRTC softphone is often the most direct way to give modern, secure, remote-friendly calling to users without throwing away the telephony stack you’ve already built.
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