Guide

Choosing a WebRTC Softphone for Your PBX

Most WebRTC softphones claim to work with any PBX. The compatibility question is the first one to answer. Here's what your PBX needs and what to ask before you commit to a vendor.

What WebRTC means for softphones

WebRTC (Web Real-Time Communication) is a set of browser APIs that handle audio and video calls. A WebRTC softphone runs entirely in the browser. There's nothing to install. Users open a URL, log in, and they're on the phone.

From a call quality standpoint, WebRTC uses encrypted audio by default. The same technology runs Google Meet and Microsoft Teams. For most VoIP deployments, the audio is indistinguishable from a native desktop application.

The practical benefit is deployment simplicity. You share a login link instead of distributing software. Updates happen on the server without user action. No installer, no driver configuration, no desktop support tickets about broken audio.

What your PBX needs

A WebRTC softphone connects to your PBX over a secure WebSocket (WSS) rather than the standard SIP connections used by desktop phones. This means your PBX needs two things: a PJSIP-based SIP stack (not the older chan_sip), and a WSS transport configured with a valid TLS certificate.

Most modern PBX systems already support this. The configuration is a one-time change, not an ongoing maintenance task. Once it's done, every user who logs in connects the same way without any client-side setup.

Older PBX systems: If you're running a PBX that only uses the older chan_sip module (not chan_pjsip), a WebRTC softphone will not connect to it without a SIP-to-WebRTC gateway in between. This adds complexity and a potential point of failure. It's worth checking which SIP module your system uses before evaluating softphones.

PBX systems that work

These are the platforms where WebRTC softphones connect without gateways or extra modules:

✓
Asterisk
Version 17 and above. Uses chan_pjsip with a WSS transport. The most common setup for self-hosted deployments.
✓
FreePBX
Built on Asterisk. Same chan_pjsip requirements. Web-based admin makes the transport configuration accessible without command line.
✓
FusionPBX
Multi-tenant FreeSWITCH-based. WebRTC support is built into the PJSIP profile configuration.
✓
VitalPBX
PJSIP-based with WSS transport support included. Commonly used in MSP environments for its multi-tenant architecture.

The right question to ask any vendor

Before committing to a WebRTC softphone, ask this directly: "Will it work with my PBX without additional modules, gateways, or hosted infrastructure from your side?"

Some vendors require a proprietary media server or SIP proxy they host. Some only support specific PBX platforms. Some need custom modules installed on your PBX. The answer should be specific: here are the platforms we support, here's what's required on the PBX, and here's how the connection works.

If the answer is vague or requires a lengthy call to "assess compatibility," treat that as a signal to keep looking.

What to watch out for

Proprietary infrastructure. If the vendor's softphone requires a server they control in the call path, your dependency on them extends beyond the software license. If their server goes down or the relationship ends, so does your calling.

Custom modules on your PBX. Extra modules mean extra maintenance. A WebRTC softphone that connects to a standard PJSIP+WSS configuration requires no modifications to your PBX beyond the initial transport setup.

Self-signed TLS certificates. WSS requires a valid TLS certificate on your PBX domain. Most modern browsers reject self-signed certificates silently. If your PBX domain doesn't have a real cert, this needs to be resolved before a WebRTC softphone will connect reliably.

The clearest sign of a compatible setup: The vendor can point you to a live demo running against a standard Asterisk or FreePBX configuration, with no proprietary add-ons.

A WebRTC softphone that connects to your existing PBX

Stylish Softphone works with any PJSIP-based PBX with a WSS transport. No gateways, no proprietary modules, no hosted media servers. Users connect directly from their browser.

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