Browser SIP Client (WebRTC Softphone) for PBX

Modern teams live in the browser. Installing and maintaining desktop softphones slows everyone down. A WebRTC softphone (browser SIP client) puts secure, crystal-clear calling into any modern browser—no downloads, no VPN, no IT tickets. Below you’ll see how it works, why it’s different from UCaaS “phone apps,” how it connects to Asterisk/FreePBX or cloud PBX, and how to roll it out in minutes.

Quick start: try the product on the browser SIP client page.

What Is a WebRTC Softphone?

A WebRTC softphone is a SIP client that runs entirely in the browser. It uses WebRTC for media (audio/video) and encryption, and speaks SIP to your PBX or SBC—often via a lightweight SIP↔WebRTC proxy.

Key benefits

• Zero installs: open a tab, log in, start calling—ideal for remote agents and hot-desking. See the full list on features.
• Secure by default: SRTP/TLS with STUN/TURN for NAT traversal; details on security.
• PBX-friendly: works with Asterisk, FreePBX, FusionPBX, Kamailio, OpenSIPS, and most SIP trunks; see integrations for Asterisk and integrations for FreePBX.
• Scale fast: onboard a new agent in under 5 minutes—follow getting started.

How a Browser SIP Client Connects to Your PBX

1. User signs in to the web softphone.
2. The client registers to your PBX via TLS using SIP over WebSockets (WSS).
3. Calls negotiate media with WebRTC (ICE + STUN/TURN) and stream with DTLS-SRTP.
4. Optional: a SIP↔WebRTC proxy normalizes codecs, handles ICE candidates, and simplifies firewalling—diagram in how it works.

Result: encrypted signaling + media, reliable setup behind NAT, excellent call quality—without installing anything.

Features Teams Actually Use

• Click-to-call & call pop from CRM/helpdesk pages (see CRM shortcuts)
• Call controls: transfer (blind/attended), hold, mute, DTMF, recording (see call control)
• Presence & BLF so transfers land first time
• Multi-device login alongside desk phones or mobiles
• Policy & audit: enforce TLS/SRTP; log call events for QA/compliance (security & compliance)
• Asterisk/FreePBX integration: reuse extensions, dialplans, and queues—setup guides for Asterisk and FreePBX

WebRTC Softphone vs. Traditional Softphone or UCaaS App

• Installs: WebRTC runs in the browser; apps require downloads/updates.
• Security: WebRTC ships with DTLS-SRTP/TLS; legacy RTP can be unencrypted (see security overview).
• Control: keep your own PBX and routing; avoid vendor lock-in.
• Time to value: first call within minutes—start on webrtc softphone setup.

Who Benefits Most

• Contact centers & support desks needing rapid onboarding
• Hybrid/remote teams on Chromebook/Mac/Windows/Linux
• PBX owners wanting modern browser calling without replacing their stack
• Compliance-minded orgs needing encryption and auditability

Setup Checklist (≈10 Minutes)

PBX readiness

• Enable SIP over TLS
• Codecs: Opus, G.711
• NAT helpers (Asterisk: icesupport=yes, directmedia=no) — see Asterisk notes

Edge / proxy (recommended)

• Deploy SIP↔WebRTC proxy (WSS + ICE/TURN)
• Open ports for WSS and TURN (often TCP/UDP 443) — more in network requirements

Users

• Create or reuse SIP extensions
• Provide credentials (username, domain, TLS) — user onboarding

Go live

• User opens browser, signs in, allows mic, places test call
• Verify transfers/recording policies, then roll out to team via deployment checklist

Security & Call Quality (What IT Cares About)

• Encryption: TLS for signaling + DTLS-SRTP for media
• NAT traversal: built-in ICE/STUN/TURN for home/guest networks
• QoS tips: prefer Opus, prioritize UDP 443, keep TURN close to agents—see quality guide

Pricing & Deployment Options

• Free tier / pilot: validate call flows with a small team—pick pricing to compare tiers.
• Business: SSO, call recording, audit logs, SLA support
• Enterprise: HA TURN pools, custom SBC rules, and compliance features

You keep numbers, trunks, dialplans, and recording strategy. The browser softphone is a thin client that upgrades your existing PBX—see why keep your PBX.

Example Use Cases

• Sales & SDR: dial from CRM, auto-log, coaching recordings (see sales workflows)
• Support: warm transfers, presence-aware routing, in-browser QA review (see support workflows)
• Field ops: calling on lightweight laptops—no corporate image needed

FAQs

Is it compatible with Asterisk/FreePBX?

Yes—registers as a SIP endpoint over WSS/TLS. You can reuse dialplans, queues, and trunks.

Do I need a TURN server?

For remote users and strict NATs, yes. TURN guarantees media when peer-to-peer paths fail—see TURN/STUN guide.

Does it support call recording and transfers?

Yes—attended/blind transfer, hold, DTMF, and recording (subject to PBX policies). See call controls.

Can I use desk phones and the browser together?

Yes—multi-registration lets a desk phone and the web softphone run in parallel on the same extension.

What about video calling?

Video is native to WebRTC. Enable it per policy; audio-only can remain default for call-center use. Planning a rollout? Read adding video to your PBX.

Call to Action

Ready to try the browser SIP client? Start free today and place your first secure WebRTC call in minutes—no installs required.

• Primary CTA: Start Free
• Secondary CTA: Talk to an engineer

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