WebRTC-SIP Gateway
Extend Your PBX to the Browser
Add WebRTC capabilities to your existing SIP phone system. Let employees and customers make and receive calls directly from web browsers — no additional hardware, no separate platform, just a seamless extension of your PBX.
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Key Features
WebRTC-to-SIP Bridge
The gateway translates WebRTC protocols to SIP in real time, allowing any modern web browser to register as an extension on your existing phone system. No need to replace or modify your SIP infrastructure — the gateway sits between your PBX and browser-based users, handling codec transcoding (Opus to G.711), signal translation (WebSocket to SIP UDP/TCP), and media relay. Your PBX sees each WebRTC client as a standard SIP endpoint, fully integrated into your dial plan, call routing rules, and queue configurations.
Browser Softphone
A full-featured SIP phone running entirely in the web browser. Agents and employees get a complete telephony experience — make and receive calls, put callers on hold, transfer to extensions or external numbers, and see BLF (Busy Lamp Field) status of colleagues. The softphone integrates with your PBX directory, supports speed dial, call history, and voicemail access. No desktop application to install or maintain — just open the browser and start working.
Click-to-Call
Embed a call button on your website and let visitors connect to your PBX instantly from their browser. The call is routed through the WebRTC-SIP gateway directly to your IVR, queue, or specific extension — exactly like a regular incoming call. The widget is customizable to match your brand and can be placed on any page. Each call appears in your PBX CDR reports with full tracking of origin, duration, and outcome.
No Plugins Needed
Works natively in Chrome, Firefox, Safari, and Edge without any plugins, downloads, or installations. WebRTC is built into the browser itself, so users simply open a web page and connect to your PBX. No IT approvals, no Java or Flash dependencies, and no compatibility issues. This makes it the fastest way to onboard remote workers, temporary staff, or branch offices — anyone with a browser and internet connection can be on your phone system in minutes.
Secure Communications
All voice traffic between the browser and your PBX is encrypted with SRTP and DTLS. Signaling is protected with TLS over WebSocket (WSS). The gateway handles the encryption boundary — WebRTC clients use mandatory encryption while SIP-side security follows your PBX configuration. Compliant with GDPR and enterprise security requirements including SOC 2 and ISO 27001.
Mobile Support
The browser softphone works on smartphones and tablets with adaptive audio quality that adjusts to cellular and WiFi conditions. Employees can use their mobile browser to connect to the PBX while traveling or working remotely, using the same extension number and features as their desk phone. Push notifications alert users of incoming calls even when the browser is in the background. Seamless handoff between WiFi and cellular data keeps calls connected on the move.
Capabilities
- WebRTC-to-SIP protocol translation
- Codec transcoding (Opus ↔ G.711)
- Call transfer, hold & park
- DTMF relay
- BLF / Presence indication
- Call recording integration
- NAT/firewall traversal (TURN/STUN)
- Multi-tenant support
- Caller ID passthrough
- Auto-provisioning via REST API
Technical Specs
- JavaScript SDK & softphone widget
- SIP over WebSocket (WSS)
- Opus & G.711 audio codecs
- SRTP & DTLS encryption
- ICE/STUN/TURN traversal
- REST API for provisioning
- Compatible with any SIP PBX
- 99.9% uptime SLA
- Scalable media relay servers
- GDPR compliant
Ready to Get Started?
Contact our sales team for a personalized demo and pricing.
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