WebRTC Asterisk: How to Power Browser-Based VoIP with Asterisk
For decades, Asterisk has been the open-source backbone of countless PBXs and VoIP systems. Meanwhile, WebRTC has emerged as the browser-native standard for secure, real-time communication. Put them together, and you unlock a powerful solution: WebRTC Asterisk integration, enabling direct browser-to-PBX calling without plugins or dedicated softphone apps.
Why Bring WebRTC and Asterisk Together?
- Browser Calling
Users can make and receive calls from Chrome, Firefox, or Safari without installing softphone software. - Cost Savings
No need for dedicated SIP phones or licence-heavy applications — WebRTC enables direct, low-cost communication. - Security by Default
WebRTC uses DTLS-SRTP encryption, ensuring voice and video streams are protected end-to-end. - Remote-Friendly
Perfect for hybrid teams: employees only need a browser and internet connection to access the corporate PBX.
For an overview of how WebRTC works, see Mozilla’s WebRTC guide.
How WebRTC Works with Asterisk
Asterisk already supports SIP signalling, but enabling WebRTC requires specific configuration. In practice, you’ll be:
- Enabling WebRTC Support
Use the PJSIP stack in Asterisk, which handles WebRTC-compatible SIP endpoints. - Configuring STUN/TURN Servers
These assist with NAT traversal so browser clients can establish peer-to-peer connections reliably. - Enabling TLS + SRTP
Essential for encrypted signalling and media streams. - Using a WebRTC Softphone
Browser clients like JsSIP can register to Asterisk and make calls directly from a webpage. For teams that prefer a ready-to-use solution instead of coding from scratch, platforms like Siperb provide a secure WebRTC SIP interface with minimal setup.
Example Use Cases
- Call Centres
Agents can log in from any browser and start taking calls instantly. - CRM Integration
Embed a WebRTC softphone into a web-based CRM so staff can dial customers without switching applications. - Remote Teams
Browser-first communication reduces IT overhead and ensures flexibility for distributed workforces.
Challenges to Keep in Mind
- Firewall/NAT Issues: Proper STUN/TURN setup is critical for call reliability.
- Codec Compatibility: Asterisk and the browser must share a common codec (Opus and G.711 are safe bets).
- SSL Certificates: Browsers require valid TLS certificates to establish WebRTC sessions.
The Future of Asterisk with WebRTC
By combining Asterisk’s proven SIP capabilities with WebRTC’s browser-native strengths, businesses get a PBX that’s not only flexible but also future-ready. Whether for call centres, enterprise VoIP, or web-embedded communication tools, WebRTC Asterisk integration is fast becoming the standard.
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