Make Browser SIP Calls with SIP over WebSocket
Use StartACall to run SIP signaling through secure WebSocket connections so your browser can act as a full SIP endpoint. Bridge to PSTN via Twilio and call any phone number without installing software.
What is SIP over WebSocket?
SIP over WebSocket wraps standard SIP signaling inside WebSocket frames so modern browsers can register as SIP clients. Combined with WebRTC media, it enables a true browser based telephony experience.
Browser Native
No plugins or downloads. Use Chrome, Firefox, Edge, or Safari to place and receive SIP calls directly from the web. StartACall handles the SIP over WebSocket client stack for you.
PSTN Reach
StartACall bridges WSS to Twilio and SIP trunks so your browser calls can reach any phone number worldwide. Manage virtual numbers and inbound routing from the dashboard.
AI and Analytics
Enable AI agents, live copilot assistance, transcriptions, and real-time analytics while using SIP over WebSocket to power smarter voice workflows.
How StartACall implements SIP over WebSocket
A typical call flow connects the browser to PSTN using secure WebSocket signaling and WebRTC media. StartACall orchestrates the components so you can focus on calling.
Step by step
- Browser creates a secure WebSocket (WSS) to StartACall signaling gateway.
- StartACall translates WSS SIP signaling to Twilio or your SIP trunk using standard SIP messages.
- Media is negotiated with WebRTC and routed through StartACall or Twilio as required.
- Calls reach PSTN numbers via Twilio bridging, and call events are reported back for analytics and AI features.
Developer friendly
StartACall provides tokens and SDKs to register browser clients using SIP over WebSocket. Integrate quickly and use webhooks for call events, recordings, and transcriptions.
Security and Reliability
SIP over WebSocket runs over TLS, and StartACall uses SRTP for media. We integrate with Twilio and deploy hardened gateways to ensure call integrity and uptime.
Encrypted Signaling
WSS over TLS secures SIP messages between browser and gateway.
Secure Media
SRTP encrypts audio streams so media remains private.
Monitoring
Real time analytics for call quality and usage with alerts.
Common Use Cases
SIP over WebSocket unlocks new workflows for sales, support, and voice automation using just a browser.
Contact Centers
Agent consoles run in the browser with live AI assistance and call recording.
Remote Teams
Sales and support teams make secure PSTN calls from any device without softphones.
Voice Automation
Deploy AI agents to handle inbound or outbound SIP calls using StartACall voice AI stack.
Frequently Asked Questions
Is SIP over WebSocket supported by all browsers?
Modern browsers like Chrome, Firefox, Edge, and Safari support WebSocket and WebRTC. StartACall tests common configurations to ensure compatibility.
Do I need special ports open on my firewall?
SIP over WebSocket uses standard HTTPS ports when configured with WSS which minimizes firewall issues. Media uses RTP over standard ranges which StartACall documents for network configuration.
Can I use my existing SIP provider?
Yes. StartACall can bridge WSS SIP signaling to many SIP trunks and to Twilio so you can keep existing SIP infrastructure while enabling browser endpoints.
How do AI agents work with SIP calls?
StartACall can run autonomous voice agents for outbound campaigns or provide a live copilot during calls by transcribing audio and suggesting responses in real time.
Ready to run SIP in the browser?
Sign up and enable SIP over WebSocket for instant browser based calling, AI capabilities, and global PSTN reach.