Open Source Development
Free consultation with our engineers
SIP.js development brings SIP calling into any web app using a JavaScript library built on WebRTC. We build web softphones, click-to-call buttons, video calls and agent panels that register to your existing SIP server.
On the server side, we set up Asterisk, FreeSWITCH or Kamailio for WebSocket and WebRTC media. You can hire one SIP.js developer or a small team, full time or per project. Either way, we test calls across Chrome, Firefox, Safari and Edge before your launch date.
What we offer
What we build
Why Driansh
How we work
We review your requirements, traffic and existing stack, then agree scope, milestones and cost.
We plan the architecture, integrations and failover so the platform fits your growth plans.
We develop in short sprints with regular demos, then load-test and QA before go-live.
We launch on your servers or cloud and stay on for monitoring, fixes and upgrades.
Get in touch
Tell us about your app and SIP server. A SIP.js engineer will review it and suggest how to build it.
Share a few details and the right engineer will get back to you.