Browser calling that survives corporate NAT and still lands on your PBX like any other SIP endpoint.
WebRTC development for a PBX is a signaling + ICE + codec problem. UnifiedPBX.in builds browser softphones and click-to-call that register to FreeSWITCH, Asterisk or FusionPBX, then stay up when the user is on hotel Wi-Fi.
SaaS products that promised in-browser calling and now fail on corporate NAT, Safari or one-way audio after hold.
SIP.js (or equivalent) softphone, ICE/TURN that works off-LAN, click-to-call, inbound toasts and a lab that reproduces the failure.
WebRTC softphone development, SIP.js consultant, WebRTC one-way audio, click-to-call in SaaS.
In 2026 every vertical SaaS wants a phone in the tab. Most ship a library and no TURN story. The pages that rank name ICE states and the FusionPBX/FreeSWITCH side, not “WebRTC enabled.”
Mute, hold, transfer, DTMF, device picker, reconnect on network flip.
Click a number in the CRM; the browser originates through your PBX, not the user's cell.
A SIP.js harness that shows ICE candidates, selected pair and SDP, so “NAT” stops being folklore.
Opus in the browser, transcode to PCMU/PCMA for stubborn trunks.
Short-lived credentials, WSS-only, no SIP passwords in localStorage if we can avoid it.
Incoming banner, wrap-up codes and presence that match the CCaaS state machine.
Direct answers on scope, stack, rescue vs build, and how to start, structured for search and AI summaries.
Direct answers on scope, stack, rescue vs build, and how to start, structured for search and AI summaries.
Send the current architecture, call volume, carriers and the workflow you need to automate or productize. Discovery can start from a broken PBX or a blank product brief.
Discuss Your Project hello@unifiedpbx.inA practical technical discussion focused on SIP, media, tenants, AI and the delivery path, not a generic sales deck.