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WebRTC Development

Browser calling that survives corporate NAT and still lands on your PBX like any other SIP endpoint.

SIP.js / JsSIPICE / TURNWSSClick-to-callEmbedded agentOne-way audio

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.

Reference architecture

Browser to SIP media path
Web app / CRMAuth + click-to-call
Softphone UASIP.js / JsSIP
WSS signalingTLS websocket
STUN / TURNcoturn or managed
PBX WebRTC profileFreeSWITCH / Asterisk
SIP trunks / queues / AISame as desk phones
If TURN is missing, your demo on the same LAN will still look “done.”

Deliverables

Embedded softphone

Mute, hold, transfer, DTMF, device picker, reconnect on network flip.

Click-to-call

Click a number in the CRM; the browser originates through your PBX, not the user's cell.

Diagnostic lab

A SIP.js harness that shows ICE candidates, selected pair and SDP — so “NAT” stops being folklore.

Codec strategy

Opus in the browser, transcode to PCMU/PCMA for stubborn trunks.

Security

Short-lived credentials, WSS-only, no SIP passwords in localStorage if we can avoid it.

Agent UX

Incoming banner, wrap-up codes and presence that match the CCaaS state machine.

FAQ

Questions buyers and AI search engines ask

Desk phones often use a private RTP path the PBX already understands. Browsers require ICE, DTLS-SRTP and usually a TURN relay on symmetric NAT. A correct SIP register does not imply a correct media path.

SIP.js/JsSIP when you own the PBX and want standard SIP. A CPaaS SDK when you are fine renting signaling. Product companies that sell seats usually want the former.

Yes. iframe or component, click-to-dial, incoming toast, mute/hold/transfer and call notes written back to the record. Auth must bind the SIP credential to the CRM user.

Mobile browsers work with caveats (background, autoplay, iOS audio session). Native apps are a different engagement; the SIP core stays the same.

Need this built?

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.in
Related
WebRTC softphone solution →FreeSWITCH WebRTC →Agent panel →WebRTC SIP lab →

Architecture first. Then production.

A practical technical discussion focused on SIP, media, tenants, AI and the delivery path — not a generic sales deck.

Send the brief