WebRTC-to-SIP Gateway Deployment

Companies with an established PBX or carrier SIP trunk often want to add browser-based calling — a click-to-call widget on a website, a web-based agent softphone for a contact center, or a telehealth video consultation feature — without replatforming the telephony they already depend on. Cloud gateway APIs solve this quickly but bill per-minute and route media through a third party's network, which is a non-starter for some compliance-sensitive deployments.

CelloIP is repeatedly engaged to build a self-hosted WebRTC-to-SIP gateway that sits directly in front of the client's existing PBX, giving them browser calling without a third-party media path or a rip-and-replace of infrastructure already in production.

The typical pattern: a browser client establishes a WebRTC connection, negotiating ICE candidates via a dedicated coturn STUN/TURN server (required for the 20–40% of sessions that can't connect directly due to NAT or firewall restrictions). FreeSWITCH, running mod_verto, terminates the WebRTC media, transcodes DTLS-SRTP to standard RTP, and hands the call to the existing PBX over a normal SIP trunk — so from the PBX's perspective, it's just another SIP call. No changes are required to the PBX's existing dialplan, IVR, or agent routing.

Sizing the TURN server correctly is the detail most teams underestimate: TURN relays all media for any session that needs it, so bandwidth planning has to account for a meaningful fraction of sessions relaying full-duplex audio (and video, if applicable) rather than just the signalling load.