What is WebRTC development?
WebRTC (Web Real-Time Communication) development involves building browser and mobile applications that enable real-time audio, video, and data communication without plugins. CelloIP Technologies builds WebRTC click-to-call widgets, video conferencing platforms, SIP-WebRTC gateways, and LiveKit-based voice AI systems.
How does WebRTC connect to traditional phone systems?
WebRTC calls connect to SIP/PSTN networks via a WebRTC-to-SIP gateway — typically FreeSWITCH with mod_verto or Kamailio with rtpengine. The gateway handles DTLS-SRTP to RTP transcoding and ICE/STUN/TURN for NAT traversal. CelloIP has deployed WebRTC gateways handling 3,200+ concurrent sessions.
How long does WebRTC development take?
A basic WebRTC click-to-call widget takes 3–4 weeks. A full WebRTC video conferencing platform with SIP integration and mobile SDKs takes 12–20 weeks. CelloIP provides fixed-price WebRTC project packages and dedicated WebRTC developer retainers from $25/hr.
Custom WebRTC Development Services
CelloIP builds browser-native voice and video communication apps using WebRTC — click-to-call, conferencing, telehealth, SIP gateways, and real-time data channels. No plugins. No downloads. Works everywhere.
WebRTC Development Services
Every WebRTC use case — from a simple click-to-call button to a full-scale video conferencing platform.
Click-to-Call Integration
One-click browser calling from any web page or CRM. Visitors call your team without a phone number or app. Works on all modern browsers via WebRTC with SIP gateway to your existing PBX or contact centre.
Video Conferencing Platform
Custom multi-party video conferencing built on WebRTC with Mediasoup or Janus as the SFU. Up to 100 participants, screen sharing, recording, virtual backgrounds, breakout rooms, and chat. White-label branded.
SIP-WebRTC Gateway
Bridge browser-based WebRTC calls to your SIP PBX or carrier network. Users call PSTN numbers from the browser; PBX users receive calls on their SIP phones. Asterisk PJSIP-WS, FreeSWITCH mod_verto, or Janus SIP plugin.
Telehealth & Telemedicine
HIPAA-compliant video consultation platform: scheduled and on-demand sessions, waiting room, patient consent capture, session recording with access control, EHR integration, and encrypted media (DTLS-SRTP).
WebRTC Mobile SDK
Native WebRTC integration for iOS (Swift) and Android (Kotlin/Java) apps using Google's WebRTC library. Custom peer connection management, ICE handling, camera/mic management, and background call support.
WebRTC Signalling Server
Custom WebSocket-based signalling server for WebRTC peer connection negotiation. Built in Node.js, Python, or Go. Room management, offer/answer exchange, ICE candidate relay, and presence signalling.
STUN/TURN Infrastructure
Dedicated coturn STUN/TURN server deployment and configuration for reliable NAT traversal. TCP/TLS TURN fallback for restrictive firewalls, authentication via HMAC-SHA1, and traffic monitoring.
Real-Time Data Channels
WebRTC DataChannel applications for real-time collaborative tools, file transfer, P2P game networking, live document editing, IoT sensor data streaming, and financial tick data broadcast.
WebRTC Technology Stack
A complete picture of every layer in a production WebRTC application.
Client SDKs
- browser WebRTC API
- SIP.js
- JsSIP
- Google WebRTC (iOS/Android)
Media Servers
- Mediasoup 3.x (SFU)
- Janus WebRTC Gateway
- coturn (STUN/TURN)
- rtpengine
Signalling
- WebSocket (Node.js/Go)
- Socket.io
- WHIP/WHEP
- SIP over WebSocket
SIP Gateways
- Asterisk PJSIP-WS
- FreeSWITCH mod_verto
- OpenSIPS ws_transport
- Kamailio websocket
Protocols
- ICE (RFC 8445)
- DTLS-SRTP
- SCTP (DataChannel)
- VP8/H.264 video
- Opus audio
WebRTC Use Cases We Build
Contact Centre Web Softphone
Agents handle calls directly in the browser-based CRM — no desk phone required. Click-to-answer inbound, click-to-dial outbound, mute, hold, transfer, and disposition — all in the web UI.
Telehealth Platform
HIPAA-compliant browser video consultations between doctors and patients. Zero-install, works on mobile browsers, encrypted media, session recording with patient consent.
EdTech Virtual Classroom
Live video lectures, student Q&A, screen sharing, interactive whiteboard, recording, and breakout rooms. Scales to 200 concurrent video participants via SFU architecture.
Financial Trading Floor
Low-latency one-click voice communication between traders, brokers, and market makers. Squawk box simulation, group listen, broadcast to rooms — sub-100ms audio latency.
Field Service Video Support
Technicians in the field video-call a remote expert from their phone browser. Expert sees the field via rear camera, draws annotations on screen, guides the technician in real time.
IoT & Smart Device Audio
Browser-based two-way audio for smart doorbells, IP cameras, industrial machinery, and access control systems. SIP gateway bridges intercom calls to mobile SIP dialers.
WebRTC Development: What You Need to Know
WebRTC (Web Real-Time Communication) is a W3C and IETF standard that enables browser-based audio, video, and arbitrary data communication without plugins. Introduced in Chrome in 2012 and now universally supported, WebRTC has become the infrastructure underlying Google Meet, Zoom's browser client, Teams on the web, and thousands of custom communication applications.
Why is WebRTC Development Complex?
Despite WebRTC being a browser API, production WebRTC applications require substantial server-side infrastructure. ICE candidate gathering, STUN/TURN server configuration, signalling server architecture, SDP negotiation edge cases, codec compatibility across browsers, network quality adaptation, and SFU media server selection are all engineering decisions that dramatically affect call quality and reliability. Getting NAT traversal wrong, for example, results in 15–20% of calls failing silently for users behind symmetric NATs.
Mediasoup vs Janus: Choosing the Right WebRTC SFU
Mediasoup is a Node.js library (with a C++ worker) that provides low-level WebRTC SFU primitives. It requires more custom code to build the application layer but provides maximum control and performance — suitable for large-scale conferencing platforms. Janus is a more complete WebRTC gateway with built-in plugins for conferencing (janus-plugin-videoroom), SIP gateway (janus-plugin-sip), and streaming — better for applications that need multiple WebRTC use cases in one server. CelloIP recommends Mediasoup for purpose-built conferencing platforms and Janus for hybrid SIP+WebRTC deployments.
WebRTC Development FAQ
What is WebRTC development?
WebRTC (Web Real-Time Communication) development is the engineering process of building browser-based or app-based audio, video, and data communication features using the WebRTC standard APIs. It includes peer connection establishment, ICE/STUN/TURN configuration for NAT traversal, signalling server development, media server integration for multi-party calls, and SIP gateway integration for PSTN connectivity. No browser plugin or native app download is required for WebRTC calls.
What browsers support WebRTC?
WebRTC is supported natively by Google Chrome (version 28+), Mozilla Firefox (version 22+), Apple Safari (version 11+ on iOS and macOS), Microsoft Edge (Chromium-based), and Opera. Market coverage is effectively universal for modern browsers. CelloIP tests all WebRTC applications across Chrome, Firefox, Safari, and Edge on desktop and mobile — including Safari on iOS which has historically had the most WebRTC limitations.
What is the difference between a STUN server and a TURN server?
A STUN server helps a WebRTC peer discover its public IP address and port mapping (useful when behind a NAT). It does not relay media. A TURN server acts as a media relay when direct peer-to-peer communication is impossible (e.g., behind symmetric NATs or strict firewalls). TURN is bandwidth-intensive because all media flows through it. Best practice is to use STUN first and fall back to TURN only when direct connectivity fails — CelloIP configures ICE candidates appropriately for each deployment.
Can WebRTC calls connect to regular phone numbers (PSTN)?
Yes, through a SIP-WebRTC gateway. The browser establishes a WebRTC connection to a gateway server (such as Asterisk with PJSIP WebSocket transport, FreeSWITCH with mod_verto, or OpenSIPS with websocket module). The gateway converts the WebRTC media to standard SIP/RTP and routes the call to a SIP trunk provider or PSTN gateway. The caller hears a regular phone dial tone; the called party sees a standard incoming call.
What is a Selective Forwarding Unit (SFU) and when do I need one?
An SFU is a media server that receives media streams from all participants and selectively forwards them to others without mixing them (unlike an MCU which mixes all streams). SFUs are required for multi-party video calls beyond 3–4 participants because P2P WebRTC doesn't scale — each participant would need to send N-1 video streams. CelloIP uses Mediasoup and Janus as SFU engines for scalable conferencing deployments.
Flexible Engagement
Need a Dedicated WebRTC Developer?
Scale your engineering team with a vetted WebRTC specialist from CelloIP. NDA Day 1, full IP ownership, onboard in 48 hours. From $25/hr.
Start Your WebRTC Project
From a click-to-call widget to a 100-seat video conferencing platform — book a free technical assessment with a CelloIP WebRTC engineer.