HSG IT WebRTC Development for US Companies
— senior engineers, shipped products
We're an IT recruitment specialist with deep WebRTC roots — we built and operate WebCallHub, a browser-to-browser calling SaaS. We source senior WebRTC, Asterisk, SIP and Kamailio engineers from our EU + India network. C2C, W2, FTE. Typical placement 2–4 weeks.
What We Build
Browser Calling Widgets
Embed-anywhere call buttons (JsSIP-based), website-to-agent calls, click-to-call, call queues, IVR — like WebCallHub but custom for your product.
Video Conferencing
Multi-party video on mediasoup, Janus, or Jitsi. Adaptive bitrate, simulcast, recording, screen-share, breakout rooms.
SIP Infrastructure
Asterisk, FreeSWITCH, Kamailio, OpenSIPS. PBX builds, SIP trunk integration, SBC deployment, DID provisioning.
SFU / MCU Builds
Scalable media servers for 1:1, 1:many, and many:many. Coturn + TURN for NAT traversal. Geo-distributed routing.
AI Voice Layer
Source engineers experienced with real-time STT (Deepgram, Whisper) and AI voice integration — same stack we built into WebCallHub.
Call Analytics & CDR
Call detail record pipelines, ML-based fraud detection, quality scoring, agent performance dashboards, and BI integrations.
We Eat Our Own Dogfood
WebCallHub — our flagship WebRTC product
A browser-to-browser calling SaaS that lets businesses add a "Call us" button to their website. Visitors click, the agent answers in their browser — no phone numbers, no apps, no downloads. AI transcribes the call live, logs it in the CRM, and surfaces insights.
Tech stack: JsSIP, Asterisk PBX (host network), Coturn (TURN+STUN), FastAPI, PostgreSQL, Deepgram/Whisper, React, AWS. Built and shipped by the same WebRTC engineers you'd hire.
Try WebCallHub Free →How We Engage
WebRTC Audit
2-week audit of your existing real-time stack: codec selection, NAT traversal, MOS scores, infra cost, scaling bottlenecks.
Book AuditDedicated WebRTC Team
2–3 engineer WebRTC pod owning your real-time product. Front-end (JsSIP/RTC), media server, and SIP gateway.
Start TeamWebRTC Staff Aug
Senior WebRTC engineer embedded in your team. Asterisk, FreeSWITCH, SIP, Kamailio specialists.
Get QuoteWebRTC FAQ
What WebRTC services do you offer?
End-to-end WebRTC product development: browser-to-browser calling, video conferencing, real-time messaging, SFU/MCU integrations (mediasoup, Janus, Jitsi), SIP gateway integration (Asterisk, FreeSWITCH, Kamailio), recording, transcription, and TURN/STUN infrastructure.
Have you actually shipped a WebRTC product?
Yes — we built WebCallHub, a browser-to-browser calling SaaS used by businesses worldwide. JsSIP + Asterisk-based, with AI live transcription, multi-tenant architecture, and 99.9% uptime. We dogfood it on our own websites.
Can you scale to thousands of concurrent calls?
Yes. Our engineers have built telephony platforms processing 10,000+ concurrent calls using Kamailio + Asterisk + SFU clusters. Horizontal scaling with autoscaling Kubernetes pools, geo-distributed TURN, and adaptive bitrate.
Do you handle PSTN integration / SIP trunks?
Yes. SIP trunk integration with Twilio, Telnyx, Vonage, Bandwidth, and EU carriers. E.164 number provisioning, DID/DOD routing, fraud detection, and CDR pipelines.
Need WebRTC engineers who've actually shipped?
Book a free 30-minute WebRTC scoping call. We'll review your architecture and recommend tooling, scaling, and team structure.