WebRTC Development Company
We build browser-based voice, video and live streaming — and connect it to the phone network. LiveKit, mediasoup, Janus, Jitsi and Ant Media, backed by fourteen years of carrier-grade VoIP engineering.
Core Capabilities
Browser Voice & Video
One-to-one calls, group rooms, screen sharing and recording on LiveKit, mediasoup, Janus or Jitsi — built into your product rather than bolted on as a third-party iframe.
Browser-to-PSTN Bridging
A web call and a phone number as one conversation, bridged through Asterisk, FreeSWITCH or Kamailio. This is where most WebRTC shops stop and where our telephony practice starts.
SFU Scaling
Selective forwarding, simulcast and bandwidth adaptation so a room holds up beyond a handful of participants — with multi-region deployment when your users are not in one place.
NAT & Firewall Traversal
ICE, STUN and coturn TURN relays configured properly, so calls still connect from corporate networks and mobile carriers that block direct peer connections.
Live Streaming & Recording
Ant Media and RTMP ingest with HLS and low-latency HLS delivery, server-side recording, and storage you own rather than a per-minute platform bill.
Call Quality Debugging
getStats instrumentation, packet loss and jitter tracing, and real captures rather than guesswork — the same diagnostic discipline we apply to SIP.
Tools & Platforms
RelatedCase study: a browser softphone on FreeSWITCHLiveKit vs Jitsi: choosing a WebRTC SFU
Technical Questions
Answered
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Both, and they are different engagements. We take fixed-scope projects end to end — architecture, build, load testing and deployment — and we also place WebRTC engineers into your team if you already have a direction and want capacity. If you are not sure which you need, the consultation call is the fastest way to find out.
Yes, and it is the thing we are asked for most. We bridge WebRTC to the PSTN through Asterisk, FreeSWITCH or Kamailio, so a user in a browser and a caller on a mobile are in the same conversation. Most WebRTC specialists do not do telephony and most telephony shops do not do WebRTC; we have run both for years.
It depends on what you are building. LiveKit is the fastest route to a working product and has the best AI-agent story. mediasoup gives the most control and the lowest overhead if you have the engineering appetite. Janus suits gateway and mixed-protocol work. Jitsi is strongest when you want a conferencing product rather than a component. We will tell you which one fits before you commit to it.
Only if TURN is set up properly, which is where most self-built deployments fail. We deploy and tune coturn, configure ICE candidate gathering, and test from genuinely restrictive networks rather than assuming a relay works because it started.
Ongoing engineering runs $25–$50 an hour. For fixed-scope work we quote after a technical call: a one-to-one video feature and a multi-party platform that has to hold thousands of concurrent participants are different orders of magnitude, and a range quoted before we know which one you need would be a guess rather than an estimate.
In-Depth Technical Reads
Building a Browser Softphone on FreeSWITCH and JsSIP: What Actually Broke
A case study from our own call-centre platform: SIP over WebSocket behind a TLS proxy, transfers that raced the audio, and SIP isolation between workspaces.
Read Article →LiveKit vs Jitsi in 2026: Architecture, Scaling and What Each Actually Costs
Two Apache-2.0 SFUs that are not interchangeable. Architecture, how each scales, and why LiveKit Cloud and Jitsi as a Service bill on meters that do not convert.
Read Article →VAPI vs LiveKit for AI Voice Agents in 2026: A Developer Head-to-Head
VAPI vs LiveKit in 2026: measured latency, what each platform charges per minute, free tiers, SIP trunking and how to migrate. Vendor pricing checked 14 Sep 2026.
Read Article →Get a Free WebRTC Consultation
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