WebRTC Development Company

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.