How to tell if a WebRTC developer actually knows their stuff

Hiring for WebRTC is tricky, because plenty of developers have used it once and now list it on their profile. Using WebRTC once and running it in production are two very different things. If you are about to hire WebRTC developers for a real time voice or video project, these are the questions that separate the people who can do the job from the people who have only ever seen the happy path.

Ask about TURN, not STUN

Almost everyone can explain STUN. Fewer people will talk honestly about TURN, and TURN is where real calls live or die. Someone who has actually shipped WebRTC will have opinions about running relay servers, about which regions to place them in, about the cost of relaying media versus letting it flow peer to peer. If a candidate waves off TURN as an edge case, they have not run this at any real scale.

Ask what they do when audio goes one way

This is the classic WebRTC failure. One side hears, the other does not. The answer you want is not a single magic fix, it is a diagnostic process. They should be reaching for ICE candidates, symmetric NAT, whether media is even arriving at the relay. People who have debugged this in production carry a mental checklist. People who have not will just start guessing.

Ask how they scale past a handful of people

Peer to peer falls over quickly once a room has more than a few participants. Anyone serious will bring up SFUs without being prompted, and probably name Janus, mediasoup, or Jitsi. Bonus points if they can explain why an SFU beats an MCU for most cases, and the situations where it does not.

Ask what they measure

If the answer is nothing, walk away. Good WebRTC people are a little obsessive about quality metrics, because they have been burned before. Packet loss, jitter, round trip time, MOS. They will want monitoring in place before launch, not bolted on after the first angry support ticket.

Individual developers or a full team?

This part depends on where you are. If you already have a solid engineering team and just need the real time expertise they are missing, hiring one or two specialists to embed with your people usually works best. They handle the media layer, your team keeps its context, and knowledge rubs off along the way.

If real time is central to your whole product and you do not have anyone in house who has done it, it can make more sense to work with a WebRTC development company that already runs this kind of infrastructure and can own the build end to end. You lose a little day to day control, you gain a team that has already solved the problems ahead of you.

Neither option is automatically better. It comes down to how much of the hard part you want to keep inside your own walls.

A couple of quick answers

How long does it take to get a WebRTC developer productive? 

Less than most other specialties, honestly, if they are genuinely experienced. The domain is narrow. A good one can read your signaling and media setup and be useful within days. A weak one will take weeks and still be guessing at NAT issues.

Should the same person do frontend and infrastructure? 

Sometimes, on a small project. But the frontend of a WebRTC app and the media infrastructure behind it are almost separate disciplines. On anything serious, do not assume one person is strong at both.