A guide to hiring a WebRTC developer
WebRTC is an open standard for live audio, video and data between browsers and apps. It powers video calls, voice chat, live classes, support calls and screen sharing without a plug-in. The standard is free to use and the main browsers support it. A WebRTC developer is someone who can turn that standard into a call feature that works for ordinary users, on ordinary networks, on ordinary phones.
Why WebRTC is harder than it looks
A sample project can show two browser tabs on a call in an hour. A product has to cope with firewalls, mobile networks, users who switch from Wi-Fi to mobile data, old devices and rooms with many people. The standard handles the media and the connection, but it leaves out several parts that you must build or choose. That is why a developer with real experience saves a lot of time.
Signaling
Before two users can talk, they must exchange a few messages that describe what each side can send and receive. The standard does not say how to carry those messages, so each app builds a signaling service. Many teams use WebSocket for it, and it is often written in Node.js. It also handles rooms, joins, leaves and users who reconnect. Keep it small and reliable. If you need this server side work, see our Node.js developers.
ICE, STUN and TURN
Most users sit behind a router or a firewall. ICE is the process that tries different paths between two users. A STUN server tells a user what their public address looks like. When no direct path works, a TURN server relays the media instead. Coturn is a well known open source TURN server. A TURN server costs bandwidth, so a developer plans where to place it and how to protect it from misuse. Without it, some users simply cannot connect.
Peer to peer or a media server
For a call between two people, media can often go straight from one to the other. As the room grows, every person would have to send their video to everyone else, which is too much for a phone. A media server solves this by receiving each stream once and forwarding it to the others. Janus, mediasoup, Jitsi Videobridge and LiveKit are open source examples. A developer picks one based on your room sizes, your platforms and the skills in your team, and explains the trade offs in plain words.
Audio and video choices
Calls use codecs to shrink audio and video. Opus is the common choice for audio, and video can use VP8, VP9, H.264 or AV1 depending on the device. Browsers include echo cancellation and noise reduction, and they can lower the video quality when the network gets weak. A developer decides how to set resolution and bitrate, how to handle devices with poor cameras and how to show a clear message when the network is the problem.
Mobile apps
WebRTC works inside iOS and Android apps through native libraries, and there are packages for cross platform tools. Phones add their own work. The app must ask for camera and microphone access, switch between speaker and earpiece, survive a change of network and deal with a call that comes in while the app is in the background. Teams that build with React Native often need a native specialist for the call screen.
Recording and playback
Recording is common in classes, interviews and support calls. It can happen on a media server, where each stream is saved and then mixed or joined into one file, or in the browser for simple cases. The developer also plans where files are stored, how long they are kept and who can watch them. Think early about permission from the people on the call and about the rules where your users live.
Security and privacy
WebRTC encrypts media in transit using DTLS and SRTP. Your part is the rest: who is allowed to join a room, how room links and tokens expire, how TURN credentials are issued and how recordings are stored. A careful developer treats these as design questions from day one. For services that connect your app to your user accounts and billing, see our API developers.
Call quality and testing
The browser can report statistics on packet loss, delay and bitrate. Collecting them across many calls turns a vague complaint into a cause. Testing should include slow and lossy networks, several browsers, older phones and rooms of different sizes. Browsers have flags that feed a fake camera into a test, which helps automated checks run in your build. A developer who tests like this finds problems before your users do.
When to hire a WebRTC developer
- You want to add video or voice calls to a web or mobile app.
- Calls fail for some users, or the video freezes.
- You need group rooms, live classes or a one to many stream.
- You want to record calls and play them back.
- A hosted video service has become too costly or too limited, and you want more control.
If you only need a simple meeting link with no custom features, a hosted service may be enough. A developer can tell you honestly which side of that line you are on.
How to read a WebRTC developer’s background
Ask which call features they have shipped and how many people were on a call. Ask them to draw how two users connect, from signaling to media. Listen for TURN, ICE and how they handled users on poor networks. Look for people who talk about measuring quality, not only about getting a call to start.
Interview questions to try
- What happens between pressing the call button and seeing the other person?
- When would you add a TURN server, and how do you protect it?
- When is a media server needed, and which one would you try first for our case?
- A user says the video freezes after a minute. What do you check?
- How would you test a call on a slow, lossy network?
A strong answer names a real project and what was learned on it. If a candidate only lists terms, ask for the story behind them.
Run a small paid exercise
Hand every finalist the same small job, such as a working one to one call with a simple signaling service, a test under a poor network or a review of how your current calls fail. Pay them for it. Compare the results, the questions they asked and how plainly they explain what they did.
Agree how the system is looked after
Your servers, domain names and cloud accounts should be registered to your company, and each person should get their own login. Ask for a short note on how to deploy the signaling service and the media server, and where the call statistics are kept. If your developer moves on, you lose nothing.
Settling in
Start with access to the code, a test environment and one small first task. One weekly call and a shared task list are enough to follow progress. Point out the odd cases early, such as the customers on strict company networks or the older tablets in the field. Those are the places where surprises hide.
Mistakes to avoid
- Testing only on the office network and on the latest laptop.
- Skipping a TURN server and finding out later that some users cannot connect.
- Choosing a media server for its fame rather than for your room sizes and skills.
- Letting the cloud account or server logins sit under a contractor’s name.
To read more on related roles, see our Node.js developers for server code and our API developers for connecting services.