Camdiv matches you with a random stranger and puts you both on live video. I wrote separately about the genuinely hard part, which is moderation. This post is about the part people assume is hard and mostly isn't: getting two browsers to see and hear each other. WebRTC handles the media. The interesting work lives around it.
Think about the clock the user feels. They click Start, then they wait, staring at their own face, until a stranger appears. Every millisecond in that gap is something we have to earn back: finding a partner, telling both browsers about each other, negotiating a peer connection, and punching through whatever router or firewall each person sits behind. Four problems, and all of them are latency.
The whole path looks like this:
Step one: find a partner fast
Matching is the first place you can blow the latency budget, and the easiest place to over-engineer. The instinct is to reach for a database or a Redis sorted set and query it on every request. We don't. The matching queues live in memory, in the Node process.
When you click Start, the server first checks whether anyone is already waiting for your chat type (video, audio, or text). If someone is, you're paired on the spot and a room is created. If nobody's there, you go into an in-memory queue, and a loop running every 200ms does a greedy pass over that queue pairing people off. The worst case for a waiting user is a couple hundred milliseconds, not a network round trip to a datastore.
Redis is in the picture, but as a backup, not the source of truth. The in-memory queue is authoritative; Redis gets a best-effort copy in the background so a restart doesn't strand everyone, and so we can add a second server later without rewriting matching. Today it runs as a single instance, and the code says so out loud: the queues are the source of truth, single-instance assumption noted right there in the comment. I like comments that admit their assumptions. The day we scale out, that comment is the first thing the next person needs to read.
One thing that's easy to miss until it bites you: the partner you just matched can vanish in the same instant you matched them. Closed tab, dropped wifi, whatever. So before either side is told about the match, the server checks both sockets are alive, mutates the chat state, then checks again that both are still connected. If one disappeared, the survivor doesn't get dumped onto a dead screen. They go straight back into matchmaking. Getting matched with a ghost is one of the worst feelings on an app like this, and most of avoiding it is just being paranoid at the right three lines of code.
Step two: signaling is a dumb relay, on purpose
Once two people are matched, their browsers need to swap connection details: an SDP offer, an answer, and a stream of ICE candidates (the possible network routes each side can be reached on). That exchange rides over the Socket.IO connection each client already has open.
The backend's job here is almost nothing, and that's the point. It relays. An offer from A gets forwarded to B, B's answer comes back to A, ICE candidates trickle across in both directions. The server never parses SDP, never touches media, never becomes part of the conversation. It's a switchboard.
The one optimization on this path is a hot cache. Every signaling packet has to find the recipient's socket by user id, and a Redis lookup for each one would pile latency onto the busiest path in the app. So there's a plain in-memory map from user id to socket id, checked first, with the slower lookup only as a fallback.
There's a classic WebRTC trap here called glare: if both peers create an offer at the same moment, the negotiation collides. Our fix is boring and cheap. A deterministic tiebreak from the two peers' ids decides which side sends the offer; the other waits to answer. It isn't the full "perfect negotiation" pattern from the spec, and I'd reach for that if we did a lot of mid-call renegotiation. We don't. Each match is a fresh connection, so a deterministic initiator is enough to keep the two sides from talking over each other.
When a connection does drop into the failed state, the client calls restartIce() rather than tearing everything down. That re-gathers routes and often recovers a connection that only hiccuped, with the user seeing nothing worse than a brief freeze.
Step three: ICE without drowning in candidates
This is the part that actually decides whether a connection feels instant or takes three seconds, and where I learned the most counterintuitive lesson.
WebRTC connects peers directly when it can. To do that it gathers candidates (network paths) and tries them. STUN servers help a browser discover its own public address, so two people behind ordinary home routers can talk directly. That covers most users: in our experience roughly 80 to 85 percent connect peer to peer with STUN alone, no relay involved. We point at Google's and Cloudflare's public STUN servers for that.
The rest sit behind strict NATs or corporate firewalls that won't allow a direct path. Those need a TURN server, which relays the media for them. We run our own coturn servers in three regions (New York, Amsterdam, Singapore), and the backend hands each client a TURN config when they ask for one.
Here's the counterintuitive bit. You'd think handing the browser more TURN servers gives it more chances to connect. For speed, the opposite is true. Every TURN server you list multiplies the candidates the browser has to gather and test, and ICE won't settle until it has worked through them. So we don't return all three regions. The backend geo-locates the client by IP and returns the two closest, and only those. Fewer candidates, faster gathering, faster connection. The comment in the code is blunt about it: fewer TURN servers means fewer ICE candidates means faster pairing.
Each TURN server is offered on a few transports, and one of them earns its keep: TURN over TLS on port 443. To a firewall that looks exactly like HTTPS, so it slips through corporate and school networks that block everything else. Plain UDP is tried first because it's lower latency, and 443 is the fallback that decides whether a locked-down network connects at all.
TURN credentials, without leaving the door open
A TURN server that relays media for anyone is a free bandwidth piñata. So you can't hardcode a username and password in the client where anyone can read them out of the network tab.
Instead the credentials are short-lived and computed. The backend shares a secret with the coturn servers. When a client asks, the backend builds a username that is really an expiry timestamp, then signs it with HMAC to produce the password. coturn runs the same signature check and honors the credential until it expires, which for us is 24 hours. Nothing reusable ever sits in the client, and a credential someone scrapes today is dead weight tomorrow.
The client caches that config for six hours and dedupes concurrent fetches, so a page that mounts three things asking for ICE servers still makes one request. If the backend is unreachable, the client falls back to STUN-only, which still connects that 80-something percent. The credential fetch has a five second timeout, because TURN is worth waiting a beat for when you happen to be one of the people who genuinely needs it.
Staying connected on bad networks
Connecting once isn't the job. People walk out of wifi range, switch from wifi to cellular, step into elevators. Two things help.
On the server side, coturn runs with mobility enabled, so a relayed session can survive the client's network changing underneath it. On the client side, the failed-state ICE restart from earlier re-gathers routes and reconnects without rebuilding the whole session.
It isn't magic. A hard network change on a direct peer-to-peer connection can still drop the call, and then you land back in matchmaking. But matchmaking is fast, so the trip from "lost them" to "talking to someone new" is a couple of seconds, which is about as good as this format gets.
The bill
Here's the thing the WebRTC tutorials skip: relays cost money, because they move real bytes. Direct peer-to-peer is free to us, since the media never touches our servers. TURN traffic does touch them, at video bitrates.
The math is what makes the whole format viable. Because only the ~15 percent who can't go direct ever hit a relay, three small droplets are enough. Each coturn box is capped at a few Mbps per session and a few hundred concurrent users, and the three regions together run around twenty dollars a month. If our direct-connect rate fell, that number would climb fast. So the STUN-first, fewest-candidates approach pays off twice: connections settle quicker, and the relay bill stays small.
What I'd flag if you're building this
- A single signaling instance is fine until it isn't. The Redis backup means scaling out is a config change rather than a rewrite, but we haven't had to prove that under load yet.
- The deterministic-initiator trick is enough for fresh one-to-one calls. The moment you renegotiate mid-call, say to add screen share, budget time for proper perfect negotiation instead.
- Geo-selection is only as good as your IP database. It's right the large majority of the time and occasionally wrong, and a wrong guess costs a slightly slower connect, not a broken one. Acceptable trade.
- Watch your direct-connect percentage closely. It's the single number that sets your TURN bill and your connect speed at the same time.
WebRTC gets framed as the hard part of building something like this. Once it works, it mostly keeps working, and the real effort goes into what sits on either side of it: pairing people in milliseconds, relaying their handshake without becoming a bottleneck, and getting through the messy reality of home and office networks without paying to relay everyone. If you want to see it from the user's side, it's live at Camdiv.
The in-memory matchmaking call and the fewest-TURN-candidates trick are the two I'd most happily defend in the comments.
