How many people can join one call?

Four or five before device upload limits bite, if everyone connects to everyone. With a media server, dozens of active participants and hundreds of viewers are achievable. Which of those you need changes both the architecture and the cost considerably.

Do we need our own servers?

For anything beyond a demo, yes, at minimum TURN relay servers. Even one-to-one calling fails for a portion of users behind restrictive networks without them, and those failures look to the user like your product simply not working.

Does it work in mobile apps?

Yes, natively on both iOS and Android as well as in mobile browsers. We build it into apps written in Swift, Kotlin, Flutter, and React Native. Battery and network handling need more attention on mobile than they do on desktop.

Is WebRTC secure and compliant?

Encryption is mandatory in the standard rather than optional, so media is always protected in transit. Compliance depends on the rest: where recordings are stored, who can reach them, and what consent was captured. That is design work, not a checkbox.

WebRTC
Development Services

Stateside Digitals provides WebRTC development services, building live video, voice, and screen sharing directly into web and mobile apps, with the relay and media infrastructure that makes them work in practice.

WebRTC is the technology behind live audio and video inside a browser or an app, with no plugin and nothing to download. It is built into every modern browser and available natively on iOS and Android, which is why a video call can now happen on a web page rather than inside separate software everyone involved has to install and configure first.

For a business, the distinction that matters is latency. Streaming protocols built for broadcast introduce several seconds of delay, which is perfectly fine when nobody is talking back. WebRTC keeps it under a second, and that gap is the difference between a conversation and two people repeatedly talking over each other and apologizing.

Under A Second

Not Streaming Latency

Delay stays low enough for people to interrupt each other naturally, which is the minimum requirement for anything resembling a conversation.

No Plugin, No Install

Works In The Browser

Participants join by opening a link, which removes the support burden that comes with asking customers to install anything first.

Direct Between Peers

Less Server, Less Cost

Small calls connect devices to each other rather than routing everything through infrastructure you are paying to run.

Telehealth Consultations

Patient appointments held in your own app rather than a third-party tool, with recording and consent handled to suit your regulator.

In-App Voice And Video

Calling built into a marketplace, a dating product, or a support flow, so people talk without exchanging phone numbers or leaving the application they are already inside.

Live Support Sessions

Screen sharing and co-browsing so an agent can see what a customer sees, instead of asking them to describe a screen over the phone.

A one-to-one call is genuinely straightforward. The two devices connect more or less directly to each other and your servers barely take part in it at all. That is why WebRTC demonstrations are so persuasive, and why a prototype built in a fortnight works flawlessly for the two people sitting in the room testing it together.

Group calls change the arithmetic completely. Connecting everyone to everyone means each participant uploading their video separately to every other participant, and a phone runs out of upload capacity somewhere around four or five. Beyond that you need a media server taking each stream once and redistributing it, which is real infrastructure at real cost. That line decides your budget.

We build the parts WebRTC leaves to you: signaling over WebSockets, STUN and TURN servers for connections that cannot reach each other directly, and an SFU through mediasoup or LiveKit once calls grow past a handful of people. That extends to recording, simulcast for participants on weaker connections, native iOS and Android integration, and call quality monitoring.

WebRTC
WebSockets
STUN
TURN
SFU
mediasoup
LiveKit
Recording
Simulcast
Native iOS
Native Android
Quality Monitoring

TURN Servers Provisioned

Tested On Real Networks

Call Quality Measured

Recording Consent Handled

Relay capacity is in place before launch, because a call that cannot find a direct route and has nowhere to fall back to simply fails to connect.

Calls get tried on mobile data, hotel wifi, and corporate firewalls, not only on the office connection where everything always works.

Packet loss, jitter, and connection failures are recorded per call, so a complaint about a bad meeting can be checked rather than guessed at.

Who is told, what they agreed to, and where the file ends up are settled in the design, since the rules differ by state and by sector.

TURN Servers Provisioned

Tested On Real Networks

Call Quality Measured

Recording Consent Handled

For most products, use a service. Twilio, Agora, Daily, and LiveKit handle the infrastructure that consumes the budget on a raw build. Building direct makes sense at high volume, where per-minute pricing overtakes running your own, or under strict data rules.

WebRTC
Development Services