Driansh Softtech

VoIP Development

WebRTC development company for voice and video apps

We build WebRTC apps for voice, video, chat and screen sharing. They run in the browser and on mobile, and connect to your SIP or VoIP platform.
Talk to a WebRTC engineerSee what we build

Free consultation with our engineers

What we build with WebRTC

WebRTC development lets your users talk, see and share in real time without installing plugins. We build web softphones, video meeting apps, contact center agent screens and in-app calling for web and mobile. Each one is designed around your users, workflows and product.

We also handle the server side: signaling, TURN and STUN, media servers and gateways to SIP. That way your WebRTC app can call phone numbers, desk phones and existing VoIP systems as well as other browsers.

Technology we work with

  • WebRTC
  • SIP.js
  • JsSIP
  • Janus
  • mediasoup
  • coturn
  • FreeSWITCH
  • Kamailio
  • React
  • Node.js
  • Flutter

What we offer

WebRTC development services we offer

Web softphones

Give users a browser phone that registers to your SIP platform, so they can call from any computer.

Video calling & meetings

Build one-to-one and group video with layouts, moderation and recording that fit your product.

Screen & file sharing

Add screen sharing, chat and file transfer so teams can work together inside one call.

SIP & PSTN gateways

Bridge WebRTC to SIP trunks and phone numbers, so browser users can reach any phone.

Mobile WebRTC apps

Ship native or cross-platform calling apps that share accounts and features with your web app.

Media servers & support

Set up TURN, SFU and recording servers, then monitor and tune call quality after launch.

Building a WebRTC app?

Tell us what your users need to do. A WebRTC engineer will review it and suggest the right setup.
Talk to a WebRTC engineer

What we build

Solutions we build with WebRTC

Audio & video conferencing

Browser meetings with dial-in, moderation and recording.
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Call center agent apps

Browser-based agent desktops with softphone and CRM data.
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Unified communications

Calls, video, chat and presence in one web workspace.
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Omnichannel inbox

Voice and chat conversations handled from a single inbox.
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Live call monitoring

Listen, whisper or barge into live calls from the browser.
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Why Driansh

Why teams choose Driansh for WebRTC

Browser to carrier

We build the client, the media servers and the SIP side, so calls work end to end.

Secure by default

WebRTC encrypts media with DTLS-SRTP, and we add authentication and access control as well.

Fits your product

We embed calling into your existing web or mobile app instead of forcing a separate tool.

You own the code

Source code and documentation are handed over, so your team can extend the app freely.

How we work

From idea to live platform in four steps

  1. 01

    Discover

    We review your requirements, traffic and existing stack, then agree scope, milestones and cost.

  2. 02

    Design

    We plan the architecture, integrations and failover so the platform fits your growth plans.

  3. 03

    Build & test

    We develop in short sprints with regular demos, then load-test and QA before go-live.

  4. 04

    Deploy & support

    We launch on your servers or cloud and stay on for monitoring, fixes and upgrades.

Frequently asked questions

Get in touch

Building a WebRTC app?

Tell us what your users need to do. A WebRTC engineer will review it and suggest the right setup.

Tell us about your project

Share a few details and the right engineer will get back to you.

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