VoipArchitects
Service / WebRTC

WebRTC voice & video integration

Put calling directly in the browser or your app — click-to-call, in-app support, and agent consoles — bridged cleanly into the SIP world and tuned to connect on real networks.

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Browser & App Real-Time Voice/Video

WebRTC is the technology behind real-time voice and video that runs with nothing but a browser. It is what turns a support page into a click-to-call experience and an internal tool into a full agent console, but its real-world reliability depends entirely on signaling, media, and NAT traversal being configured correctly.

We build the WebRTC layer and connect it to your existing telephony — Asterisk, FreeSWITCH, or a SIP carrier — so browser and app calls interoperate with your phones, numbers, and queues instead of living on an island.

What we do

Our WebRTC engagement, in scope

WebRTC-to-SIP gateway

SIP-over-WebSocket into Asterisk or Verto/mod_verto into FreeSWITCH, so browser calls flow into your existing extensions, trunks, and queues without a separate phone system.

Click-to-call & in-app calling

Embed voice or video into your website or product — support buttons, browser softphones, and agent consoles — with secure signaling and DTLS-SRTP media by default.

TURN/STUN & NAT traversal

coturn deployment and ICE configuration so calls connect reliably behind firewalls, symmetric NAT, and mobile networks — the conditions that break most naive WebRTC setups.

Codec & network tuning

Codec selection, bandwidth management, and load testing against real network conditions so audio and video hold up outside a clean office LAN.

Deliverables

What you receive

  • WebRTC signaling gateway configured against Asterisk or FreeSWITCH
  • TURN/STUN (coturn) deployed and traversal tested
  • DTLS-SRTP media encryption verified end to end
  • Client-side integration guidance for your developers
  • Load-test results and handover documentation
Who it's for

Typical clients

  • Products embedding click-to-call or in-app voice/video
  • Browser-based agent consoles and softphones for remote teams
  • Bridging web or mobile users into an existing SIP/PBX platform
Deployment options

Your server or a dedicated server from us

We can install and configure WebRTC on infrastructure you already own, or provide a fully managed dedicated server so you don't have to source hardware or a data center yourself.

Install on your own server

Already have a cloud VM or on-premise hardware? We deploy WebRTC straight onto your infrastructure — your cloud account, your VPS, or your own metal. You keep full ownership of the server, the data, and the credentials; we handle the build, hardening, and handover.

Dedicated server provided by us

Prefer a turnkey setup? We can provision a dedicated server sized for your call volume with WebRTC pre-installed, secured, and ready to run — plus optional ongoing management, monitoring, and backups so you have a single point of contact for the whole stack.

Not sure which fits? Compare managed hosting or tell us your requirements and we'll recommend the right deployment model.

In depth

Understanding WebRTC

Voice and video that runs in the browser

WebRTC lets your customers place and receive calls directly from a browser tab or mobile app — no desktop softphone, no plugin, no phone number to dial. That is what powers click-to-call buttons, in-app support calls, browser-based agent consoles, and embedded video meetings. Because the media path is encrypted end to end with DTLS-SRTP by default, it is also one of the most secure ways to move real-time audio and video.

We build the WebRTC layer and connect it to the telephony you already run, so a call that starts in a browser can land on a SIP phone, a PSTN number, or a contact-center queue without the caller ever knowing there was a bridge in between.

Bridging WebRTC to SIP, Asterisk, and FreeSWITCH

On its own, WebRTC only connects browsers to each other. The value for most businesses comes from bridging it into a real phone system. We configure the signaling gateway — SIP-over-WebSocket into Asterisk, or Verto and mod_verto into FreeSWITCH — and the media handling so browser calls interoperate cleanly with SIP trunks, extensions, and carriers.

That means one platform can serve a browser softphone, a desk phone, and a mobile app at the same time, all reaching the same numbers, queues, and recording rules.

Making WebRTC connect reliably in the real world

The hardest part of WebRTC is not the demo — it is the calls that fail behind corporate firewalls, symmetric NAT, and mobile networks. We deploy and tune STUN and TURN servers (coturn) so media traverses restrictive networks, configure ICE correctly, and select codecs and bandwidth settings that hold up on real connections rather than a clean office LAN.

You get a deployment that is load-tested against real network conditions, with guidance for your developers on the client-side SDK or JavaScript integration they will build against.

FAQ

Frequently asked questions

Yes — that is the most common request. We bridge browser and app calls into Asterisk via SIP-over-WebSocket or into FreeSWITCH via Verto, so WebRTC users share the same numbers, extensions, queues, and recording as your desk phones.

Pairs well with

Platforms that work together

Most deployments combine more than one platform. Here is what commonly pairs with this service.

Carrier-Grade Softswitch

FreeSWITCH setup & configuration

FreeSWITCH deployments for high-concurrency voice — softswitch routing, WebRTC, and conferencing that scales.

PBX & Voice Application Server

Asterisk PBX setup & configuration

Production Asterisk PBX builds — dial plans, SIP trunking, IVR, and call routing configured to your business logic.

Deploy WebRTC with confidence

Tell us your requirements and we'll scope a fixed-fee deployment configured to how your business actually runs.

Request a quoteView pricing