{"id":18664,"date":"2026-09-01T13:34:17","date_gmt":"2026-09-01T04:34:17","guid":{"rendered":"https:\/\/circleip.com\/?p=18664"},"modified":"2026-09-04T11:10:16","modified_gmt":"2026-09-04T02:10:16","slug":"how-webrtc-works","status":"publish","type":"post","link":"https:\/\/circleip.com\/en\/how-webrtc-works\/","title":{"rendered":"How WebRTC Works: Signaling, ICE, STUN, TURN & Browser Calling"},"content":{"rendered":"\t\t
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If you’ve looked at a cloud phone system for your business, you’ve probably run into the term WebRTC without much explanation of what it actually is. It shows up in feature lists next to things like “no software install” and “call from any browser,” and then the explanation stops there.<\/p>\n

That’s fine if you just want to make calls. It’s not fine if you’re the person responsible for deciding whether this technology will hold up on your office Wi-Fi, your employees’ home networks, and your company VPN, or if you want to understand what “encrypted by default” actually means for a system that’s about to carry every customer call your business makes.<\/p>\n

This guide explains the mechanism itself: what WebRTC is, how a call actually gets from one browser to another (or to a real phone number), what’s encrypted and what isn’t, and what tends to go wrong in real offices rather than in vendor demos. It’s written for the person evaluating or running a phone system, not for a developer building one, though the underlying facts are the same either way.<\/p>\t\t\t\t\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t

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How WebRTC Works: What is it?<\/h2>\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t
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WebRTC (Web Real-Time Communication) is a set of open technical standards that lets web browsers and other WebRTC-compatible endpoints send and receive live audio, video, and data in real time. In a compatible web browser, it can do this without a plugin, a downloaded app, or dedicated hardware. Depending on the connection, that traffic may travel directly between endpoints or through TURN relays, media servers, or other provider infrastructure. It’s standardized jointly by the<\/span> W3C<\/span><\/a>, which governs the browser-side API, and the<\/span> IETF<\/span><\/a>, which governs the underlying network protocols.<\/span><\/p>\n

Before WebRTC existed, browser-based calling meant Flash, Java applets, or a proprietary plugin, all of which required installation, all of which had their own security patch cycles, and none of which worked consistently across browsers. Google open-sourced the early version of WebRTC around 2011 and helped push it through standardization over the following decade. As of today, every major browser (Chrome, Firefox, Safari, and Edge) ships WebRTC natively. That universality is precisely why a browser-based phone works the same way whether your employee is on a Mac at home, a Windows laptop in the office, or a Chromebook in a hotel lobby: the underlying mechanism doesn’t change per device.<\/span><\/p>\n

However, it’s worth being precise about what WebRTC is <\/span>not.<\/span><\/i><\/p>\n

It is not a phone system. It doesn’t know what a business phone number is, doesn’t route calls to the right department, doesn’t handle voicemail, and has no built-in concept of a queue or an extension. WebRTC solves exactly one problem: getting audio, video, or data from one point to another, reliably and securely, over an internet connection that was never designed to make that easy. Everything that makes a browser phone feel like a phone system, the number, the routing rules, the call history, is built by the platform provider on top of WebRTC, not by WebRTC itself.\u00a0<\/span><\/p>\t\t\t\t\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t

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For a broader look at how phone numbers, lines, and call routing work together, see:\u00a0What is a Cloud PBX?<\/strong><\/span><\/a><\/p>\t\t\t\t\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t

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The 3 Jobs WebRTC Has to Do<\/h2>\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t
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Every WebRTC call, no matter how simple it looks on screen, solves three separate technical problems in sequence. Signaling is required to establish the call, but WebRTC itself does not define how signaling works. The application or phone service implements it.<\/p>\t\t\t\t\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t

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Introduction (signaling):<\/h3>\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t
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Two devices that have never spoken before need a way to exchange basic information about each other, including what audio\/video formats they support and how they might be reached on the network.<\/p>\t\t\t\t\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t

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Pathfinding (connectivity):<\/h3>\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t
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Once introduced, the two devices need to find an actual working route across the internet between them, which is harder than it sounds because of how home and office networks are built.<\/p>\t\t\t\t\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t

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Secure delivery (media transport):<\/h3>\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t
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Once a route exists, the actual audio and video need to travel across it encrypted, in real time, adjusting automatically if the network gets congested partway through the call.<\/p>\t\t\t\t\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t

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If employees currently use their personal mobile numbers for customer calls, it\u2019s also worth reviewing the risks of using personal phone numbers for business<\/strong><\/a> before choosing a PC softphone.<\/p>\t\t\t\t\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t

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Job 1: How Do Two Browsers Introduce Themselves to Each Other?<\/h2>\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t
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A signaling channel is how two browsers exchange the setup information a call needs before any audio ever flows: what codecs each side can use, whether video is involved, and the technical details ICE will need in the next step.<\/p>\n

Here’s the detail that surprises most people: WebRTC itself does not define how signaling works. The standard leaves it entirely up to the application. In practice, this is the part of the system a phone provider actually builds and controls. Both browsers connect to the provider’s signaling service to swap setup information, and that service may remain involved after the call connects for functions such as hold, transfer, renegotiation, connection recovery, or hang-up.<\/p>\n

This matters more than it sounds like it should, because the provider’s signaling and call-control layer is also where the “phone system” part of a phone system lives. It’s what decides that a call to your main business number should ring the sales team’s browser phones in a specific order, or fall to voicemail after four rings, or route differently outside business hours. None of that is WebRTC. All of it sits in the layer a provider builds around WebRTC.<\/p>\t\t\t\t\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t

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To better understand the technology behind PC softphones, it also helps to understand the difference between a traditional PBX and a cloud PBX<\/strong><\/a>.<\/p>\t\t\t\t\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t

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Job 2: How Does a Browser Call Find a Path Through Two Different Networks?<\/h2>\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t
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ICE<\/span><\/a> (Interactive Connectivity Establishment) is the process WebRTC uses to find a working network route between two devices that can’t see each other directly. It’s the part of the mechanism that explains most real-world call problems, so it’s worth understanding even at a non-technical level.<\/span><\/p>\n

Almost every device on a business or home network sits behind a router doing something called NAT (Network Address Translation), which is why your laptop has a private address like 192.168.1.14 that means nothing outside your own network. Two browsers on two different networks, each only aware of their own private address, have no direct way to find each other. There’s no dial tone equivalent that just works.<\/span><\/p>\n

Each browser gathers a set of possible ways it might be reachable and tests the available candidate pairs in priority order: <\/span><\/p>\t\t\t\t\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t

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Direct local connection:<\/h3>\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t
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Works when the endpoints can reach each other directly using host candidates. This commonly happens on the same local network, but can also happen across networks when publicly routable addresses are available.<\/p>\t\t\t\t\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t

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STUN-assisted connection (Session Traversal Utilities for NAT): <\/h3>\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t
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A lightweight server briefly tells each browser what its address looks like from outside its own router, letting many home and office NAT setups establish a direct connection anyway. Whether this works depends on the NAT type, firewall rules, VPN configuration, mobile carrier network, and other environmental factors.<\/p>\n

Learn More<\/em><\/a><\/p>\t\t\t\t\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t

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TURN-relayed connection (Traversal Using Relays around NAT):<\/h3>\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t
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The fallback for the networks STUN can’t solve, such as strict corporate firewalls, certain mobile carrier networks, and “symmetric” NAT setups common in larger enterprises. A TURN server sits in the middle and relays every packet of the call between the two sides. It is an important fallback, but it can still fail if the network blocks the available TURN transports, authentication fails, or the provider doesn’t have enough capacity. It also isn’t free in either sense of the word: it adds latency, and it consumes real bandwidth on the provider’s infrastructure for the full duration of every relayed call, not just the setup.<\/p>\n

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This is the single most important operational fact about WebRTC calling for anyone choosing a provider: if a provider doesn’t run TURN infrastructure, or runs too little of it, calls from certain office networks may not connect at all. Not degraded, not choppy, simply unable to establish. It’s also why “it worked fine in the demo” doesn’t guarantee it’ll work from your specific office; the demo may have been run on a network permissive enough that STUN alone was sufficient.<\/p>\n

Once ICE finds a working path, that path is what carries the actual call. How long this takes depends on candidate gathering, firewall behaviour, TURN use, authentication, and general network conditions.<\/p>\t\t\t\t\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t

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Job 3: What Does \"Encrypted by Default\" Actually Mean in WebRTC?<\/h2>\t\t\t\t<\/div>\n\t\t\t\t<\/div>\n\t\t\t\t
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Encryption in WebRTC isn’t a setting anyone turns on. It’s mandatory for every media stream, with no unencrypted mode to fall back to. The<\/span> IETF’s WebRTC Security Architecture (RFC 8827)<\/span><\/a> requires it outright, which is a stronger baseline than a lot of older telephony technology, where encryption is often an add-on rather than the default.\u00a0<\/span><\/p>\n

Once a path exists, the call needs to travel across it without being readable by an ordinary network observer, including the office network, the ISP, or a TURN server that is only relaying the encrypted packets.<\/span><\/p>\n

In practical terms, two mechanisms do the work:<\/span><\/p>\n